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[alsa-devel] [PATCH] ASoC: sgtl5000: Allow LRCLK pad drive strength to be changed
by Fabio Estevam 10 Apr '17

10 Apr '17

Introduce the "lrclk-strength" property to allow LRCLK pad drive strength to be changed via device tree. When running a stress playback loop test on a mx6dl wandboard channel swap can be noticed on about 10% of the times. While debugging this issue I noticed that when probing the SGTL5000 LRCLK pin with the scope the swap did not happen. After removing the probe the swap started to happen again. After changing the LRCLK pad drive strength to the maximum value the issue is gone. Same fix works on a mx6dl Colibri board as well. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 9 +++++++++ sound/soc/codecs/sgtl5000.c | 19 ++++++++++++++++++- 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 5666da7..7a73a9d 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -26,6 +26,15 @@ Optional properties: If this node is not mentioned or the value is unknown, then the value is set to 1.25V. +- lrclk-strength: the LRCLK pad strength. Possible values are: +0, 1, 2 and 3 as per the table below: + +VDDIO 1.8V 2.5V 3.3V +0 = Disable +1 = 1.66 mA 2.87 mA 4.02 mA +2 = 3.33 mA 5.74 mA 8.03 mA +3 = 4.99 mA 8.61 mA 12.05 mA + Example: codec: sgtl5000@0a { diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1589325..5a2702e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -99,6 +99,13 @@ enum sgtl5000_micbias_resistor { SGTL5000_MICBIAS_8K = 8, }; +enum { + I2S_LRCLK_STRENGTH_DISABLE, + I2S_LRCLK_STRENGTH_LOW, + I2S_LRCLK_STRENGTH_MEDIUM, + I2S_LRCLK_STRENGTH_HIGH, +}; + /* sgtl5000 private structure in codec */ struct sgtl5000_priv { int sysclk; /* sysclk rate */ @@ -111,6 +118,7 @@ struct sgtl5000_priv { int revision; u8 micbias_resistor; u8 micbias_voltage; + u8 lrclk_strength; }; /* @@ -1089,6 +1097,7 @@ static int sgtl5000_enable_regulators(struct i2c_client *client) static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; + u16 reg; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* power up sgtl5000 */ @@ -1118,7 +1127,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_DAC_MUTE_RIGHT | SGTL5000_DAC_MUTE_LEFT); - snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + reg = ((sgtl5000->lrclk_strength) << SGTL5000_PAD_I2S_LRCLK_SHIFT | 0x5f); + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, reg); snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, SGTL5000_HP_ZCD_EN | @@ -1347,6 +1357,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, } } + sgtl5000->lrclk_strength = I2S_LRCLK_STRENGTH_LOW; + if (!of_property_read_u32(np, "lrclk-strength", &value)) { + if (value > I2S_LRCLK_STRENGTH_HIGH) + value = I2S_LRCLK_STRENGTH_LOW; + sgtl5000->lrclk_strength = value; + } + /* Ensure sgtl5000 will start with sane register values */ sgtl5000_fill_defaults(client); -- 2.7.4

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[alsa-devel] [PATCH v3 1/2] ASoC: fsl_ssi: Remove FSLSSI_I2S_RATES definition
by Fabio Estevam 10 Apr '17

10 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> The comment for the FSLSSI_I2S_RATES definition states that the driver currently only supports I2S slave mode, which is no longer correct. As FSLSSI_I2S_RATES is the same as the standard SNDRV_PCM_RATE_CONTINUOUS, just remove its definition and its comments to make the code simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Chnages since v1: - Simply remove FSLSSI_I2S_RATES sound/soc/fsl/fsl_ssi.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fde08660..0b74d1c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -55,16 +55,6 @@ #include "imx-pcm.h" /** - * FSLSSI_I2S_RATES: sample rates supported by the I2S - * - * This driver currently only supports the SSI running in I2S slave mode, - * which means the codec determines the sample rate. Therefore, we tell - * ALSA that we support all rates and let the codec driver decide what rates - * are really supported. - */ -#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS - -/** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI * * The SSI has a limitation in that the samples must be in the same byte @@ -1212,14 +1202,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 32, - .rates = FSLSSI_I2S_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 32, - .rates = FSLSSI_I2S_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, .formats = FSLSSI_I2S_FORMATS, }, .ops = &fsl_ssi_dai_ops, -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: fsl_ssi: Remove an outdated comment
by Fabio Estevam 10 Apr '17

10 Apr '17

The comment for the FSLSSI_I2S_RATES definition states that the driver currently only supports I2S slave mode, which is no longer correct. Remove such outdated comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fde08660..b450d8d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -54,14 +54,6 @@ #include "fsl_ssi.h" #include "imx-pcm.h" -/** - * FSLSSI_I2S_RATES: sample rates supported by the I2S - * - * This driver currently only supports the SSI running in I2S slave mode, - * which means the codec determines the sample rate. Therefore, we tell - * ALSA that we support all rates and let the codec driver decide what rates - * are really supported. - */ #define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** -- 2.7.4

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[alsa-devel] Applied "ASoC: add hi6210-i2s DT bindings" to the asoc tree
by Mark Brown 10 Apr '17

10 Apr '17

The patch ASoC: add hi6210-i2s DT bindings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9e615f63cbafa2147fd302ab495d7b785306bc26 Mon Sep 17 00:00:00 2001 From: John Stultz <john.stultz(a)linaro.org> Date: Fri, 31 Mar 2017 15:05:59 -0700 Subject: [PATCH] ASoC: add hi6210-i2s DT bindings Adds DT bindings documentation for the hi6210-i2s driver. Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/hisilicon,hi6210-i2s.txt | 34 ++++++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt new file mode 100644 index 000000000000..680bb03577ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt @@ -0,0 +1,34 @@ +* Hisilicon 6210 i2s controller + +Required properties: + +- compatible: should be one of the following: + - "hisilicon,hi6210-i2s" +- reg: physical base address of the i2s controller unit and length of + memory mapped region. +- interrupts: should contain the i2s interrupt. +- clocks: a list of phandle + clock-specifier pairs, one for each entry + in clock-names. +- clock-names: should contain following: + - "dacodec" + - "i2s-base" +- dmas: DMA specifiers for tx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should be "tx" and "rx" +- hisilicon,sysctrl-syscon: phandle to sysctrl syscon +- #sound-dai-cells: Should be set to 1 (for multi-dai) + +Example for the hi6210 i2s controller: + +i2s0: i2s@f7118000{ + compatible = "hisilicon,hi6210-i2s"; + reg = <0x0 0xf7118000 0x0 0x8000>; /* i2s unit */ + interrupts = <GIC_SPI 123 IRQ_TYPE_LEVEL_HIGH>; /* 155 "DigACodec_intr"-32 */ + clocks = <&sys_ctrl HI6220_DACODEC_PCLK>, + <&sys_ctrl HI6220_BBPPLL0_DIV>; + clock-names = "dacodec", "i2s-base"; + dmas = <&dma0 15 &dma0 14>; + dma-names = "rx", "tx"; + hisilicon,sysctrl-syscon = <&sys_ctrl>; + #sound-dai-cells = <1>; +}; -- 2.11.0

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[alsa-devel] Applied "ASoC: hisilicon: Add hi6210 i2s audio driver" to the asoc tree
by Mark Brown 10 Apr '17

10 Apr '17

The patch ASoC: hisilicon: Add hi6210 i2s audio driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0bf750f4cbe1406d8819cbec2a3ff6beeb824617 Mon Sep 17 00:00:00 2001 From: Andy Green <andy.green(a)linaro.org> Date: Fri, 31 Mar 2017 15:06:00 -0700 Subject: [PATCH] ASoC: hisilicon: Add hi6210 i2s audio driver Add driver for hi6210 i2s controller found on hi6220 boards. Signed-off-by: Andy Green <andy.green(a)linaro.org> [jstultz: Forward ported to mainline, fairly major rework based on suggestions from Mark Brown] Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/hisilicon/Kconfig | 5 + sound/soc/hisilicon/Makefile | 1 + sound/soc/hisilicon/hi6210-i2s.c | 628 +++++++++++++++++++++++++++++++++++++++ sound/soc/hisilicon/hi6210-i2s.h | 276 +++++++++++++++++ 6 files changed, 912 insertions(+) create mode 100644 sound/soc/hisilicon/Kconfig create mode 100644 sound/soc/hisilicon/Makefile create mode 100644 sound/soc/hisilicon/hi6210-i2s.c create mode 100644 sound/soc/hisilicon/hi6210-i2s.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 182d92efc7c8..9df9658b552b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/dwc/Kconfig" source "sound/soc/fsl/Kconfig" +source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9a30f21d16ee..2f6aabb8b4c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -27,6 +27,7 @@ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ +obj-$(CONFIG_SND_SOC) += hisilicon/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ diff --git a/sound/soc/hisilicon/Kconfig b/sound/soc/hisilicon/Kconfig new file mode 100644 index 000000000000..4356d5a1d338 --- /dev/null +++ b/sound/soc/hisilicon/Kconfig @@ -0,0 +1,5 @@ +config SND_I2S_HI6210_I2S + tristate "Hisilicon I2S controller" + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Hisilicon I2S diff --git a/sound/soc/hisilicon/Makefile b/sound/soc/hisilicon/Makefile new file mode 100644 index 000000000000..e8095e2af91a --- /dev/null +++ b/sound/soc/hisilicon/Makefile @@ -0,0 +1 @@ +obj-$(CONFIG_SND_I2S_HI6210_I2S) += hi6210-i2s.o diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c new file mode 100644 index 000000000000..45691b70060c --- /dev/null +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -0,0 +1,628 @@ +/* + * linux/sound/soc/m8m/hi6210_i2s.c - I2S IP driver + * + * Copyright (C) 2015 Linaro, Ltd + * Author: Andy Green <andy.green(a)linaro.org> + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver only deals with S2 interface (BT) + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/jiffies.h> +#include <linux/io.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <linux/interrupt.h> +#include <linux/reset.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> +#include <linux/mfd/syscon.h> +#include <linux/reset-controller.h> +#include <linux/clk.h> + +#include "hi6210-i2s.h" + +struct hi6210_i2s { + struct device *dev; + struct reset_control *rc; + struct clk *clk[8]; + int clocks; + struct snd_soc_dai_driver dai; + void __iomem *base; + struct regmap *sysctrl; + phys_addr_t base_phys; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int clk_rate; + spinlock_t lock; + int rate; + int format; + u8 bits; + u8 channels; + u8 id; + u8 channel_length; + u8 use; + u32 master:1; + u32 status:1; +}; + +#define SC_PERIPH_CLKEN1 0x210 +#define SC_PERIPH_CLKDIS1 0x214 + +#define SC_PERIPH_CLKEN3 0x230 +#define SC_PERIPH_CLKDIS3 0x234 + +#define SC_PERIPH_CLKEN12 0x270 +#define SC_PERIPH_CLKDIS12 0x274 + +#define SC_PERIPH_RSTEN1 0x310 +#define SC_PERIPH_RSTDIS1 0x314 +#define SC_PERIPH_RSTSTAT1 0x318 + +#define SC_PERIPH_RSTEN2 0x320 +#define SC_PERIPH_RSTDIS2 0x324 +#define SC_PERIPH_RSTSTAT2 0x328 + +#define SOC_PMCTRL_BBPPLLALIAS 0x48 + +enum { + CLK_DACODEC, + CLK_I2S_BASE, +}; + +static inline void hi6210_write_reg(struct hi6210_i2s *i2s, int reg, u32 val) +{ + writel(val, i2s->base + reg); +} + +static inline u32 hi6210_read_reg(struct hi6210_i2s *i2s, int reg) +{ + return readl(i2s->base + reg); +} + +int hi6210_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + int ret, n; + u32 val; + + /* deassert reset on ABB */ + regmap_read(i2s->sysctrl, SC_PERIPH_RSTSTAT2, &val); + if (val & BIT(4)) + regmap_write(i2s->sysctrl, SC_PERIPH_RSTDIS2, BIT(4)); + + for (n = 0; n < i2s->clocks; n++) { + ret = clk_prepare_enable(i2s->clk[n]); + if (ret) { + while (n--) + clk_disable_unprepare(i2s->clk[n]); + return ret; + } + } + + ret = clk_set_rate(i2s->clk[CLK_I2S_BASE], 49152000); + if (ret) { + dev_err(i2s->dev, "%s: setting 49.152MHz base rate failed %d\n", + __func__, ret); + return ret; + } + + /* enable clock before frequency division */ + regmap_write(i2s->sysctrl, SC_PERIPH_CLKEN12, BIT(9)); + + /* enable codec working clock / == "codec bus clock" */ + regmap_write(i2s->sysctrl, SC_PERIPH_CLKEN1, BIT(5)); + + /* deassert reset on codec / interface clock / working clock */ + regmap_write(i2s->sysctrl, SC_PERIPH_RSTEN1, BIT(5)); + regmap_write(i2s->sysctrl, SC_PERIPH_RSTDIS1, BIT(5)); + + /* not interested in i2s irqs */ + val = hi6210_read_reg(i2s, HII2S_CODEC_IRQ_MASK); + val |= 0x3f; + hi6210_write_reg(i2s, HII2S_CODEC_IRQ_MASK, val); + + + /* reset the stereo downlink fifo */ + val = hi6210_read_reg(i2s, HII2S_APB_AFIFO_CFG_1); + val |= (BIT(5) | BIT(4)); + hi6210_write_reg(i2s, HII2S_APB_AFIFO_CFG_1, val); + + val = hi6210_read_reg(i2s, HII2S_APB_AFIFO_CFG_1); + val &= ~(BIT(5) | BIT(4)); + hi6210_write_reg(i2s, HII2S_APB_AFIFO_CFG_1, val); + + + val = hi6210_read_reg(i2s, HII2S_SW_RST_N); + val &= ~(HII2S_SW_RST_N__ST_DL_WORDLEN_MASK << + HII2S_SW_RST_N__ST_DL_WORDLEN_SHIFT); + val |= (HII2S_BITS_16 << HII2S_SW_RST_N__ST_DL_WORDLEN_SHIFT); + hi6210_write_reg(i2s, HII2S_SW_RST_N, val); + + val = hi6210_read_reg(i2s, HII2S_MISC_CFG); + /* mux 11/12 = APB not i2s */ + val &= ~HII2S_MISC_CFG__ST_DL_TEST_SEL; + /* BT R ch 0 = mixer op of DACR ch */ + val &= ~HII2S_MISC_CFG__S2_DOUT_RIGHT_SEL; + val &= ~HII2S_MISC_CFG__S2_DOUT_TEST_SEL; + + val |= HII2S_MISC_CFG__S2_DOUT_RIGHT_SEL; + /* BT L ch = 1 = mux 7 = "mixer output of DACL */ + val |= HII2S_MISC_CFG__S2_DOUT_TEST_SEL; + hi6210_write_reg(i2s, HII2S_MISC_CFG, val); + + val = hi6210_read_reg(i2s, HII2S_SW_RST_N); + val |= HII2S_SW_RST_N__SW_RST_N; + hi6210_write_reg(i2s, HII2S_SW_RST_N, val); + + return 0; +} +void hi6210_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + int n; + + for (n = 0; n < i2s->clocks; n++) + clk_disable_unprepare(i2s->clk[n]); + + regmap_write(i2s->sysctrl, SC_PERIPH_RSTEN1, BIT(5)); +} + +static void hi6210_i2s_txctrl(struct snd_soc_dai *cpu_dai, int on) +{ + struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + u32 val; + + spin_lock(&i2s->lock); + if (on) { + /* enable S2 TX */ + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val |= HII2S_I2S_CFG__S2_IF_TX_EN; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + } else { + /* disable S2 TX */ + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val &= ~HII2S_I2S_CFG__S2_IF_TX_EN; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + } + spin_unlock(&i2s->lock); +} + +static void hi6210_i2s_rxctrl(struct snd_soc_dai *cpu_dai, int on) +{ + struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + u32 val; + + spin_lock(&i2s->lock); + if (on) { + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val |= HII2S_I2S_CFG__S2_IF_RX_EN; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + } else { + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val &= ~HII2S_I2S_CFG__S2_IF_RX_EN; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + } + spin_unlock(&i2s->lock); +} + +static int hi6210_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + /* + * We don't actually set the hardware until the hw_params + * call, but we need to validate the user input here. + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + break; + default: + return -EINVAL; + } + + i2s->format = fmt; + i2s->master = (i2s->format & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBS_CFS; + + return 0; +} + +static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct hi6210_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + u32 bits = 0, rate = 0, signed_data = 0, fmt = 0; + u32 val; + struct snd_dmaengine_dai_dma_data *dma_data; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U16_LE: + signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; + /* fallthru */ + case SNDRV_PCM_FORMAT_S16_LE: + bits = HII2S_BITS_16; + break; + case SNDRV_PCM_FORMAT_U24_LE: + signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; + /* fallthru */ + case SNDRV_PCM_FORMAT_S24_LE: + bits = HII2S_BITS_24; + break; + default: + dev_err(cpu_dai->dev, "Bad format\n"); + return -EINVAL; + } + + + switch (params_rate(params)) { + case 8000: + rate = HII2S_FS_RATE_8KHZ; + break; + case 16000: + rate = HII2S_FS_RATE_16KHZ; + break; + case 32000: + rate = HII2S_FS_RATE_32KHZ; + break; + case 48000: + rate = HII2S_FS_RATE_48KHZ; + break; + case 96000: + rate = HII2S_FS_RATE_96KHZ; + break; + case 192000: + rate = HII2S_FS_RATE_192KHZ; + break; + default: + dev_err(cpu_dai->dev, "Bad rate: %d\n", params_rate(params)); + return -EINVAL; + } + + if (!(params_channels(params))) { + dev_err(cpu_dai->dev, "Bad channels\n"); + return -EINVAL; + } + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + + switch (bits) { + case HII2S_BITS_24: + i2s->bits = 32; + dma_data->addr_width = 3; + break; + default: + i2s->bits = 16; + dma_data->addr_width = 2; + } + i2s->rate = params_rate(params); + i2s->channels = params_channels(params); + i2s->channel_length = i2s->channels * i2s->bits; + + val = hi6210_read_reg(i2s, HII2S_ST_DL_FIFO_TH_CFG); + val &= ~((HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AEMPTY_MASK << + HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AEMPTY_SHIFT) | + (HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AFULL_MASK << + HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AFULL_SHIFT) | + (HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AEMPTY_MASK << + HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AEMPTY_SHIFT) | + (HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AFULL_MASK << + HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AFULL_SHIFT)); + val |= ((16 << HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AEMPTY_SHIFT) | + (30 << HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AFULL_SHIFT) | + (16 << HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AEMPTY_SHIFT) | + (30 << HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AFULL_SHIFT)); + hi6210_write_reg(i2s, HII2S_ST_DL_FIFO_TH_CFG, val); + + + val = hi6210_read_reg(i2s, HII2S_IF_CLK_EN_CFG); + val |= (BIT(19) | BIT(18) | BIT(17) | + HII2S_IF_CLK_EN_CFG__S2_IF_CLK_EN | + HII2S_IF_CLK_EN_CFG__S2_OL_MIXER_EN | + HII2S_IF_CLK_EN_CFG__S2_OL_SRC_EN | + HII2S_IF_CLK_EN_CFG__ST_DL_R_EN | + HII2S_IF_CLK_EN_CFG__ST_DL_L_EN); + hi6210_write_reg(i2s, HII2S_IF_CLK_EN_CFG, val); + + + val = hi6210_read_reg(i2s, HII2S_DIG_FILTER_CLK_EN_CFG); + val &= ~(HII2S_DIG_FILTER_CLK_EN_CFG__DACR_SDM_EN | + HII2S_DIG_FILTER_CLK_EN_CFG__DACR_HBF2I_EN | + HII2S_DIG_FILTER_CLK_EN_CFG__DACR_AGC_EN | + HII2S_DIG_FILTER_CLK_EN_CFG__DACL_SDM_EN | + HII2S_DIG_FILTER_CLK_EN_CFG__DACL_HBF2I_EN | + HII2S_DIG_FILTER_CLK_EN_CFG__DACL_AGC_EN); + val |= (HII2S_DIG_FILTER_CLK_EN_CFG__DACR_MIXER_EN | + HII2S_DIG_FILTER_CLK_EN_CFG__DACL_MIXER_EN); + hi6210_write_reg(i2s, HII2S_DIG_FILTER_CLK_EN_CFG, val); + + + val = hi6210_read_reg(i2s, HII2S_DIG_FILTER_MODULE_CFG); + val &= ~(HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_IN2_MUTE | + HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_IN2_MUTE); + hi6210_write_reg(i2s, HII2S_DIG_FILTER_MODULE_CFG, val); + + val = hi6210_read_reg(i2s, HII2S_MUX_TOP_MODULE_CFG); + val &= ~(HII2S_MUX_TOP_MODULE_CFG__S2_OL_MIXER_IN1_MUTE | + HII2S_MUX_TOP_MODULE_CFG__S2_OL_MIXER_IN2_MUTE | + HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_MIXER_IN1_MUTE | + HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_MIXER_IN2_MUTE); + hi6210_write_reg(i2s, HII2S_MUX_TOP_MODULE_CFG, val); + + + switch (i2s->format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = false; + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val |= HII2S_I2S_CFG__S2_MST_SLV; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = true; + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val &= ~HII2S_I2S_CFG__S2_MST_SLV; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + break; + default: + WARN_ONCE(1, "Invalid i2s->fmt MASTER_MASK. This shouldn't happen\n"); + } + + switch (i2s->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + fmt = HII2S_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + fmt = HII2S_FORMAT_LEFT_JUST; + break; + case SND_SOC_DAIFMT_RIGHT_J: + fmt = HII2S_FORMAT_RIGHT_JUST; + break; + default: + WARN_ONCE(1, "Invalid i2s->fmt FORMAT_MASK. This shouldn't happen\n"); + } + + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val &= ~(HII2S_I2S_CFG__S2_FUNC_MODE_MASK << + HII2S_I2S_CFG__S2_FUNC_MODE_SHIFT); + val |= fmt << HII2S_I2S_CFG__S2_FUNC_MODE_SHIFT; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + + + val = hi6210_read_reg(i2s, HII2S_CLK_SEL); + val &= ~(HII2S_CLK_SEL__I2S_BT_FM_SEL | /* BT gets the I2S */ + HII2S_CLK_SEL__EXT_12_288MHZ_SEL); + hi6210_write_reg(i2s, HII2S_CLK_SEL, val); + + dma_data->maxburst = 2; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data->addr = i2s->base_phys + HII2S_ST_DL_CHANNEL; + else + dma_data->addr = i2s->base_phys + HII2S_STEREO_UPLINK_CHANNEL; + + switch (i2s->channels) { + case 1: + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val |= HII2S_I2S_CFG__S2_FRAME_MODE; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + break; + default: + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val &= ~HII2S_I2S_CFG__S2_FRAME_MODE; + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + } + + /* clear loopback, set signed type and word length */ + val = hi6210_read_reg(i2s, HII2S_I2S_CFG); + val &= ~HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; + val &= ~(HII2S_I2S_CFG__S2_CODEC_IO_WORDLENGTH_MASK << + HII2S_I2S_CFG__S2_CODEC_IO_WORDLENGTH_SHIFT); + val &= ~(HII2S_I2S_CFG__S2_DIRECT_LOOP_MASK << + HII2S_I2S_CFG__S2_DIRECT_LOOP_SHIFT); + val |= signed_data; + val |= (bits << HII2S_I2S_CFG__S2_CODEC_IO_WORDLENGTH_SHIFT); + hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + + + if (!i2s->master) + return 0; + + /* set DAC and related units to correct rate */ + val = hi6210_read_reg(i2s, HII2S_FS_CFG); + val &= ~(HII2S_FS_CFG__FS_S2_MASK << HII2S_FS_CFG__FS_S2_SHIFT); + val &= ~(HII2S_FS_CFG__FS_DACLR_MASK << HII2S_FS_CFG__FS_DACLR_SHIFT); + val &= ~(HII2S_FS_CFG__FS_ST_DL_R_MASK << + HII2S_FS_CFG__FS_ST_DL_R_SHIFT); + val &= ~(HII2S_FS_CFG__FS_ST_DL_L_MASK << + HII2S_FS_CFG__FS_ST_DL_L_SHIFT); + val |= (rate << HII2S_FS_CFG__FS_S2_SHIFT); + val |= (rate << HII2S_FS_CFG__FS_DACLR_SHIFT); + val |= (rate << HII2S_FS_CFG__FS_ST_DL_R_SHIFT); + val |= (rate << HII2S_FS_CFG__FS_ST_DL_L_SHIFT); + hi6210_write_reg(i2s, HII2S_FS_CFG, val); + + return 0; +} + +static int hi6210_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + pr_debug("%s\n", __func__); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + hi6210_i2s_rxctrl(cpu_dai, 1); + else + hi6210_i2s_txctrl(cpu_dai, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + hi6210_i2s_rxctrl(cpu_dai, 0); + else + hi6210_i2s_txctrl(cpu_dai, 0); + break; + default: + dev_err(cpu_dai->dev, "uknown cmd\n"); + return -EINVAL; + } + return 0; +} + +static int hi6210_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct hi6210_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, + &i2s->dma_data[SNDRV_PCM_STREAM_PLAYBACK], + &i2s->dma_data[SNDRV_PCM_STREAM_CAPTURE]); + + return 0; +} + + +static struct snd_soc_dai_ops hi6210_i2s_dai_ops = { + .trigger = hi6210_i2s_trigger, + .hw_params = hi6210_i2s_hw_params, + .set_fmt = hi6210_i2s_set_fmt, + .startup = hi6210_i2s_startup, + .shutdown = hi6210_i2s_shutdown, +}; + +struct snd_soc_dai_driver hi6210_i2s_dai_init = { + .probe = hi6210_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE, + .rates = SNDRV_PCM_RATE_48000, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE, + .rates = SNDRV_PCM_RATE_48000, + }, + .ops = &hi6210_i2s_dai_ops, +}; + +static const struct snd_soc_component_driver hi6210_i2s_i2s_comp = { + .name = "hi6210_i2s-i2s", +}; + +static int hi6210_i2s_probe(struct platform_device *pdev) +{ + struct device_node *node = pdev->dev.of_node; + struct device *dev = &pdev->dev; + struct hi6210_i2s *i2s; + struct resource *res; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + i2s->dev = dev; + spin_lock_init(&i2s->lock); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(dev, res); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); + + i2s->base_phys = (phys_addr_t)res->start; + i2s->dai = hi6210_i2s_dai_init; + + dev_set_drvdata(&pdev->dev, i2s); + + i2s->sysctrl = syscon_regmap_lookup_by_phandle(node, + "hisilicon,sysctrl-syscon"); + if (IS_ERR(i2s->sysctrl)) + return PTR_ERR(i2s->sysctrl); + + i2s->clk[CLK_DACODEC] = devm_clk_get(&pdev->dev, "dacodec"); + if (IS_ERR_OR_NULL(i2s->clk[CLK_DACODEC])) + return PTR_ERR(i2s->clk[CLK_DACODEC]); + i2s->clocks++; + + i2s->clk[CLK_I2S_BASE] = devm_clk_get(&pdev->dev, "i2s-base"); + if (IS_ERR_OR_NULL(i2s->clk[CLK_I2S_BASE])) + return PTR_ERR(i2s->clk[CLK_I2S_BASE]); + i2s->clocks++; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + return ret; + + ret = snd_soc_register_component(&pdev->dev, &hi6210_i2s_i2s_comp, + &i2s->dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register dai\n"); + return ret; + } + + return 0; +} + +static int hi6210_i2s_remove(struct platform_device *pdev) +{ + snd_soc_unregister_component(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static const struct of_device_id hi6210_i2s_dt_ids[] = { + { .compatible = "hisilicon,hi6210-i2s" }, + { /* sentinel */ } +}; + +MODULE_DEVICE_TABLE(of, hi6210_i2s_dt_ids); + +static struct platform_driver hi6210_i2s_driver = { + .probe = hi6210_i2s_probe, + .remove = hi6210_i2s_remove, + .driver = { + .name = "hi6210_i2s", + .of_match_table = hi6210_i2s_dt_ids, + }, +}; + +module_platform_driver(hi6210_i2s_driver); + +MODULE_DESCRIPTION("Hisilicon HI6210 I2S driver"); +MODULE_AUTHOR("Andy Green <andy.green(a)linaro.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/hisilicon/hi6210-i2s.h b/sound/soc/hisilicon/hi6210-i2s.h new file mode 100644 index 000000000000..85cecc4939a0 --- /dev/null +++ b/sound/soc/hisilicon/hi6210-i2s.h @@ -0,0 +1,276 @@ +/* + * linux/sound/soc/hisilicon/hi6210-i2s.h + * + * Copyright (C) 2015 Linaro, Ltd + * Author: Andy Green <andy.green(a)linaro.org> + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Note at least on 6220, S2 == BT, S1 == Digital FM Radio IF + */ + +#ifndef _HI6210_I2S_H +#define _HI6210_I2S_H + +#define HII2S_SW_RST_N 0 + +#define HII2S_SW_RST_N__STEREO_UPLINK_WORDLEN_SHIFT 28 +#define HII2S_SW_RST_N__STEREO_UPLINK_WORDLEN_MASK 3 +#define HII2S_SW_RST_N__THIRDMD_UPLINK_WORDLEN_SHIFT 26 +#define HII2S_SW_RST_N__THIRDMD_UPLINK_WORDLEN_MASK 3 +#define HII2S_SW_RST_N__VOICE_UPLINK_WORDLEN_SHIFT 24 +#define HII2S_SW_RST_N__VOICE_UPLINK_WORDLEN_MASK 3 +#define HII2S_SW_RST_N__ST_DL_WORDLEN_SHIFT 20 +#define HII2S_SW_RST_N__ST_DL_WORDLEN_MASK 3 +#define HII2S_SW_RST_N__THIRDMD_DLINK_WORDLEN_SHIFT 18 +#define HII2S_SW_RST_N__THIRDMD_DLINK_WORDLEN_MASK 3 +#define HII2S_SW_RST_N__VOICE_DLINK_WORDLEN_SHIFT 16 +#define HII2S_SW_RST_N__VOICE_DLINK_WORDLEN_MASK 3 + +#define HII2S_SW_RST_N__SW_RST_N BIT(0) + +enum hi6210_bits { + HII2S_BITS_16, + HII2S_BITS_18, + HII2S_BITS_20, + HII2S_BITS_24, +}; + + +#define HII2S_IF_CLK_EN_CFG 4 + +#define HII2S_IF_CLK_EN_CFG__THIRDMD_UPLINK_EN BIT(25) +#define HII2S_IF_CLK_EN_CFG__THIRDMD_DLINK_EN BIT(24) +#define HII2S_IF_CLK_EN_CFG__S3_IF_CLK_EN BIT(20) +#define HII2S_IF_CLK_EN_CFG__S2_IF_CLK_EN BIT(16) +#define HII2S_IF_CLK_EN_CFG__S2_OL_MIXER_EN BIT(15) +#define HII2S_IF_CLK_EN_CFG__S2_OL_SRC_EN BIT(14) +#define HII2S_IF_CLK_EN_CFG__S2_IR_PGA_EN BIT(13) +#define HII2S_IF_CLK_EN_CFG__S2_IL_PGA_EN BIT(12) +#define HII2S_IF_CLK_EN_CFG__S1_IR_PGA_EN BIT(10) +#define HII2S_IF_CLK_EN_CFG__S1_IL_PGA_EN BIT(9) +#define HII2S_IF_CLK_EN_CFG__S1_IF_CLK_EN BIT(8) +#define HII2S_IF_CLK_EN_CFG__VOICE_DLINK_SRC_EN BIT(7) +#define HII2S_IF_CLK_EN_CFG__VOICE_DLINK_EN BIT(6) +#define HII2S_IF_CLK_EN_CFG__ST_DL_R_EN BIT(5) +#define HII2S_IF_CLK_EN_CFG__ST_DL_L_EN BIT(4) +#define HII2S_IF_CLK_EN_CFG__VOICE_UPLINK_R_EN BIT(3) +#define HII2S_IF_CLK_EN_CFG__VOICE_UPLINK_L_EN BIT(2) +#define HII2S_IF_CLK_EN_CFG__STEREO_UPLINK_R_EN BIT(1) +#define HII2S_IF_CLK_EN_CFG__STEREO_UPLINK_L_EN BIT(0) + +#define HII2S_DIG_FILTER_CLK_EN_CFG 8 +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACR_SDM_EN BIT(30) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACR_HBF2I_EN BIT(28) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACR_MIXER_EN BIT(25) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACR_AGC_EN BIT(24) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACL_SDM_EN BIT(22) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACL_HBF2I_EN BIT(20) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACL_MIXER_EN BIT(17) +#define HII2S_DIG_FILTER_CLK_EN_CFG__DACL_AGC_EN BIT(16) + +#define HII2S_FS_CFG 0xc + +#define HII2S_FS_CFG__FS_S2_SHIFT 28 +#define HII2S_FS_CFG__FS_S2_MASK 7 +#define HII2S_FS_CFG__FS_S1_SHIFT 24 +#define HII2S_FS_CFG__FS_S1_MASK 7 +#define HII2S_FS_CFG__FS_ADCLR_SHIFT 20 +#define HII2S_FS_CFG__FS_ADCLR_MASK 7 +#define HII2S_FS_CFG__FS_DACLR_SHIFT 16 +#define HII2S_FS_CFG__FS_DACLR_MASK 7 +#define HII2S_FS_CFG__FS_ST_DL_R_SHIFT 8 +#define HII2S_FS_CFG__FS_ST_DL_R_MASK 7 +#define HII2S_FS_CFG__FS_ST_DL_L_SHIFT 4 +#define HII2S_FS_CFG__FS_ST_DL_L_MASK 7 +#define HII2S_FS_CFG__FS_VOICE_DLINK_SHIFT 0 +#define HII2S_FS_CFG__FS_VOICE_DLINK_MASK 7 + +enum hi6210_i2s_rates { + HII2S_FS_RATE_8KHZ = 0, + HII2S_FS_RATE_16KHZ = 1, + HII2S_FS_RATE_32KHZ = 2, + HII2S_FS_RATE_48KHZ = 4, + HII2S_FS_RATE_96KHZ = 5, + HII2S_FS_RATE_192KHZ = 6, +}; + +#define HII2S_I2S_CFG 0x10 + +#define HII2S_I2S_CFG__S2_IF_TX_EN BIT(31) +#define HII2S_I2S_CFG__S2_IF_RX_EN BIT(30) +#define HII2S_I2S_CFG__S2_FRAME_MODE BIT(29) +#define HII2S_I2S_CFG__S2_MST_SLV BIT(28) +#define HII2S_I2S_CFG__S2_LRCK_MODE BIT(27) +#define HII2S_I2S_CFG__S2_CHNNL_MODE BIT(26) +#define HII2S_I2S_CFG__S2_CODEC_IO_WORDLENGTH_SHIFT 24 +#define HII2S_I2S_CFG__S2_CODEC_IO_WORDLENGTH_MASK 3 +#define HII2S_I2S_CFG__S2_DIRECT_LOOP_SHIFT 22 +#define HII2S_I2S_CFG__S2_DIRECT_LOOP_MASK 3 +#define HII2S_I2S_CFG__S2_TX_CLK_SEL BIT(21) +#define HII2S_I2S_CFG__S2_RX_CLK_SEL BIT(20) +#define HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT BIT(19) +#define HII2S_I2S_CFG__S2_FUNC_MODE_SHIFT 16 +#define HII2S_I2S_CFG__S2_FUNC_MODE_MASK 7 +#define HII2S_I2S_CFG__S1_IF_TX_EN BIT(15) +#define HII2S_I2S_CFG__S1_IF_RX_EN BIT(14) +#define HII2S_I2S_CFG__S1_FRAME_MODE BIT(13) +#define HII2S_I2S_CFG__S1_MST_SLV BIT(12) +#define HII2S_I2S_CFG__S1_LRCK_MODE BIT(11) +#define HII2S_I2S_CFG__S1_CHNNL_MODE BIT(10) +#define HII2S_I2S_CFG__S1_CODEC_IO_WORDLENGTH_SHIFT 8 +#define HII2S_I2S_CFG__S1_CODEC_IO_WORDLENGTH_MASK 3 +#define HII2S_I2S_CFG__S1_DIRECT_LOOP_SHIFT 6 +#define HII2S_I2S_CFG__S1_DIRECT_LOOP_MASK 3 +#define HII2S_I2S_CFG__S1_TX_CLK_SEL BIT(5) +#define HII2S_I2S_CFG__S1_RX_CLK_SEL BIT(4) +#define HII2S_I2S_CFG__S1_CODEC_DATA_FORMAT BIT(3) +#define HII2S_I2S_CFG__S1_FUNC_MODE_SHIFT 0 +#define HII2S_I2S_CFG__S1_FUNC_MODE_MASK 7 + +enum hi6210_i2s_formats { + HII2S_FORMAT_I2S, + HII2S_FORMAT_PCM_STD, + HII2S_FORMAT_PCM_USER, + HII2S_FORMAT_LEFT_JUST, + HII2S_FORMAT_RIGHT_JUST, +}; + +#define HII2S_DIG_FILTER_MODULE_CFG 0x14 + +#define HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_GAIN_SHIFT 28 +#define HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_GAIN_MASK 3 +#define HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_IN4_MUTE BIT(27) +#define HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_IN3_MUTE BIT(26) +#define HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_IN2_MUTE BIT(25) +#define HII2S_DIG_FILTER_MODULE_CFG__DACR_MIXER_IN1_MUTE BIT(24) +#define HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_GAIN_SHIFT 20 +#define HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_GAIN_MASK 3 +#define HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_IN4_MUTE BIT(19) +#define HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_IN3_MUTE BIT(18) +#define HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_IN2_MUTE BIT(17) +#define HII2S_DIG_FILTER_MODULE_CFG__DACL_MIXER_IN1_MUTE BIT(16) +#define HII2S_DIG_FILTER_MODULE_CFG__SW_DACR_SDM_DITHER BIT(9) +#define HII2S_DIG_FILTER_MODULE_CFG__SW_DACL_SDM_DITHER BIT(8) +#define HII2S_DIG_FILTER_MODULE_CFG__LM_CODEC_DAC2ADC_SHIFT 4 +#define HII2S_DIG_FILTER_MODULE_CFG__LM_CODEC_DAC2ADC_MASK 7 +#define HII2S_DIG_FILTER_MODULE_CFG__RM_CODEC_DAC2ADC_SHIFT 0 +#define HII2S_DIG_FILTER_MODULE_CFG__RM_CODEC_DAC2ADC_MASK 7 + +enum hi6210_gains { + HII2S_GAIN_100PC, + HII2S_GAIN_50PC, + HII2S_GAIN_25PC, +}; + +#define HII2S_MUX_TOP_MODULE_CFG 0x18 + +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_MIXER_GAIN_SHIFT 14 +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_MIXER_GAIN_MASK 3 +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_MIXER_IN2_MUTE BIT(13) +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_MIXER_IN1_MUTE BIT(12) +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_MIXER_GAIN_SHIFT 10 +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_MIXER_GAIN_MASK 3 +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_MIXER_IN2_MUTE BIT(9) +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_MIXER_IN1_MUTE BIT(8) +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_SRC_RDY BIT(6) +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_SRC_MODE_SHIFT 4 +#define HII2S_MUX_TOP_MODULE_CFG__S2_OL_SRC_MODE_MASK 3 +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_SRC_RDY BIT(3) +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_SRC_MODE_SHIFT 0 +#define HII2S_MUX_TOP_MODULE_CFG__VOICE_DLINK_SRC_MODE_MASK 7 + +enum hi6210_s2_src_mode { + HII2S_S2_SRC_MODE_3, + HII2S_S2_SRC_MODE_12, + HII2S_S2_SRC_MODE_6, + HII2S_S2_SRC_MODE_2, +}; + +enum hi6210_voice_dlink_src_mode { + HII2S_VOICE_DL_SRC_MODE_12 = 1, + HII2S_VOICE_DL_SRC_MODE_6, + HII2S_VOICE_DL_SRC_MODE_2, + HII2S_VOICE_DL_SRC_MODE_3, +}; + +#define HII2S_ADC_PGA_CFG 0x1c +#define HII2S_S1_INPUT_PGA_CFG 0x20 +#define HII2S_S2_INPUT_PGA_CFG 0x24 +#define HII2S_ST_DL_PGA_CFG 0x28 +#define HII2S_VOICE_SIDETONE_DLINK_PGA_CFG 0x2c +#define HII2S_APB_AFIFO_CFG_1 0x30 +#define HII2S_APB_AFIFO_CFG_2 0x34 +#define HII2S_ST_DL_FIFO_TH_CFG 0x38 + +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AEMPTY_SHIFT 24 +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AEMPTY_MASK 0x1f +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AFULL_SHIFT 16 +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_R_AFULL_MASK 0x1f +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AEMPTY_SHIFT 8 +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AEMPTY_MASK 0x1f +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AFULL_SHIFT 0 +#define HII2S_ST_DL_FIFO_TH_CFG__ST_DL_L_AFULL_MASK 0x1f + +#define HII2S_STEREO_UPLINK_FIFO_TH_CFG 0x3c +#define HII2S_VOICE_UPLINK_FIFO_TH_CFG 0x40 +#define HII2S_CODEC_IRQ_MASK 0x44 +#define HII2S_CODEC_IRQ 0x48 +#define HII2S_DACL_AGC_CFG_1 0x4c +#define HII2S_DACL_AGC_CFG_2 0x50 +#define HII2S_DACR_AGC_CFG_1 0x54 +#define HII2S_DACR_AGC_CFG_2 0x58 +#define HII2S_DMIC_SIF_CFG 0x5c +#define HII2S_MISC_CFG 0x60 + +#define HII2S_MISC_CFG__THIRDMD_DLINK_TEST_SEL BIT(17) +#define HII2S_MISC_CFG__THIRDMD_DLINK_DIN_SEL BIT(16) +#define HII2S_MISC_CFG__S3_DOUT_RIGHT_SEL BIT(14) +#define HII2S_MISC_CFG__S3_DOUT_LEFT_SEL BIT(13) +#define HII2S_MISC_CFG__S3_DIN_TEST_SEL BIT(12) +#define HII2S_MISC_CFG__VOICE_DLINK_SRC_UP_DOUT_VLD_SEL BIT(8) +#define HII2S_MISC_CFG__VOICE_DLINK_TEST_SEL BIT(7) +#define HII2S_MISC_CFG__VOICE_DLINK_DIN_SEL BIT(6) +#define HII2S_MISC_CFG__ST_DL_TEST_SEL BIT(4) +#define HII2S_MISC_CFG__S2_DOUT_RIGHT_SEL BIT(3) +#define HII2S_MISC_CFG__S2_DOUT_TEST_SEL BIT(2) +#define HII2S_MISC_CFG__S1_DOUT_TEST_SEL BIT(1) +#define HII2S_MISC_CFG__S2_DOUT_LEFT_SEL BIT(0) + +#define HII2S_S2_SRC_CFG 0x64 +#define HII2S_MEM_CFG 0x68 +#define HII2S_THIRDMD_PCM_PGA_CFG 0x6c +#define HII2S_THIRD_MODEM_FIFO_TH 0x70 +#define HII2S_S3_ANTI_FREQ_JITTER_TX_INC_CNT 0x74 +#define HII2S_S3_ANTI_FREQ_JITTER_TX_DEC_CNT 0x78 +#define HII2S_S3_ANTI_FREQ_JITTER_RX_INC_CNT 0x7c +#define HII2S_S3_ANTI_FREQ_JITTER_RX_DEC_CNT 0x80 +#define HII2S_ANTI_FREQ_JITTER_EN 0x84 +#define HII2S_CLK_SEL 0x88 + +/* 0 = BT owns the i2s */ +#define HII2S_CLK_SEL__I2S_BT_FM_SEL BIT(0) +/* 0 = internal source, 1 = ext */ +#define HII2S_CLK_SEL__EXT_12_288MHZ_SEL BIT(1) + + +#define HII2S_THIRDMD_DLINK_CHANNEL 0xe8 +#define HII2S_THIRDMD_ULINK_CHANNEL 0xec +#define HII2S_VOICE_DLINK_CHANNEL 0xf0 + +/* shovel data in here for playback */ +#define HII2S_ST_DL_CHANNEL 0xf4 +#define HII2S_STEREO_UPLINK_CHANNEL 0xf8 +#define HII2S_VOICE_UPLINK_CHANNEL 0xfc + +#endif/* _HI6210_I2S_H */ -- 2.11.0

1 0

[alsa-devel] [RESEND PATCH] conf/cards: add VC4-HDMI card
by Boris Brezillon 10 Apr '17

10 Apr '17

Add a conf file for the VC4-HDMI sound card. Signed-off-by: Boris Brezillon <boris.brezillon(a)free-electrons.com> --- Hi, Sorry for the noise, but I didn't send this patch to the alsa-lib maintainer on my first attempt. This patch is adding a card config file for the audio sound card whose driver has been submitted here [1] (not accepted yet). Since I am a total newbie to the alsa world, I'd like to get some feedback on this patch. Also, the card only supports 2 to 8 channels, and I wonder if we should add a plug element to support mono streams, and where this element should be added (after or before the iec958 element). Regards, Boris [1]https://www.spinics.net/lists/arm-kernel/msg565182.html src/conf/cards/Makefile.am | 1 + src/conf/cards/aliases.conf | 1 + src/conf/cards/vc4-hdmi.conf | 64 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 66 insertions(+) create mode 100644 src/conf/cards/vc4-hdmi.conf diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index e8b530e80a8f..00999f0186d6 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -51,6 +51,7 @@ cfg_files = aliases.conf \ TRID4DWAVENX.conf \ USB-Audio.conf \ YMF744.conf \ + vc4-hdmi.conf \ VIA686A.conf \ VIA8233.conf \ VIA8233A.conf \ diff --git a/src/conf/cards/aliases.conf b/src/conf/cards/aliases.conf index 60f9d26974fd..18a920f41e46 100644 --- a/src/conf/cards/aliases.conf +++ b/src/conf/cards/aliases.conf @@ -56,6 +56,7 @@ AV200 cards.CMI8788 CMI8786 cards.CMI8788 CMI8787 cards.CMI8788 pistachio cards.pistachio-card +VC4-HDMI cards.vc4-hdmi <confdir:pcm/default.conf> <confdir:pcm/dmix.conf> diff --git a/src/conf/cards/vc4-hdmi.conf b/src/conf/cards/vc4-hdmi.conf new file mode 100644 index 000000000000..027804a145d1 --- /dev/null +++ b/src/conf/cards/vc4-hdmi.conf @@ -0,0 +1,64 @@ +# +# Configuration for the VC4-HDMI sound card using software IEC958 +# subframe conversion +# + +<confdir:pcm/front.conf> + +vc4-hdmi.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD +} + +# default with dmix +vc4-hdmi.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + @func concat + strings [ "dmix:" $CARD ] + } + } +} + +<confdir:pcm/iec958.conf> + +vc4-hdmi.pcm.iec958.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type iec958 + slave { + format IEC958_SUBFRAME_LE + pcm { + type plug + slave.pcm { + type hw + card $CARD + } + } + } + status [ $AES0 $AES1 $AES2 $AES3 ] +} -- 2.7.4

3 9

[alsa-devel] [PATCH] ALSA: pcm: dmix: Disable var_periodsize as default
by Takashi Iwai 10 Apr '17

10 Apr '17

The recently added variable period-size feature for dmix & co seems causing a regression on some old applications; e.g. Audacious aborts with an error: ALSA error: snd_pcm_hw_params_set_buffer_time_near failed: Invalid argument. As a quick fix, disable the feature as default. One can still enable it via asoundrc. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=1033179 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 8a75c421dd7b..89468d3537d0 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1875,7 +1875,7 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->ipc_gid = -1; rec->slowptr = 1; rec->max_periods = 0; - rec->var_periodsize = 1; + rec->var_periodsize = 0; rec->direct_memory_access = 1; /* read defaults */ -- 2.11.1

1 0

[alsa-devel] [PATCH] [alsa-lib] Removed compilation warning from test/namehint.c
by soodvarun78＠gmail.com 10 Apr '17

10 Apr '17

From: varun <soodvarun78(a)gmail.com> Changed hint variable from char ** to void ** to match function snd_device_name_hint Signed-off-by: varun <soodvarun78(a)gmail.com> --- test/namehint.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/test/namehint.c b/test/namehint.c index e978d5c..18bad1d 100644 --- a/test/namehint.c +++ b/test/namehint.c @@ -4,7 +4,8 @@ int main(int argc, char *argv[]) { const char *iface = "pcm"; - char **hints, **n; + void **hints; + char **n; int err; if (argc > 1) @@ -12,7 +13,7 @@ int main(int argc, char *argv[]) err = snd_device_name_hint(-1, iface, &hints); if (err < 0) errx(1, "snd_device_name_hint error: %s", snd_strerror(err)); - n = hints; + n = (char **)hints; while (*n != NULL) { printf("%s\n", *n); n++; -- 1.9.1

2 1

[alsa-devel] [PATCH v2] aplay: Introduce and use xwrite helper
by Daniel Baluta 10 Apr '17

10 Apr '17

Write can return less then requested bytes, but we treat this as an error thus ending up with confusing error messages. Fix this by introducing xwrite helper, which makes sure all bytes are written or an error is returned. With this patch an usecase where disk is filled by recording will print: $ /mnt/msc/audio.wav: No space left on device instead of random messages like: $/mnt/msc/audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: * return the correct number of bytes aplay/aplay.c | 55 +++++++++++++++++++++++++++++++++++++------------------ 1 file changed, 37 insertions(+), 18 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..f793c82 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -429,6 +429,25 @@ enum { OPT_FATAL_ERRORS, }; +/* + * make sure we write all bytes or return an error + */ +static ssize_t xwrite(int fd, const void *buf, size_t count) +{ + ssize_t written; + size_t offset = 0; + + while (offset < count) { + written = write(fd, buf + offset, count - offset); + if (written <= 0) + return written; + + offset += written; + }; + + return offset; +} + static long parse_long(const char *str, int *err) { long val; @@ -2428,7 +2447,7 @@ static void voc_play(int fd, int ofs, char *name) l = nextblock; if (l) { if (output && !quiet_mode) { - if (write(2, data, l) != l) { /* to stderr */ + if (xwrite(2, data, l) != l) { /* to stderr */ error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2480,7 +2499,7 @@ static void begin_voc(int fd, size_t cnt) vh.version = LE_SHORT(VOC_ACTUAL_VERSION); vh.coded_ver = LE_SHORT(0x1233 - VOC_ACTUAL_VERSION); - if (write(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { + if (xwrite(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2489,14 +2508,14 @@ static void begin_voc(int fd, size_t cnt) bt.type = 8; bt.datalen = 4; bt.datalen_m = bt.datalen_h = 0; - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } eb.tc = LE_SHORT(65536 - 256000000L / (hwparams.rate << 1)); eb.pack = 0; eb.mode = 1; - if (write(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { + if (xwrite(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2506,13 +2525,13 @@ static void begin_voc(int fd, size_t cnt) bt.datalen = (u_char) (cnt & 0xFF); bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } vd.tc = (u_char) (256 - (1000000 / hwparams.rate)); vd.pack = 0; - if (write(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { + if (xwrite(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2581,10 +2600,10 @@ static void begin_wave(int fd, size_t cnt) cd.type = WAV_DATA; cd.length = LE_INT(cnt); - if (write(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || - write(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || - write(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || - write(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { + if (xwrite(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || + xwrite(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || + xwrite(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || + xwrite(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2614,7 +2633,7 @@ static void begin_au(int fd, size_t cnt) } ah.sample_rate = BE_INT(hwparams.rate); ah.channels = BE_INT(hwparams.channels); - if (write(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { + if (xwrite(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2628,7 +2647,7 @@ static void end_voc(int fd) size_t cnt; char dummy = 0; /* Write a Terminator */ - if (write(fd, &dummy, 1) != 1) { + if (xwrite(fd, &dummy, 1) != 1) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2644,7 +2663,7 @@ static void end_voc(int fd) bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &bt, sizeof(VocBlockType)); + xwrite(fd, &bt, sizeof(VocBlockType)); if (fd != 1) close(fd); } @@ -2664,9 +2683,9 @@ static void end_wave(int fd) filelen = fdcount + 2*sizeof(WaveChunkHeader) + sizeof(WaveFmtBody) + 4; rifflen = filelen > 0x7fffffff ? LE_INT(0x7fffffff) : LE_INT(filelen); if (lseek64(fd, 4, SEEK_SET) == 4) - write(fd, &rifflen, 4); + xwrite(fd, &rifflen, 4); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &cd, sizeof(WaveChunkHeader)); + xwrite(fd, &cd, sizeof(WaveChunkHeader)); if (fd != 1) close(fd); } @@ -2679,7 +2698,7 @@ static void end_au(int fd) length_seek = (char *)&ah.data_size - (char *)&ah; ah.data_size = fdcount > 0xffffffff ? 0xffffffff : BE_INT(fdcount); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &ah.data_size, sizeof(ah.data_size)); + xwrite(fd, &ah.data_size, sizeof(ah.data_size)); if (fd != 1) close(fd); } @@ -3078,7 +3097,7 @@ static void capture(char *orig_name) in_aborting = 1; break; } - if (write(fd, audiobuf, c) != c) { + if (xwrite(fd, audiobuf, c) != c) { perror(name); in_aborting = 1; break; @@ -3187,7 +3206,7 @@ static void capturev_go(int* fds, unsigned int channels, off64_t count, int rtyp break; rv = r * bits_per_sample / 8; for (channel = 0; channel < channels; ++channel) { - if ((size_t)write(fds[channel], bufs[channel], rv) != rv) { + if ((size_t)xwrite(fds[channel], bufs[channel], rv) != rv) { perror(names[channel]); prg_exit(EXIT_FAILURE); } -- 2.7.4

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Re: [alsa-devel] DSD native format on SoC
by Michael Trimarchi 09 Apr '17

09 Apr '17

Hi all On Mon, Mar 13, 2017 at 05:42:43PM +0900, Naoki Matsumoto wrote: > Retry send because alsa-dev ML doesn't delivery yet. > > -------- Original Message -------- > From: Naoki Matsumoto > Sent: Monday, Mar 13, 2017 10:39 AM GMT+0900 > To: michael(a)amarulasolutions.com > Cc: alsa-devel(a)alsa-project.org > Subject: DSD native format on SoC > > Hello > > I know only xmos-native-dsd(DSD_U32_BE). > I'll share what I know. > I know information about DSD only fragmentally... > Problem here is SoC subsytem for me https://github.com/zonque/alsa-dsd-player.git Cannot set sample format tyring U16_LE DSD. I have changed a bit the core and some components but seems that does not go in. Any idea? diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c index cc34b16..0dea529 100644 --- a/sound/soc/codecs/pcm179x.c +++ b/sound/soc/codecs/pcm179x.c @@ -89,18 +89,19 @@ static int pcm179x_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct pcm179x_private *priv = snd_soc_codec_get_drvdata(codec); - u64 formats = PCM1792A_FORMATS; + u64 formats = PCM1795_FORMATS; switch (priv->codec_model) { - case PCM1795: - formats = PCM1795_FORMATS; + case PCM1792A: + formats = PCM1792A_FORMATS; break; default: break; } - snd_pcm_hw_constraint_mask64(substream->runtime, - SNDRV_PCM_HW_PARAM_FORMAT, formats); + if (formats != PCM1795_FORMATS) + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, formats); msleep(50); return 0; @@ -227,7 +228,7 @@ static struct snd_soc_dai_driver pcm179x_dai = { .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 10000, .rate_max = 200000, - .formats = PCM179X_FORMATS, }, + .formats = PCM1795_FORMATS, }, .ops = &pcm179x_dai_ops, }; @@ -252,9 +253,9 @@ static struct snd_soc_codec_driver soc_codec_dev_pcm179x = { }; const struct of_device_id pcm179x_of_match[] = { - { .compatible = "ti,pcm1792a", }, - { .compatible = "ti,pcm1795", .data = (void *)PCM1795, }, - { .compatible = "ti,pcm1796", }, + { .compatible = "ti,pcm1792a", .data = (void *)PCM1792A }, + { .compatible = "ti,pcm1795", }, + { .compatible = "ti,pcm1796", .data = (void *)PCM1792A }, { } }; MODULE_DEVICE_TABLE(of, pcm179x_of_match); diff --git a/sound/soc/codecs/pcm179x.h b/sound/soc/codecs/pcm179x.h index 4c00047..0665ec8 100644 --- a/sound/soc/codecs/pcm179x.h +++ b/sound/soc/codecs/pcm179x.h @@ -22,7 +22,8 @@ #define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -#define PCM1795_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define PCM1795_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_DSD_U16_LE) extern const struct regmap_config pcm179x_regmap_config; extern const struct of_device_id pcm179x_of_match[]; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index c55cbfa..d3afa89 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -55,7 +55,7 @@ config SND_SOC_ROCHCHIP_DACMAX tristate "SoC Audio support for DACMAX boards using a pcm1792 codec" depends on SND_SOC_ROCKCHIP && SPI && GPIOLIB select SND_SOC_ROCKCHIP_I2S - select SND_SOC_PCM179X + select SND_SOC_PCM179X_SPI help Say Y or M here if you want to add support for SoC audio on Dacmax boards using the pcm1792a codec. diff --git a/sound/soc/rockchip/dacmax.c b/sound/soc/rockchip/dacmax.c index fb302a8..fa79549 100644 --- a/sound/soc/rockchip/dacmax.c +++ b/sound/soc/rockchip/dacmax.c @@ -22,6 +22,7 @@ * */ +#define DEBUG #include <linux/module.h> #include <linux/of.h> #include <linux/of_gpio.h> @@ -39,6 +40,7 @@ #define CLK1 (1 << 1) #define CLK0 (1 << 2) #define W32 (1 << 3) +#define DSD_EN (1 << 4) #define DAI_NAME_SIZE 32 @@ -52,6 +54,7 @@ struct dacmax_data { int clk_1; int clk_2; int w32; + int dsd_enable; }; static const struct snd_soc_dapm_widget dacmax_dapm_widgets[] = { @@ -78,6 +81,10 @@ static void dacmax_change_freq(struct dacmax_data *data, u8 mask) gpio_set_value(data->w32, value); pr_debug("%s: BITSXWORD(%d)\n", __func__, value); + value = (mask & DSD_EN) ? 1 : 0; + gpio_set_value(data->dsd_enable, value); + pr_debug("%s: DSD ENABLE (%d)\n", __func__, value); + mdelay(20); } @@ -91,6 +98,7 @@ static int dacmax_ext_clock_update(struct dacmax_data *data, params_format(params)); switch (params_format(params)) { + case SNDRV_PCM_FORMAT_DSD_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S24_LE: @@ -103,6 +111,8 @@ static int dacmax_ext_clock_update(struct dacmax_data *data, } switch (params_rate(params)) { + case 2822400: + mask |= DSD_EN; case 44100: break; case 48000: @@ -251,6 +261,7 @@ static int dacmax_probe(struct platform_device *pdev) data->clk_1 = clk_1; data->clk_2 = clk_2; data->w32 = w32; + data->dsd_enable = dsd_enable; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 68fea0a..67fd6ec 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -302,6 +302,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S8: val |= I2S_TXCR_VDW(8); break; + case SNDRV_PCM_FORMAT_DSD_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: val |= I2S_TXCR_VDW(16); break; @@ -457,7 +458,8 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_DSD_U16_LE), }, .capture = { .stream_name = "Capture", diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a1305f8..1ae0aea 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2980,6 +2980,7 @@ static u64 codec_format_map[] = { SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE, SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE, SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE, + SNDRV_PCM_FORMAT_DSD_U16_LE | SNDRV_PCM_FORMAT_DSD_U16_BE, SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, }; > on Wed, 8 Mar 2017 08:21:23 +0100, Michael Nazzareno Trimarchi wrote: > DSD is a continuous stream of bits that flows over two separate > channels, left and right, synchronized by a clock, while I2S is a > single data wire and an additional wire that states if the incoming > sample refers to the left channel or the right channel. There is no > way to get DSD data other than a circuit which decode DSD streams > coming directly from a conventional source, like, for examples, a > SACD. > > The Volta dac has an internal decoding circuitry that still employs > the I2S standard as a usual 44100Hz/16 bits, but sourcing the shift > clock at twice the DSD standard frequency of 2.8224MHz, since it has > to split two the16 bits words received on a single wire into the two > DSD channels, as the standard wants. > > When DSD is enabled, the control signal PCD/DSD must be high, CK3..CK0 > must be all low. The I2S interface works as a standard 44100Hz/16 bit > and the DSD streaming must be packed into 16 bits lenght words > left/right as per LRCK logic. The BCLK frequency supplied from the > interface is 5644800Hz in case of DSD and 11289600 for DSD2. > I don't know your device. > but I've understood that it's device layer implantation topic. > I think that your sound device need to support SND_PCM_FORMAT_DSD_*. > > Refer:enum snd_pcm_format_t > http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > > If you use xmos-natvie-dsd, the implement is linux.git/sound/usb/quirks.c > > Can you explain how now DSD can be pass to audio card? The idea is to > declare it on the audio card and then configure it as a PCM card but I > don't know how data are suppose to arrive from alsa userspace. > I know DSD playback on sound device that xmos. > Have you checked this page? > https://github.com/lintweaker/xmos-native-dsd > > As alsa-lib > 1. snd_pcm_open > e.g, USB sound device. In your case, soc internal sound device > 2. snd_ocm_hw_params_set_format > e.g, SND_PCM_FORMAT_DSD_U32_BE > 3. snd_pcm_writei > > Other elements > * DSD decoder(e.g,dsf/dsdiff) > * Player > if you use xmos-native-dsd(DSD_U32_BE) device, we can use MPD Ver0.20.2+ > I don't know other DSD pcm_formats. > > Just information. It may be wrong. > Thank you > > > -- > ********************************************** > Naoki MATSUMOTO > Email:n-matsumoto@melcoinc.co.jp > Tel :050-5830-8916 > ********************************************** -- | Michael Nazzareno Trimarchi Amarula Solutions BV | | COO - Founder Cruquiuskade 47 | | +31(0)851119172 Amsterdam 1018 AM NL | | [`as] http://www.amarulasolutions.com |

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[alsa-devel] [alsa-lib] test/namehint.c: Removed compilation warning
by varun sood 09 Apr '17

09 Apr '17

Removed compilation warning Changed hint variable from char ** to void ** to match function snd_device_name_hint Signed-off-by: varun <soodvarun78(a)gmail.com>--- test/namehint.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/test/namehint.c b/test/namehint.cindex e978d5c..18bad1d 100644--- a/test/namehint.c+++ b/test/namehint.c@@ -4,7 +4,8 @@ int main(int argc, char *argv[]) { const char *iface = "pcm";- char **hints, **n;+ void **hints;+ char **n; int err; if (argc > 1)@@ -12,7 +13,7 @@ int main(int argc, char *argv[]) err = snd_device_name_hint(-1, iface, &hints); if (err < 0) errx(1, "snd_device_name_hint error: %s", snd_strerror(err));- n = hints;+ n = (char **)hints; while (*n != NULL) { printf("%s\n", *n); n++;-- 1.9.1

2 1

[alsa-devel] [PATCH] aplay: Introduce and use xwrite helper
by Daniel Baluta 07 Apr '17

07 Apr '17

Write can return less then requested bytes, but we treat this as an error thus ending up with confusing error messages. Fix this by introducing xwrite helper, which makes sure all bytes are written or an error is returned. With this patch an usecase where disk is filled by recording will print: $ /mnt/msc/audio.wav: No space left on device instead of random messages like: $/mnt/msc/audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- This is a follow up of: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119632.html where I tried to fix the error message instead of write :). aplay/aplay.c | 55 +++++++++++++++++++++++++++++++++++++------------------ 1 file changed, 37 insertions(+), 18 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..74d3d37 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -429,6 +429,25 @@ enum { OPT_FATAL_ERRORS, }; +/* + * make sure we write all bytes or return an error + */ +static ssize_t xwrite(int fd, const void *buf, size_t count) +{ + ssize_t written; + size_t offset = 0; + + while (offset < count) { + written = write(fd, buf + offset, count - offset); + if (written <= 0) + return written; + + offset += written; + }; + + return written; +} + static long parse_long(const char *str, int *err) { long val; @@ -2428,7 +2447,7 @@ static void voc_play(int fd, int ofs, char *name) l = nextblock; if (l) { if (output && !quiet_mode) { - if (write(2, data, l) != l) { /* to stderr */ + if (xwrite(2, data, l) != l) { /* to stderr */ error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2480,7 +2499,7 @@ static void begin_voc(int fd, size_t cnt) vh.version = LE_SHORT(VOC_ACTUAL_VERSION); vh.coded_ver = LE_SHORT(0x1233 - VOC_ACTUAL_VERSION); - if (write(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { + if (xwrite(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2489,14 +2508,14 @@ static void begin_voc(int fd, size_t cnt) bt.type = 8; bt.datalen = 4; bt.datalen_m = bt.datalen_h = 0; - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } eb.tc = LE_SHORT(65536 - 256000000L / (hwparams.rate << 1)); eb.pack = 0; eb.mode = 1; - if (write(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { + if (xwrite(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2506,13 +2525,13 @@ static void begin_voc(int fd, size_t cnt) bt.datalen = (u_char) (cnt & 0xFF); bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } vd.tc = (u_char) (256 - (1000000 / hwparams.rate)); vd.pack = 0; - if (write(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { + if (xwrite(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2581,10 +2600,10 @@ static void begin_wave(int fd, size_t cnt) cd.type = WAV_DATA; cd.length = LE_INT(cnt); - if (write(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || - write(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || - write(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || - write(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { + if (xwrite(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || + xwrite(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || + xwrite(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || + xwrite(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2614,7 +2633,7 @@ static void begin_au(int fd, size_t cnt) } ah.sample_rate = BE_INT(hwparams.rate); ah.channels = BE_INT(hwparams.channels); - if (write(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { + if (xwrite(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2628,7 +2647,7 @@ static void end_voc(int fd) size_t cnt; char dummy = 0; /* Write a Terminator */ - if (write(fd, &dummy, 1) != 1) { + if (xwrite(fd, &dummy, 1) != 1) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2644,7 +2663,7 @@ static void end_voc(int fd) bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &bt, sizeof(VocBlockType)); + xwrite(fd, &bt, sizeof(VocBlockType)); if (fd != 1) close(fd); } @@ -2664,9 +2683,9 @@ static void end_wave(int fd) filelen = fdcount + 2*sizeof(WaveChunkHeader) + sizeof(WaveFmtBody) + 4; rifflen = filelen > 0x7fffffff ? LE_INT(0x7fffffff) : LE_INT(filelen); if (lseek64(fd, 4, SEEK_SET) == 4) - write(fd, &rifflen, 4); + xwrite(fd, &rifflen, 4); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &cd, sizeof(WaveChunkHeader)); + xwrite(fd, &cd, sizeof(WaveChunkHeader)); if (fd != 1) close(fd); } @@ -2679,7 +2698,7 @@ static void end_au(int fd) length_seek = (char *)&ah.data_size - (char *)&ah; ah.data_size = fdcount > 0xffffffff ? 0xffffffff : BE_INT(fdcount); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &ah.data_size, sizeof(ah.data_size)); + xwrite(fd, &ah.data_size, sizeof(ah.data_size)); if (fd != 1) close(fd); } @@ -3078,7 +3097,7 @@ static void capture(char *orig_name) in_aborting = 1; break; } - if (write(fd, audiobuf, c) != c) { + if (xwrite(fd, audiobuf, c) != c) { perror(name); in_aborting = 1; break; @@ -3187,7 +3206,7 @@ static void capturev_go(int* fds, unsigned int channels, off64_t count, int rtyp break; rv = r * bits_per_sample / 8; for (channel = 0; channel < channels; ++channel) { - if ((size_t)write(fds[channel], bufs[channel], rv) != rv) { + if ((size_t)xwrite(fds[channel], bufs[channel], rv) != rv) { perror(names[channel]); prg_exit(EXIT_FAILURE); } -- 2.7.4

2 1

[alsa-devel] [PATCH v2] aplay: Fix error message when writing captured data
by Daniel Baluta 07 Apr '17

07 Apr '17

Read can return less then requested bytes, but we treat this as an error. Anyhow, errno is not updated in this case and we can end up with a confusing error message. For example, when there is no room to write data into the output file we receive: $ arecord -d 2000 -c 2 -r 192000 -f S16_LE -Dplughw:0,0 audio.wav Recording WAVE '/mnt/msc/audio.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Stereo audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..96f5e1b 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3079,7 +3079,7 @@ static void capture(char *orig_name) break; } if (write(fd, audiobuf, c) != c) { - perror(name); + fprintf(stderr, _("Couldn't write all data to %s\n"), name); in_aborting = 1; break; } -- 2.7.4

3 4

[alsa-devel] [PATCH] aplay: Fix error message when writing captured data
by Daniel Baluta 07 Apr '17

07 Apr '17

Write can return less then requested bytes, but we treat this as an error. Anyhow, errno is not updated in this case and we can end up with a confusing error message. For example, when there is no room to write data into the output file we receive: $ arecord -d 2000 -c 2 -r 192000 -f S16_LE -Dplughw:0,0 audio.wav Recording WAVE '/mnt/msc/audio.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Stereo audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..9547d1b 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3079,7 +3079,7 @@ static void capture(char *orig_name) break; } if (write(fd, audiobuf, c) != c) { - perror(name); + fprintf(stderr, "Couldn't write all data to %s\n", name); in_aborting = 1; break; } -- 2.7.4

2 2

[alsa-devel] [PATCH 35/38] Annotate hardware config module parameters in sound/drivers/
by David Howells 07 Apr '17

07 Apr '17

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/drivers/. Suggested-by: Alan Cox <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/drivers/mpu401/mpu401.c | 4 ++-- sound/drivers/mtpav.c | 4 ++-- sound/drivers/serial-u16550.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index fed7e7e2177b..9b86e00d7d95 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -53,9 +53,9 @@ MODULE_PARM_DESC(enable, "Enable MPU-401 device."); module_param_array(pnp, bool, NULL, 0444); MODULE_PARM_DESC(pnp, "PnP detection for MPU-401 device."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for MPU-401 device."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for MPU-401 device."); module_param_array(uart_enter, bool, NULL, 0444); MODULE_PARM_DESC(uart_enter, "Issue UART_ENTER command at open."); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 00b31f92c504..0f6392001e30 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -86,9 +86,9 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for MotuMTPAV MIDI."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for MotuMTPAV MIDI."); -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "Parallel port # for MotuMTPAV MIDI."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "Parallel IRQ # for MotuMTPAV MIDI."); module_param(hwports, int, 0444); MODULE_PARM_DESC(hwports, "Hardware ports # for MotuMTPAV MIDI."); diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 60d51ac4ccfe..88e66ea0306d 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -84,9 +84,9 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Serial MIDI."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable UART16550A chip."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for UART16550A chip."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for UART16550A chip."); module_param_array(speed, int, NULL, 0444); MODULE_PARM_DESC(speed, "Speed in bauds.");

2 1

[alsa-devel] [PATCH 1/2 v2] ALSA: hda - add more ML register definitions
by libin.yang＠intel.com 07 Apr '17

07 Apr '17

From: Libin Yang <libin.yang(a)intel.com> This patch refines the definition of AZX_MLCTL_SPA and AZX_MLCTL_CPA and add more definitions of ML registers Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/hda_register.h | 8 +++++--- sound/hda/ext/hdac_ext_controller.c | 6 +++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0013063..8df12e4 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -243,9 +243,11 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 -#define AZX_MLCTL_SPA (1<<16) -#define AZX_MLCTL_CPA 23 - +#define ML_LCTL_SCF_MASK 0xF +#define AZX_MLCTL_SPA (0x1 << 16) +#define AZX_MLCTL_CPA (0x1 << 23) +#define AZX_MLCTL_SPA_SHIFT 16 +#define AZX_MLCTL_CPA_SHIFT 23 /* registers for DMA Resume Capability Structure */ #define AZX_DRSM_CAP_ID 0x5 diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 2614691..84f3b81 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -171,7 +171,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) { int timeout; u32 val; - int mask = (1 << AZX_MLCTL_CPA); + int mask = (1 << AZX_MLCTL_CPA_SHIFT); udelay(3); timeout = 150; @@ -179,10 +179,10 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) do { val = readl(link->ml_addr + AZX_REG_ML_LCTL); if (enable) { - if (((val & mask) >> AZX_MLCTL_CPA)) + if (((val & mask) >> AZX_MLCTL_CPA_SHIFT)) return 0; } else { - if (!((val & mask) >> AZX_MLCTL_CPA)) + if (!((val & mask) >> AZX_MLCTL_CPA_SHIFT)) return 0; } udelay(3); -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: es8328: Remove extra blank lines
by Narcisa Ana Maria Vasile 07 Apr '17

07 Apr '17

This was reported by checkpatch.pl Signed-off-by: Narcisa Ana Maria Vasile <narcisaanamaria12(a)gmail.com> --- sound/soc/codecs/es8328.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f84fbd..1967157 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -177,8 +177,6 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, return 0; } - - static const struct snd_kcontrol_new es8328_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Digital Volume", ES8328_ADCCONTROL8, ES8328_ADCCONTROL9, -- 1.9.1

2 2

[alsa-devel] [PATCH v2 1/8] regulator: arizona-micsupp: Avoid potential memory leak reading init_data
by Charles Keepax 06 Apr '17

06 Apr '17

The device argument passed to of_get_regulator_init_data is used to do some devres memory allocation. Currently the driver passes the MFD device pointer to this function, this could result in the init_data allocation being leaked if the regulator is unbound but the MFD isn't. Correct this issue by correctly passing the local platform device. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- New patch in the series since v1. drivers/regulator/arizona-micsupp.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/drivers/regulator/arizona-micsupp.c b/drivers/regulator/arizona-micsupp.c index 22bd714..5e38861 100644 --- a/drivers/regulator/arizona-micsupp.c +++ b/drivers/regulator/arizona-micsupp.c @@ -199,7 +199,8 @@ static const struct regulator_init_data arizona_micsupp_ext_default = { .num_consumer_supplies = 1, }; -static int arizona_micsupp_of_get_pdata(struct arizona *arizona, +static int arizona_micsupp_of_get_pdata(struct device *dev, + struct arizona *arizona, struct regulator_config *config, const struct regulator_desc *desc) { @@ -213,7 +214,7 @@ static int arizona_micsupp_of_get_pdata(struct arizona *arizona, if (np) { config->of_node = np; - init_data = of_get_regulator_init_data(arizona->dev, np, desc); + init_data = of_get_regulator_init_data(dev, np, desc); if (init_data) { init_data->consumer_supplies = &micsupp->supply; @@ -268,8 +269,8 @@ static int arizona_micsupp_probe(struct platform_device *pdev) if (IS_ENABLED(CONFIG_OF)) { if (!dev_get_platdata(arizona->dev)) { - ret = arizona_micsupp_of_get_pdata(arizona, &config, - desc); + ret = arizona_micsupp_of_get_pdata(&pdev->dev, arizona, + &config, desc); if (ret < 0) return ret; } -- 2.1.4

3 15

[alsa-devel] [PATCH 0/2] ASoC: codec: wm8960: Relax bit clock computation when using PLL
by Daniel Baluta 06 Apr '17

06 Apr '17

This is a follow up of patch series sent here: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119517.html where only patch 1/2 was merged. In this new patch series, we fix a problem with patch already merged and then we relax bit clock computation when using PLL the same as we did in patch 2/2 of previous series. Daniel Baluta (2): ASoC: codec: wm8960: Stop when a matching PLL freq is found ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) -- 2.7.4

2 3

[alsa-devel] [PATCH 1/3] ASoC: cs35l35: Improve power down time
by Charles Keepax 06 Apr '17

06 Apr '17

Shorten the time it takes to power down the amp by disabling the volume ramp whilst doing the final shutdown. The driver has already muted the amplifier at this stage so doing the volume ramp serves no purpose. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 8 ++++++++ sound/soc/codecs/cs35l35.h | 3 +++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 9688274..1db07a6 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -187,6 +187,10 @@ static int cs35l35_sdin_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, CS35L35_PDN_ALL_MASK, 1); + /* Already muted, so disable volume ramp for faster shutdown */ + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_DIG_VOL_CTL, + CS35L35_AMP_DIGSFT_MASK, 0); + reinit_completion(&cs35l35->pdn_done); ret = wait_for_completion_timeout(&cs35l35->pdn_done, @@ -199,6 +203,10 @@ static int cs35l35_sdin_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MCLK_DIS_MASK, 1 << CS35L35_MCLK_DIS_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_DIG_VOL_CTL, + CS35L35_AMP_DIGSFT_MASK, + 1 << CS35L35_AMP_DIGSFT_SHIFT); break; default: dev_err(codec->dev, "Invalid event = 0x%x\n", event); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index 156d2f0..54e9ac5 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -190,6 +190,9 @@ #define CS35L35_AMP_GAIN_ZC_MASK 0x10 #define CS35L35_AMP_GAIN_ZC_SHIFT 4 +#define CS35L35_AMP_DIGSFT_MASK 0x02 +#define CS35L35_AMP_DIGSFT_SHIFT 1 + /* CS35L35_SP_FMT_CTL3 */ #define CS35L35_SP_I2S_DRV_MASK 0x03 #define CS35L35_SP_I2S_DRV_SHIFT 0 -- 2.1.4

3 6

[alsa-devel] [PATCH v3] ASoC: STI: Fix null ptr deference in IRQ handler
by Arnaud Pouliquen 06 Apr '17

06 Apr '17

With RTlinux a race condition has been found that leads to NULL ptr crash: - On CPU 0: uni_player_irq_handler is called to treat XRUN "(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked, dev_err(player->dev, "FIFO underflow error detected") is printed and then snd_pcm_stream_lock should be called to lock stream for stopping. - On CPU 1: application stop and close the stream. Issue is that the stop and shutdown functions are executed while "FIFO underflow error detected" is printed. So when CPU 0 calls snd_pcm_stream_lock, player->substream is already null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- V3: - Lower case for labels, - Fix lock-unlock balance. V2: Add spinlock to protect player/reader->substream --- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 35 ++++++++++++++++++++++++----------- sound/soc/sti/uniperif_reader.c | 24 ++++++++++++++++++++---- 3 files changed, 45 insertions(+), 15 deletions(-) diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index d487dd2..cfcb0ea 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1299,6 +1299,7 @@ struct uniperif { int ver; /* IP version, used by register access macros */ struct regmap_field *clk_sel; struct regmap_field *valid_sel; + spinlock_t irq_lock; /* use to prevent race condition with IRQ */ /* capabilities */ const struct snd_pcm_hardware *hw; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 60ae31a..d7e8dd4 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -65,10 +65,13 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) unsigned int status; unsigned int tmp; - if (player->state == UNIPERIF_STATE_STOPPED) { - /* Unexpected IRQ: do nothing */ - return IRQ_NONE; - } + spin_lock(&player->irq_lock); + if (!player->substream) + goto irq_spin_unlock; + + snd_pcm_stream_lock(player->substream); + if (player->state == UNIPERIF_STATE_STOPPED) + goto stream_unlock; /* Get interrupt status & clear them immediately */ status = GET_UNIPERIF_ITS(player); @@ -88,9 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); } ret = IRQ_HANDLED; @@ -104,9 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } @@ -116,7 +115,8 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) if (!player->underflow_enabled) { dev_err(player->dev, "unexpected Underflow recovering\n"); - return -EPERM; + ret = -EPERM; + goto stream_unlock; } /* Read the underflow recovery duration */ tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player); @@ -138,13 +138,16 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } +stream_unlock: + snd_pcm_stream_unlock(player->substream); +irq_spin_unlock: + spin_unlock(&player->irq_lock); + return ret; } @@ -588,6 +591,7 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; struct snd_aes_iec958 *iec958 = &player->stream_settings.iec958; + unsigned long flags; mutex_lock(&player->ctrl_lock); iec958->status[0] = ucontrol->value.iec958.status[0]; @@ -596,12 +600,14 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[3] = ucontrol->value.iec958.status[3]; mutex_unlock(&player->ctrl_lock); + spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) uni_player_set_channel_status(player, player->substream->runtime); else uni_player_set_channel_status(player, NULL); + spin_unlock_irqrestore(&player->irq_lock, flags); return 0; } @@ -686,9 +692,12 @@ static int uni_player_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + unsigned long flags; int ret; + spin_lock_irqsave(&player->irq_lock, flags); player->substream = substream; + spin_unlock_irqrestore(&player->irq_lock, flags); player->clk_adj = 0; @@ -986,12 +995,15 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + unsigned long flags; + spin_lock_irqsave(&player->irq_lock, flags); if (player->state != UNIPERIF_STATE_STOPPED) /* Stop the player */ uni_player_stop(player); player->substream = NULL; + spin_unlock_irqrestore(&player->irq_lock, flags); } static int uni_player_parse_dt_audio_glue(struct platform_device *pdev, @@ -1096,6 +1108,7 @@ int uni_player_init(struct platform_device *pdev, } mutex_init(&player->ctrl_lock); + spin_lock_init(&player->irq_lock); /* Ensure that disabled by default */ SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(player); diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 93a8df6..ee0055e 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -46,10 +46,15 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) struct uniperif *reader = dev_id; unsigned int status; + spin_lock(&reader->irq_lock); + if (!reader->substream) + goto irq_spin_unlock; + + snd_pcm_stream_lock(reader->substream); if (reader->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ dev_warn(reader->dev, "unexpected IRQ\n"); - return IRQ_HANDLED; + goto stream_unlock; } /* Get interrupt status & clear them immediately */ @@ -60,13 +65,16 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stream_lock(reader->substream); snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(reader->substream); - return IRQ_HANDLED; + ret = IRQ_HANDLED; } +stream_unlock: + snd_pcm_stream_unlock(reader->substream); +irq_spin_unlock: + spin_unlock(&reader->irq_lock); + return ret; } @@ -347,9 +355,12 @@ static int uni_reader_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *reader = priv->dai_data.uni; + unsigned long flags; int ret; + spin_lock_irqsave(&reader->irq_lock, flags); reader->substream = substream; + spin_unlock_irqrestore(&reader->irq_lock, flags); if (!UNIPERIF_TYPE_IS_TDM(reader)) return 0; @@ -375,12 +386,15 @@ static void uni_reader_shutdown(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *reader = priv->dai_data.uni; + unsigned long flags; + spin_lock_irqsave(&reader->irq_lock, flags); if (reader->state != UNIPERIF_STATE_STOPPED) { /* Stop the reader */ uni_reader_stop(reader); } reader->substream = NULL; + spin_unlock_irqrestore(&reader->irq_lock, flags); } static const struct snd_soc_dai_ops uni_reader_dai_ops = { @@ -415,6 +429,8 @@ int uni_reader_init(struct platform_device *pdev, return -EBUSY; } + spin_lock_init(&reader->irq_lock); + return 0; } EXPORT_SYMBOL_GPL(uni_reader_init); -- 1.9.1

2 1

[alsa-devel] No Headphone Jack Audio from Lenovo Docking Station with a T470 connected
by dmummenschanz＠web.de 06 Apr '17

06 Apr '17

Hello, I'm unable to get any sound from the Headphone Jack of a Lenovo Ultra Docking Station with a new Lenovo T470 connected. With a T440p however I get sound from the Docking Station Headphone Jack. I did some research on my alsa configuration and found out that there is no pin configuration supplied from the Laptop BIOS that connects to the Docking Headphone Jack. The hdajacksensetest tool only shows this when the Laptop is docked and a Headphone is pluged in: Pin 0x18 (Black Mic, Left side): present = No Pin 0x21 (Black Headphone, Left side): present = No So I did some patching using a early firmware load patch floating around the web targeting the same docking station for my older T440p laptop: echo 0x17 0x21211010 > /sys/class/sound/hwC0D0/user_pin_configs echo 1 > /sys/class/sound/hwC0D0/reconfig Now my Docking Station Headphone Jack is being detected! :) With the Headphone Connected: Pin 0x17 (Black Headphone, Docking station, Rear side): present = Yes Pin 0x18 (Black Mic, Left side): present = No Pin 0x21 (Black Headphone, Left side): present = No And disconnected: Pin 0x17 (Black Headphone, Docking station, Rear side): present = No Pin 0x18 (Black Mic, Left side): present = No Pin 0x21 (Black Headphone, Left side): present = No However there is STILL no sound even after unmuting everything in alsamixer! I'm pretty sure 0x17 is the right pincfg but I'm not sure about 0x21211010 and how I obtain such an address for my laptop. Could someone please help me out here or give me a hint? Here is a alsa-info.sh dump when the Laptop is not connected to the Dock: https://pastebin.com/eTNnfnzL And here when the Laptop is connected to the Dock (No changes): https://pastebin.com/ySBsKUeH And here after setting the pin config 0x17 0x21211010: https://pastebin.com/xmDfsNtQ I'm sure I'm very close. But something is still missing. Any help is much apprechiated! Thanks in advance Dieter

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Re: [alsa-devel] [PATCH 4/5] vfio: platform: Convert to use devm_*()
by Leo Yan 06 Apr '17

06 Apr '17

On Sun, Apr 02, 2017 at 04:45:28PM +0200, Auger Eric wrote: > Hi Leo, > > On 26/03/2017 16:41, Leo Yan wrote: > > Convert driver to use devm_*() APIs so rely on driver model core layer > > to manage resources. This eliminates error path boilerplate and makes > > code neat. > > > > This patch also renames amba_id structure, the old code used some code > > which directly copied from other driver. > > > > Signed-off-by: Leo Yan <leo.yan(a)linaro.org> > Looks good to me > Reviewed-by: Eric Auger <eric.auger(a)redhat.com> Thanks for reviewing, Eric. > May be interesting to go further converting as well the vfio-platform > driver but this can be done later on. Looked a bit for vfio-platform driver and I'm not sure if need some converting within function vfio_platform_probe_common(). So I'd like to leave this to who really understand the subsystem :) [...] Thanks, Leo Yan

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[alsa-devel] ALSA, SB1240 (Sound Blaster HD USB DAC) - need help :(
by Самохин Александр Вален тинович 06 Apr '17

06 Apr '17

Hi, I’m trying adding support for Creative SB1240(Sound Blaster HD, USB DAC).. Player Volume control is working like a charm, but.. “Capture” interface (Mic, Line, Phonograph) – not working at all. Can you help me to add support for this card? I’m trying to sniff USB traffic between USB DAC & Windows Driver(with Wireshark usbdump). But.. I need more theory, to do it.. I cannot get work any capturing interface ☹ I’t my environment: # uname -a Linux kernel 4.9.20 #1 SMP Wed Apr 5 18:42:37 MSK 2017 x86_64 GNU/Linux # cat /etc/group audio:x:63:root # dmesg *************** [ 1.670745] usb 1-1.1: New USB device found, idVendor=041e, idProduct=3232 [ 1.670748] usb 1-1.1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 1.670750] usb 1-1.1: Product: USB Sound Blaster HD [ 1.670752] usb 1-1.1: Manufacturer: Creative Technology [ 1.670753] usb 1-1.1: SerialNumber: 000003II [ 1.673435] input: Creative Technology USB Sound Blaster HD as /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.1/1-1.1:1.5/0003:041E:3232.0001/input/input6 *************** # ls -la /dev/snd/ total 0 drwxr-xr-x 4 root root 220 Apr 5 21:33 . drwxr-xr-x 15 root root 3360 Apr 5 21:34 .. drwxr-xr-x 2 root root 60 Apr 5 21:33 by-id drwxr-xr-x 2 root root 60 Apr 5 21:33 by-path crw-rw---- 1 root audio 116, 2 Apr 5 21:33 controlC0 crw-rw---- 1 root audio 116, 4 Apr 5 21:33 pcmC0D0c crw-rw---- 1 root audio 116, 3 Apr 5 21:34 pcmC0D0p crw-rw---- 1 root audio 116, 6 Apr 5 22:57 pcmC0D1c crw-rw---- 1 root audio 116, 5 Apr 5 21:33 pcmC0D1p crw-rw---- 1 root audio 116, 7 Apr 5 21:33 pcmC0D2p crw-rw---- 1 root audio 116, 33 Apr 5 21:33 timer # amixer info Card default 'HD'/'Creative Technology USB Sound Blaster HD at usb-0000:00:1a.0-1.1, full speed' Mixer name : 'USB Mixer' Components : 'USB041e:3232' Controls : 8 Simple ctrls : 2 # amixer contents numid=6,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=128,step=0 : values=128,128 | dBminmax-min=-64.00dB,max=0.00dB numid=8,iface=MIXER,name='PCM Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Mic' ; Item #1 'Line' ; Item #2 'Phonograph' : values=1 numid=4,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=1,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=3,4 | container | chmap-fixed=FL,FR numid=5,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map',device=2 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR С уважением, Александр Самохин Заместитель начальника отдела Отдел информационных технологий и компьютерного обеспечения ФГУП «Большой Московский цирк» Email: a.samokhin(a)greatcircus.ru<mailto:a.samokhin@greatcircus.ru> , it(a)greatcircus.ru<mailto:it@greatcircus.ru> Tel: +7 (495) 930-22-25, вн. 2002 Mob: +7 (903) 296-77-98

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[alsa-devel] [resend][PATCH v5 00/10] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 06 Apr '17

06 Apr '17

Hi Rob, Mark 2 weeks passed. I will re-post these patches. I added already posted "base: export symbol of_phandle_iterator_init/next" as [01/10] patch which is needed on this series. Kuninori Morimoto (10): of_graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH v4 0/2] ASoC: add driver support for CX2092X DSP
by simon.ho.cnxt＠gmail.com 06 Apr '17

06 Apr '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Pierre, Mark and Takashi I corrected the driver based on your input as below. - Corrected coding style. - Removed ".owner" assignment. - Removed SPI interface support since SPI fearure is removed from this new chip. Thanks, Simon Simon Ho (2): doc: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add support for Conexant CX2092X DSP .../devicetree/bindings/sound/cnxt,cx2092x.txt | 25 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2092x.c | 436 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 19 + 5 files changed, 488 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cnxt,cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

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[alsa-devel] [PATCH 2/2] pcm:plugin: Fix sound capture via MMAP access
by sutar.mounesh＠gmail.com 05 Apr '17

05 Apr '17

From: Timo Wischer <twischer(a)de.adit-jv.com> Distorted sound is heard if the capture device of an ioplug plugin is used, which is accessed through any ALSA plugin (like copy, linear, ...) via MMAP access. E.g. aplay -> loop -> copy -> arecord -M -> aplay -> hw:0 As mentioned in the ALSA API (see pcm/pcm.c:942): The function #snd_pcm_avail_update() have to be called before any mmap begin+commit operation. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index aaea2ab..ec57f6e 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -490,7 +490,14 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) snd_pcm_uframes_t slave_offset; snd_pcm_uframes_t slave_frames = ULONG_MAX; snd_pcm_sframes_t result; - + /* As mentioned in the ALSA API (see pcm/pcm.c:942): + * The function #snd_pcm_avail_update() + * have to be called before any mmap begin+commit operation. + * Otherwise the snd_pcm_areas_copy will not called a second time. + * But this is needed, if the ring buffer wrap is reached and + * there is more data available. + */ + slave_size = snd_pcm_avail_update(slave); result = snd_pcm_mmap_begin(slave, &slave_areas, &slave_offset, &slave_frames); if (result < 0) { err = result; -- 2.7.4

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[alsa-devel] [PATCH 1/2] plugin:dshare: wrong state reporting
by sutar.mounesh＠gmail.com 05 Apr '17

05 Apr '17

From: Andreas Pape <apape(a)de.adit-jv.com> If plugin dshare detects underrun, it reports this to the user via return value -EPIPE and setting dshare state to 'xrun' which is correct. But, if user after this wants to check the stream state, it is misleadingly reported as 'running' instead of 'xrun'. With this behavior aplay e.g. will not do a proper underrun handling (restarting stream) but terminates streaming. This is due to plugin dshare always returns state of the slave pcm, in pcm_ops->state() which is not correct. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 331f799..4c57fb4 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -244,7 +244,7 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) default: break; } - + status->state = snd_pcm_state(pcm); status->trigger_tstamp = dshare->trigger_tstamp; status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = status->avail > dshare->avail_max ? status->avail : dshare->avail_max; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: firewire-lib/bebob/oxfw: improve response evaluation for AV/C commands
by Takashi Sakamoto 05 Apr '17

05 Apr '17

In ALSA firewire stack, some AV/C commands are supported, including vendor's extensions. Drivers includes response parser of each command, according to its requirements, while the parser is written with loose fashion in two points; error check and length check. This doesn't cause any issues such as kernel corruption, but should be improved. This commit modifies evaluations of return value on each parsers. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_command.c | 30 ++++++++++++++++++++++-------- sound/firewire/fcp.c | 12 +++++++++--- sound/firewire/oxfw/oxfw-command.c | 12 +++++++++--- 3 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/firewire/bebob/bebob_command.c b/sound/firewire/bebob/bebob_command.c index 9402cc1..f9b4225 100644 --- a/sound/firewire/bebob/bebob_command.c +++ b/sound/firewire/bebob/bebob_command.c @@ -31,13 +31,15 @@ int avc_audio_set_selector(struct fw_unit *unit, unsigned int subunit_id, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(8)); - if (err > 0 && err < 9) + if (err < 0) + ; + else if (err < 9) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; else if (buf[0] == 0x0a) /* REJECTED */ err = -EINVAL; - else if (err > 0) + else err = 0; kfree(buf); @@ -67,7 +69,9 @@ int avc_audio_get_selector(struct fw_unit *unit, unsigned int subunit_id, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(8)); - if (err > 0 && err < 9) + if (err < 0) + ; + else if (err < 9) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -120,7 +124,9 @@ int avc_bridgeco_get_plug_type(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(9)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 11) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -150,7 +156,9 @@ int avc_bridgeco_get_plug_ch_pos(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, 256, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(9)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 11) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -187,7 +195,9 @@ int avc_bridgeco_get_plug_section_type(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(9) | BIT(10)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 12) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -221,7 +231,9 @@ int avc_bridgeco_get_plug_input(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 16, buf, 16, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 16) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -260,7 +272,9 @@ int avc_bridgeco_get_plug_strm_fmt(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, *len, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(10)); - if ((err >= 0) && (err < 12)) + if (err < 0) + ; + else if (err < 12) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index cce1976..61dda82 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -63,7 +63,9 @@ int avc_general_set_sig_fmt(struct fw_unit *unit, unsigned int rate, /* do transaction and check buf[1-5] are the same against command */ err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5)); - if (err >= 0 && err < 8) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -106,7 +108,9 @@ int avc_general_get_sig_fmt(struct fw_unit *unit, unsigned int *rate, /* do transaction and check buf[1-4] are the same against command */ err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2) | BIT(3) | BIT(4)); - if (err >= 0 && err < 8) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -154,7 +158,9 @@ int avc_general_get_plug_info(struct fw_unit *unit, unsigned int subunit_type, buf[3] = 0xff & subfunction; err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2)); - if (err >= 0 && err < 8) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; diff --git a/sound/firewire/oxfw/oxfw-command.c b/sound/firewire/oxfw/oxfw-command.c index 12ef325..ac3e2e3 100644 --- a/sound/firewire/oxfw/oxfw-command.c +++ b/sound/firewire/oxfw/oxfw-command.c @@ -34,7 +34,9 @@ int avc_stream_set_format(struct fw_unit *unit, enum avc_general_plug_dir dir, err = fcp_avc_transaction(unit, buf, len + 10, buf, len + 10, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(8)); - if ((err > 0) && (err < len + 10)) + if (err < 0) + ; + else if (err < len + 10) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -77,7 +79,9 @@ int avc_stream_get_format(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, *len, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7)); - if ((err > 0) && (err < 10)) + if (err < 0) + ; + else if (err < 12) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -139,7 +143,9 @@ int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, /* do transaction and check buf[1-5] are the same against command */ err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5)); - if ((err > 0) && (err < 8)) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; -- 2.9.3

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[alsa-devel] [PATCH] ALSA: firewire-motu: remove invalid bitshift for register value
by Takashi Sakamoto 05 Apr '17

05 Apr '17

In protocol version 3, drivers can read current sampling clock status from register 0x'ffff'f000'0b14. 8 bits of LSB of this register represents type of signal as source of clock. Current driver code includes invalid bitshift to handle the parameter. This commit fixes the bug. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 5992e30034c4 ("ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index b463da9..ddb6472 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -14,7 +14,6 @@ #define V3_CLOCK_RATE_MASK 0x0000ff00 #define V3_CLOCK_RATE_SHIFT 8 #define V3_CLOCK_SOURCE_MASK 0x000000ff -#define V3_CLOCK_SOURCE_SHIFT 8 #define V3_OPT_IFACE_MODE_OFFSET 0x0c94 #define V3_ENABLE_OPT_IN_IFACE_A 0x00000001 @@ -101,7 +100,7 @@ static int v3_get_clock_source(struct snd_motu *motu, return err; data = be32_to_cpu(reg); - val = (data & V3_CLOCK_SOURCE_MASK) >> V3_CLOCK_SOURCE_SHIFT; + val = data & V3_CLOCK_SOURCE_MASK; if (val == 0x00) { *src = SND_MOTU_CLOCK_SOURCE_INTERNAL; } else if (val == 0x01) { -- 2.9.3

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[alsa-devel] [PATCH] ALSA: oxfw: fix regression to handle Stanton SCS.1m/1d
by Takashi Sakamoto 05 Apr '17

05 Apr '17

At a commit 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card"), ALSA oxfw driver fails to handle SCS.1m/1d, due to -EBUSY at a call of snd_card_register(). The cause is that the driver manages to register two rawmidi instances with the same device number 0. This is a regression introduced since kernel 4.7. This commit fixes the regression, by fixing up device property after discovering stream formats. Fixes: 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card") Cc: <stable(a)vger.kernel.org> # 4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 74d7fb6..413ab63 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -227,11 +227,11 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - err = detect_quirks(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error; -- 2.9.3

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[alsa-devel] [PATCH 0/4] ALSA: firewire-digi00x: integration to support console models
by Takashi Sakamoto 05 Apr '17

05 Apr '17

Hi, Disidesign Digi00x series includes two types of unit; rack and console. ALSA Digi00x driver is originally designed to handle both types. Recently I got console model and realized that it can't handle console type, as a result of packet sniffing. This patchset is to add proper support for unit of the console type, with new information from the reverse-engineering work. Takashi Sakamoto (4): ALSA: firewire-digi00x: add support for console models of Digi00x series ALSA: firewire-digi00x: handle all MIDI messages on streaming packets ALSA: firewire-digi00x: allow user space applications to read/write MIDI messages for all ports ALSA: firewire-digi00x: remove transaction handler for unknown purpose sound/firewire/digi00x/amdtp-dot.c | 55 ++++--- sound/firewire/digi00x/digi00x-midi.c | 208 ++++++++++----------------- sound/firewire/digi00x/digi00x-transaction.c | 88 ++---------- sound/firewire/digi00x/digi00x.c | 13 +- sound/firewire/digi00x/digi00x.h | 7 +- 5 files changed, 145 insertions(+), 226 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH 00/14] ALSA: fireface: new driver for RME Fireface series
by Takashi Sakamoto 05 Apr '17

05 Apr '17

Hi, This patchset updates my previous RFCv3, and go for ALSA upstream. [alsa-devel] [RFC v3][PATCH 00/13] ALSA: fireface: new driver for RME Fireface series http://mailman.alsa-project.org/pipermail/alsa-devel/2017-February/117811.h… This patchset newly adds a driver into ALSA firewire stack, to support for Fireface 400 in RME FireWire series. Unfortunately, units on this series are completely against IEC 61883-1/6. This brings much commits and descriptions into this patchset. I'd ask you to review much amount of codes, as well as MOTU FireWire series, to finish my work for AMDTP variants[0]. As I noted to my former patchset, units of the series have unique feature; resampling between packet/data block processing layer and sample processing layer. This patchset relies on this layer, in short the driver doesn't handle sampling clock frequency, just handle sampling transmission frequency and expects units to generates audio from resampled data. Users should use proper application to configure sampling clock frequency when they want to match the frequency and sampling transmission frequency. Changes from RFCv3: * Add function pointer to AMDTP structure to reduce branching on packet processing. * Improve commit description. [0]: [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants Takashi Sakamoto (14): ALSA: fireface: add skeleton for RME Fireface series ALSA: fireface: postpone sound card registration ALSA: fireface: add model specific structure ALSA: fireface: add an abstraction layer for model-specific protocols ALSA: fireface: add transaction support ALSA: fireface: add support for MIDI functionality ALSA: fireface: add proc node to help debugging ALSA: firewire-lib: use the same prototype for functions to handle packet ALSA: firewire-lib: add no-header packet processing ALSA: fireface: add unique data processing layer ALSA: fireface: add stream management functionality ALSA: fireface: add support for PCM functionality ALSA: fireface: add hwdep interface ALSA: fireface: add support for Fireface 400 include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 2 +- sound/firewire/Kconfig | 8 + sound/firewire/Makefile | 1 + sound/firewire/amdtp-stream-trace.h | 6 +- sound/firewire/amdtp-stream.c | 107 ++++++-- sound/firewire/amdtp-stream.h | 6 + sound/firewire/fireface/Makefile | 3 + sound/firewire/fireface/amdtp-ff.c | 155 +++++++++++ sound/firewire/fireface/ff-hwdep.c | 191 +++++++++++++ sound/firewire/fireface/ff-midi.c | 131 +++++++++ sound/firewire/fireface/ff-pcm.c | 409 ++++++++++++++++++++++++++++ sound/firewire/fireface/ff-proc.c | 63 +++++ sound/firewire/fireface/ff-protocol-ff400.c | 371 +++++++++++++++++++++++++ sound/firewire/fireface/ff-stream.c | 282 +++++++++++++++++++ sound/firewire/fireface/ff-transaction.c | 295 ++++++++++++++++++++ sound/firewire/fireface/ff.c | 209 ++++++++++++++ sound/firewire/fireface/ff.h | 146 ++++++++++ 18 files changed, 2365 insertions(+), 23 deletions(-) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/amdtp-ff.c create mode 100644 sound/firewire/fireface/ff-hwdep.c create mode 100644 sound/firewire/fireface/ff-midi.c create mode 100644 sound/firewire/fireface/ff-pcm.c create mode 100644 sound/firewire/fireface/ff-proc.c create mode 100644 sound/firewire/fireface/ff-protocol-ff400.c create mode 100644 sound/firewire/fireface/ff-stream.c create mode 100644 sound/firewire/fireface/ff-transaction.c create mode 100644 sound/firewire/fireface/ff.c create mode 100644 sound/firewire/fireface/ff.h -- 2.9.3

2 15

[alsa-devel] [PATCH v2 1/2] ASoC: fsl_ssi: Remove an outdated comment
by Fabio Estevam 05 Apr '17

05 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> The comment for the FSLSSI_I2S_RATES definition states that the driver currently only supports I2S slave mode, which is no longer correct. Remove such outdated comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Only remove the outdated part of the comment sound/soc/fsl/fsl_ssi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fde08660..e0639e17 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -57,10 +57,8 @@ /** * FSLSSI_I2S_RATES: sample rates supported by the I2S * - * This driver currently only supports the SSI running in I2S slave mode, - * which means the codec determines the sample rate. Therefore, we tell - * ALSA that we support all rates and let the codec driver decide what rates - * are really supported. + * Tell ALSA that we support all rates and let the codec driver decide what + * rates are really supported. */ #define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS -- 2.7.4

1 1

[alsa-devel] [PATCH] ASoC: twl6040: Add control for HS and HF mono to stereo selection
by Peter Ujfalusi 05 Apr '17

05 Apr '17

The new controls will give user the ability to route the left PDM channel data to the right headset/handsfree DAC. HS mono to stereo switch: PDM channel 1 (or mono) data to both HS DAC. HF mono to stereo switch: PDM channel 3 data to both HF DAC. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 748036e851ea..2b6ad09e0886 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -606,6 +606,14 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { twl6040_headset_power_get_enum, twl6040_headset_power_put_enum), + /* Left HS PDM data routed to Right HSDAC */ + SOC_SINGLE("Headset Mono to Stereo Playback Switch", + TWL6040_REG_HSRCTL, 7, 1, 0), + + /* Left HF PDM data routed to Right HFDAC */ + SOC_SINGLE("Handsfree Mono to Stereo Playback Switch", + TWL6040_REG_HFRCTL, 5, 1, 0), + SOC_ENUM_EXT("PLL Selection", twl6040_power_mode_enum, twl6040_pll_get_enum, twl6040_pll_put_enum), }; -- 2.12.2

2 1

[alsa-devel] [PATCH] ASoC: max9867: export OF device ID as module aliases
by Javier Martinez Canillas 05 Apr '17

05 Apr '17

The I2C core always reports a MODALIAS of the form i2c:<foo> even if the device was registered via OF, this means that exporting the OF device ID table device aliases in the module is not needed. But in order to change how the core reports modaliases to user-space, it's better to export it. While there, move the MODULE_DEVICE_TABLE(i2c, max9867_i2c_id) just next to the I2C device table declaration, for consistency with other drivers. Before this patch: $ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias alias: i2c:max9867 After this patch: $ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias alias: i2c:max9867 alias: of:N*T*Cmaxim,max9867C* alias: of:N*T*Cmaxim,max9867 Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/max9867.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6cdf15ab46de..0247edc9c84e 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -516,13 +516,13 @@ static const struct i2c_device_id max9867_i2c_id[] = { { "max9867", 0 }, { } }; +MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); static const struct of_device_id max9867_of_match[] = { { .compatible = "maxim,max9867", }, { } }; - -MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); +MODULE_DEVICE_TABLE(of, max9867_of_match); static const struct dev_pm_ops max9867_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(max9867_suspend, max9867_resume) -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: cs53l30: Set .of_match_table to OF device ID table
by Javier Martinez Canillas 05 Apr '17

05 Apr '17

The driver has an OF device ID table but the struct i2c_driver .of_match_table field is not set. Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/cs53l30.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index cb47fb595ff4..1e0d5973b758 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -1130,6 +1130,7 @@ MODULE_DEVICE_TABLE(i2c, cs53l30_id); static struct i2c_driver cs53l30_i2c_driver = { .driver = { .name = "cs53l30", + .of_match_table = cs53l30_of_match, .pm = &cs53l30_runtime_pm, }, .id_table = cs53l30_id, -- 2.9.3

2 1

[alsa-devel] [PATCH 0/6] ASoC: Add OF device table to I2C drivers that are missing it
by Javier Martinez Canillas 05 Apr '17

05 Apr '17

Hello, This series add OF device ID tables to ASoC I2C drivers whose devices are either used in Device Tree source files or are listed in binding docs as a compatible string. That's done because the plan is to change the I2C core to report proper OF modaliases instead of always reporting a MODALIAS=i2c:<foo> regardless if a device was registered via DT or using the legacy platform data mechanism. So these patches will make sure that ASoC I2C drivers modules will continue to be autoloaded once the I2C core is changed to report proper OF modalias. Best regards, Javier Javier Martinez Canillas (6): ASoc: rt5645: Add OF device ID table ASoC: ssm4567: Add OF device ID table ASoC: sta529: Add OF device ID table ASoC: uda1380: Add OF device ID table ASoC: wm8978: Add OF device ID table ASoC: rt5677: Add OF device ID table sound/soc/codecs/rt5645.c | 10 ++++++++++ sound/soc/codecs/rt5677.c | 7 +++++++ sound/soc/codecs/ssm4567.c | 9 +++++++++ sound/soc/codecs/sta529.c | 7 +++++++ sound/soc/codecs/uda1380.c | 7 +++++++ sound/soc/codecs/wm8978.c | 7 +++++++ 6 files changed, 47 insertions(+) -- 2.9.3

3 13

[alsa-devel] [PATCH 0/2] ASoC: codec: wm8960: Relax bit clock computation when using PLL
by Daniel Baluta 05 Apr '17

05 Apr '17

This is a follow up of commit 3c01b9ee2ab ("ASoC: codec: wm8960: Relax bit clock computation") where we relaxed bitclk when sysclk was derived from MCLK. Now, we do the same thing for sysclk derived using PLL. Daniel Baluta (2): ASoC: codec: wm9860: Refactor PLL out freq search ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 105 +++++++++++++++++++++++++++++++++------------- 1 file changed, 76 insertions(+), 29 deletions(-) -- 2.7.4

5 7

[alsa-devel] [RFC][PATCH] ASoC: soc-core: verify Sound Card normality
by Kuninori Morimoto 05 Apr '17

05 Apr '17

Current ALSA SoC Sound Card basically consists of CPU/Codec/Platform components. If system uses Kernel modules, we can disable these drivers by using rmmod command. In such case, we can't disable CPU/Codec/Platform driver without disabling Sound Card driver. But on the other hand, we can disable these drivers by using unbind command. In such case, we can disable these drivers randomly. In this case, we can create dirty Sound Card which is missing necessary components. (1) If user disabled Sound Card first, but did nothing to other drivers, user can't use Sound because Sound Card is no longer exists. (2) If user disabled CPU/Codec/Platform driver randomly, but did nothing to Sound Card, user still be able to use Sound Card, because dirty Sound Card still exists. In this case, Sound system will be crashed if user started sound playback/capture. But we can't block such random unbind now. To avoid Sound Card crash in (2) case, we need to unregister Sound Card whenever CPU/Codec/Platform component were unregistered. This patch solves this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 07e4eec..52760bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3236,6 +3236,11 @@ static void snd_soc_component_cleanup(struct snd_soc_component *component) static void snd_soc_component_del_unlocked(struct snd_soc_component *component) { + struct snd_soc_card *card = component->card; + + if (card) + snd_soc_unregister_card(card); + list_del(&component->list); } -- 1.9.1

2 2

[alsa-devel] Create a general function for codec jack detection
by Bard Liao 05 Apr '17

05 Apr '17

Mark, There are many codecs with the capability of jack detection. Usually, we create a jack on machine driver and call an exported codec function, like rt5670_set_jack_detect, to use the jack and report the jack type. Could we create a general function for such exported functions? The benefit is we don't need to use different function names for each codecs. So we can have a general machine driver with jack detection function. Below is what I am thinking the patch will look like. diff --git a/include/sound/soc.h b/include/sound/soc.h index cdfb55f..e5a863a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -434,6 +434,8 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_codec_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); @@ -913,6 +915,7 @@ struct snd_soc_codec_driver { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_codec *codec, struct snd_soc_jack *jack); /* codec IO */ struct regmap *(*get_regmap)(struct device *); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fbaa1bb..ec49577 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,6 +22,23 @@ #include <trace/events/asoc.h> /** + * snd_soc_codec_set_jack - configure codec jack. + * @codec: CODEC + * @jack: structure to use for the jack + * + * Configures and enables jack detection function. + */ +int snd_soc_codec_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + if (codec->driver->set_jack) + return codec->driver->set_jack(codec, jack); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); + +/** * snd_soc_card_jack_new - Create a new jack * @card: ASoC card * @id: an identifying string for this jack -- Thanks. Bard Liao

2 1

[alsa-devel] [PATCH 38/38] Annotate hardware config module parameters in sound/pci/
by David Howells 05 Apr '17

05 Apr '17

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/pci/. Suggested-by: Alan Cox <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/pci/als4000.c | 2 +- sound/pci/cmipci.c | 6 +++--- sound/pci/ens1370.c | 2 +- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/sonicvibes.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/ymfpci/ymfpci.c | 6 +++--- 7 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 92bc06d01288..7844a75d8ed9 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -102,7 +102,7 @@ MODULE_PARM_DESC(id, "ID string for ALS4000 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable ALS4000 soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address for ALS4000 soundcard. (0 = disabled)"); #endif diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index aeedc270ed9b..430f064c64da 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -68,14 +68,14 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for C-Media PCI soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable C-Media PCI soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port."); module_param_array(soft_ac3, bool, NULL, 0444); MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only)."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); #endif diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 164adad91650..90376739c5e1 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -106,7 +106,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Ensoniq AudioPCI soundcard."); #ifdef SUPPORT_JOYSTICK #ifdef CHIP1371 -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); #else module_param_array(joystick, bool, NULL, 0444); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 19c9df6b0f3d..f067c76d77f8 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -137,12 +137,12 @@ MODULE_PARM_DESC(id, "ID string for Riptide soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Riptide soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port # for Riptide soundcard."); #endif -module_param_array(mpu_port, int, NULL, 0444); +module_param_hw_array(mpu_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU401 port # for Riptide driver."); -module_param_array(opl3_port, int, NULL, 0444); +module_param_hw_array(opl3_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver."); /* diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index a6aa48c5b969..8e3d4ec39c35 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -66,7 +66,7 @@ module_param_array(reverb, bool, NULL, 0444); MODULE_PARM_DESC(reverb, "Enable reverb (SRAM is present) for S3 SonicVibes soundcard."); module_param_array(mge, bool, NULL, 0444); MODULE_PARM_DESC(mge, "MIC Gain Enable for S3 SonicVibes soundcard."); -module_param(dmaio, uint, 0444); +module_param_hw(dmaio, uint, ioport, 0444); MODULE_PARM_DESC(dmaio, "DDMA i/o base address for S3 SonicVibes soundcard."); /* diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 2d8c14e3f8d2..127834021175 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -92,7 +92,7 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port. (VT82C686x only)"); #ifdef SUPPORT_JOYSTICK module_param(joystick, bool, 0444); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 812e27a1bcbc..4faf3e1ed06a 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -55,12 +55,12 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for the Yamaha DS-1 PCI soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Yamaha DS-1 soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 Port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM OPL-3 Port."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, long, NULL, 0444); +module_param_hw_array(joystick_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address"); #endif module_param_array(rear_switch, bool, NULL, 0444);

1 0

[alsa-devel] [PATCH 37/38] Annotate hardware config module parameters in sound/oss/
by David Howells 05 Apr '17

05 Apr '17

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/oss/. Suggested-by: Alan Cox <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: Riccardo Facchetti <fizban(a)tin.it> cc: Andrew Veliath <andrewtv(a)usa.net> cc: alsa-devel(a)alsa-project.org --- sound/oss/ad1848.c | 8 ++++---- sound/oss/aedsp16.c | 12 ++++++------ sound/oss/mpu401.c | 4 ++-- sound/oss/msnd_pinnacle.c | 20 ++++++++++---------- sound/oss/opl3.c | 2 +- sound/oss/pas2_card.c | 18 +++++++++--------- sound/oss/pss.c | 14 +++++++------- sound/oss/sb_card.c | 10 +++++----- sound/oss/trix.c | 18 +++++++++--------- sound/oss/uart401.c | 4 ++-- sound/oss/uart6850.c | 4 ++-- sound/oss/waveartist.c | 8 ++++---- 12 files changed, 61 insertions(+), 61 deletions(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index f6156d8169d0..2421f59cf279 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -2805,10 +2805,10 @@ static int __initdata dma = -1; static int __initdata dma2 = -1; static int __initdata type = 0; -module_param(io, int, 0); /* I/O for a raw AD1848 card */ -module_param(irq, int, 0); /* IRQ to use */ -module_param(dma, int, 0); /* First DMA channel */ -module_param(dma2, int, 0); /* Second DMA channel */ +module_param_hw(io, int, ioport, 0); /* I/O for a raw AD1848 card */ +module_param_hw(irq, int, irq, 0); /* IRQ to use */ +module_param_hw(dma, int, dma, 0); /* First DMA channel */ +module_param_hw(dma2, int, dma, 0); /* Second DMA channel */ module_param(type, int, 0); /* Card type */ module_param(deskpro_xl, bool, 0); /* Special magic for Deskpro XL boxen */ module_param(deskpro_m, bool, 0); /* Special magic for Deskpro M box */ diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index bb477d5c8528..f058ed6bdb69 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -1303,17 +1303,17 @@ static int __initdata mpu_irq = -1; static int __initdata mss_base = -1; static int __initdata mpu_base = -1; -module_param(io, int, 0); +module_param_hw(io, int, ioport, 0); MODULE_PARM_DESC(io, "I/O base address (0x220 0x240)"); -module_param(irq, int, 0); +module_param_hw(irq, int, irq, 0); MODULE_PARM_DESC(irq, "IRQ line (5 7 9 10 11)"); -module_param(dma, int, 0); +module_param_hw(dma, int, dma, 0); MODULE_PARM_DESC(dma, "dma line (0 1 3)"); -module_param(mpu_irq, int, 0); +module_param_hw(mpu_irq, int, irq, 0); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ line (5 7 9 10 0)"); -module_param(mss_base, int, 0); +module_param_hw(mss_base, int, ioport, 0); MODULE_PARM_DESC(mss_base, "MSS emulation I/O base address (0x530 0xE80)"); -module_param(mpu_base, int, 0); +module_param_hw(mpu_base, int, ioport, 0); MODULE_PARM_DESC(mpu_base,"MPU-401 I/O base address (0x300 0x310 0x320 0x330)"); MODULE_AUTHOR("Riccardo Facchetti <fizban(a)tin.it>"); MODULE_DESCRIPTION("Audio Excel DSP 16 Driver Version " VERSION); diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 862735005b43..20e8fa46f647 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -1748,8 +1748,8 @@ static struct address_info cfg; static int io = -1; static int irq = -1; -module_param(irq, int, 0); -module_param(io, int, 0); +module_param_hw(irq, int, irq, 0); +module_param_hw(io, int, ioport, 0); static int __init init_mpu401(void) { diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index f34ec01d2239..d2abc2cf3213 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1727,22 +1727,22 @@ static int calibrate_signal __initdata = CONFIG_MSND_CALSIGNAL; #endif /* MODULE */ -module_param (io, int, 0); -module_param (irq, int, 0); -module_param (mem, int, 0); +module_param_hw (io, int, ioport, 0); +module_param_hw (irq, int, irq, 0); +module_param_hw (mem, int, iomem, 0); module_param (write_ndelay, int, 0); module_param (fifosize, int, 0); module_param (calibrate_signal, int, 0); #ifndef MSND_CLASSIC module_param (digital, bool, 0); -module_param (cfg, int, 0); +module_param_hw (cfg, int, ioport, 0); module_param (reset, int, 0); -module_param (mpu_io, int, 0); -module_param (mpu_irq, int, 0); -module_param (ide_io0, int, 0); -module_param (ide_io1, int, 0); -module_param (ide_irq, int, 0); -module_param (joystick_io, int, 0); +module_param_hw (mpu_io, int, ioport, 0); +module_param_hw (mpu_irq, int, irq, 0); +module_param_hw (ide_io0, int, ioport, 0); +module_param_hw (ide_io1, int, ioport, 0); +module_param_hw (ide_irq, int, irq, 0); +module_param_hw (joystick_io, int, ioport, 0); #endif static int __init msnd_init(void) diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index b6d19adf8f41..f0f5b5be6314 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -1200,7 +1200,7 @@ static int me; static int io = -1; -module_param(io, int, 0); +module_param_hw(io, int, ioport, 0); static int __init init_opl3 (void) { diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c index b07954a79536..769fca692d2a 100644 --- a/sound/oss/pas2_card.c +++ b/sound/oss/pas2_card.c @@ -383,15 +383,15 @@ static int __initdata sb_irq = -1; static int __initdata sb_dma = -1; static int __initdata sb_dma16 = -1; -module_param(io, int, 0); -module_param(irq, int, 0); -module_param(dma, int, 0); -module_param(dma16, int, 0); - -module_param(sb_io, int, 0); -module_param(sb_irq, int, 0); -module_param(sb_dma, int, 0); -module_param(sb_dma16, int, 0); +module_param_hw(io, int, ioport, 0); +module_param_hw(irq, int, irq, 0); +module_param_hw(dma, int, dma, 0); +module_param_hw(dma16, int, dma, 0); + +module_param_hw(sb_io, int, ioport, 0); +module_param_hw(sb_irq, int, irq, 0); +module_param_hw(sb_dma, int, dma, 0); +module_param_hw(sb_dma16, int, dma, 0); module_param(joystick, bool, 0); module_param(symphony, bool, 0); diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 81314f9e2ccb..33c3a442e162 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -1139,19 +1139,19 @@ static bool pss_no_sound = 0; /* Just configure non-sound components */ static bool pss_keep_settings = 1; /* Keep hardware settings at module exit */ static char *pss_firmware = "/etc/sound/pss_synth"; -module_param(pss_io, int, 0); +module_param_hw(pss_io, int, ioport, 0); MODULE_PARM_DESC(pss_io, "Set i/o base of PSS card (probably 0x220 or 0x240)"); -module_param(mss_io, int, 0); +module_param_hw(mss_io, int, ioport, 0); MODULE_PARM_DESC(mss_io, "Set WSS (audio) i/o base (0x530, 0x604, 0xE80, 0xF40, or other. Address must end in 0 or 4 and must be from 0x100 to 0xFF4)"); -module_param(mss_irq, int, 0); +module_param_hw(mss_irq, int, irq, 0); MODULE_PARM_DESC(mss_irq, "Set WSS (audio) IRQ (3, 5, 7, 9, 10, 11, 12)"); -module_param(mss_dma, int, 0); +module_param_hw(mss_dma, int, dma, 0); MODULE_PARM_DESC(mss_dma, "Set WSS (audio) DMA (0, 1, 3)"); -module_param(mpu_io, int, 0); +module_param_hw(mpu_io, int, ioport, 0); MODULE_PARM_DESC(mpu_io, "Set MIDI i/o base (0x330 or other. Address must be on 4 location boundaries and must be from 0x100 to 0xFFC)"); -module_param(mpu_irq, int, 0); +module_param_hw(mpu_irq, int, irq, 0); MODULE_PARM_DESC(mpu_irq, "Set MIDI IRQ (3, 5, 7, 9, 10, 11, 12)"); -module_param(pss_cdrom_port, int, 0); +module_param_hw(pss_cdrom_port, int, ioport, 0); MODULE_PARM_DESC(pss_cdrom_port, "Set the PSS CDROM port i/o base (0x340 or other)"); module_param(pss_enable_joystick, bool, 0); MODULE_PARM_DESC(pss_enable_joystick, "Enables the PSS joystick port (1 to enable, 0 to disable)"); diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c index fb5d7250de38..2a92cfe6cfe9 100644 --- a/sound/oss/sb_card.c +++ b/sound/oss/sb_card.c @@ -61,15 +61,15 @@ static int __initdata uart401 = 0; static int __initdata pnp = 0; #endif -module_param(io, int, 000); +module_param_hw(io, int, ioport, 000); MODULE_PARM_DESC(io, "Soundblaster i/o base address (0x220,0x240,0x260,0x280)"); -module_param(irq, int, 000); +module_param_hw(irq, int, irq, 000); MODULE_PARM_DESC(irq, "IRQ (5,7,9,10)"); -module_param(dma, int, 000); +module_param_hw(dma, int, dma, 000); MODULE_PARM_DESC(dma, "8-bit DMA channel (0,1,3)"); -module_param(dma16, int, 000); +module_param_hw(dma16, int, dma, 000); MODULE_PARM_DESC(dma16, "16-bit DMA channel (5,6,7)"); -module_param(mpu_io, int, 000); +module_param_hw(mpu_io, int, ioport, 000); MODULE_PARM_DESC(mpu_io, "MPU base address"); module_param(type, int, 000); MODULE_PARM_DESC(type, "You can set this to specific card type (doesn't " \ diff --git a/sound/oss/trix.c b/sound/oss/trix.c index 3c494dc93b93..a57bc635d758 100644 --- a/sound/oss/trix.c +++ b/sound/oss/trix.c @@ -413,15 +413,15 @@ static int __initdata sb_irq = -1; static int __initdata mpu_io = -1; static int __initdata mpu_irq = -1; -module_param(io, int, 0); -module_param(irq, int, 0); -module_param(dma, int, 0); -module_param(dma2, int, 0); -module_param(sb_io, int, 0); -module_param(sb_dma, int, 0); -module_param(sb_irq, int, 0); -module_param(mpu_io, int, 0); -module_param(mpu_irq, int, 0); +module_param_hw(io, int, ioport, 0); +module_param_hw(irq, int, irq, 0); +module_param_hw(dma, int, dma, 0); +module_param_hw(dma2, int, dma, 0); +module_param_hw(sb_io, int, ioport, 0); +module_param_hw(sb_dma, int, dma, 0); +module_param_hw(sb_irq, int, irq, 0); +module_param_hw(mpu_io, int, ioport, 0); +module_param_hw(mpu_irq, int, irq, 0); module_param(joystick, bool, 0); static int __init init_trix(void) diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index dae4d4344407..83dcc85b8688 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -429,8 +429,8 @@ static struct address_info cfg_mpu; static int io = -1; static int irq = -1; -module_param(io, int, 0444); -module_param(irq, int, 0444); +module_param_hw(io, int, ioport, 0444); +module_param_hw(irq, int, irq, 0444); static int __init init_uart401(void) diff --git a/sound/oss/uart6850.c b/sound/oss/uart6850.c index 1079133dd6ab..eda32d7eddbd 100644 --- a/sound/oss/uart6850.c +++ b/sound/oss/uart6850.c @@ -315,8 +315,8 @@ static struct address_info cfg_mpu; static int __initdata io = -1; static int __initdata irq = -1; -module_param(io, int, 0); -module_param(irq, int, 0); +module_param_hw(io, int, ioport, 0); +module_param_hw(irq, int, irq, 0); static int __init init_uart6850(void) { diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 0b8d0de87273..4f0c3a232e41 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -2036,8 +2036,8 @@ __setup("waveartist=", setup_waveartist); #endif MODULE_DESCRIPTION("Rockwell WaveArtist RWA-010 sound driver"); -module_param(io, int, 0); /* IO base */ -module_param(irq, int, 0); /* IRQ */ -module_param(dma, int, 0); /* DMA */ -module_param(dma2, int, 0); /* DMA2 */ +module_param_hw(io, int, ioport, 0); /* IO base */ +module_param_hw(irq, int, irq, 0); /* IRQ */ +module_param_hw(dma, int, dma, 0); /* DMA */ +module_param_hw(dma2, int, dma, 0); /* DMA2 */ MODULE_LICENSE("GPL");

1 0

[alsa-devel] [PATCH 36/38] Annotate hardware config module parameters in sound/isa/
by David Howells 05 Apr '17

05 Apr '17

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/isa/. Suggested-by: Alan Cox <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/isa/ad1848/ad1848.c | 6 +++--- sound/isa/adlib.c | 2 +- sound/isa/cmi8328.c | 12 ++++++------ sound/isa/cmi8330.c | 20 ++++++++++---------- sound/isa/cs423x/cs4231.c | 12 ++++++------ sound/isa/cs423x/cs4236.c | 18 +++++++++--------- sound/isa/es1688/es1688.c | 12 ++++++------ sound/isa/es18xx.c | 12 ++++++------ sound/isa/galaxy/galaxy.c | 16 ++++++++-------- sound/isa/gus/gusclassic.c | 8 ++++---- sound/isa/gus/gusextreme.c | 16 ++++++++-------- sound/isa/gus/gusmax.c | 8 ++++---- sound/isa/gus/interwave.c | 10 +++++----- sound/isa/msnd/msnd_pinnacle.c | 20 ++++++++++---------- sound/isa/opl3sa2.c | 16 ++++++++-------- sound/isa/opti9xx/miro.c | 14 +++++++------- sound/isa/opti9xx/opti92x-ad1848.c | 14 +++++++------- sound/isa/sb/jazz16.c | 12 ++++++------ sound/isa/sb/sb16.c | 14 +++++++------- sound/isa/sb/sb8.c | 6 +++--- sound/isa/sc6000.c | 12 ++++++------ sound/isa/sscape.c | 12 ++++++------ sound/isa/wavefront/wavefront.c | 18 +++++++++--------- 23 files changed, 145 insertions(+), 145 deletions(-) diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index a302d1f8d14f..e739b1c85c25 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -55,11 +55,11 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver."); module_param_array(thinkpad, bool, NULL, 0444); MODULE_PARM_DESC(thinkpad, "Enable only for the onboard CS4248 of IBM Thinkpad 360/750/755 series."); diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index 8d3060fd7ad7..5fb619eca5c8 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -27,7 +27,7 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); static int snd_adlib_match(struct device *dev, unsigned int n) diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index 787475084f46..8e1756c3b9bb 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -51,18 +51,18 @@ MODULE_PARM_DESC(index, "Index value for CMI8328 soundcard."); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for CMI8328 soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for CMI8328 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for CMI8328 driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 for CMI8328 driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 for CMI8328 driver."); -module_param_array(mpuport, long, NULL, 0444); +module_param_hw_array(mpuport, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8328 driver."); -module_param_array(mpuirq, int, NULL, 0444); +module_param_hw_array(mpuirq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8328 MPU-401 port."); #ifdef SUPPORT_JOYSTICK module_param_array(gameport, bool, NULL, 0444); diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index dfedfd85f205..f64b29ab5cc7 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -95,27 +95,27 @@ module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(sbport, long, NULL, 0444); +module_param_hw_array(sbport, long, ioport, NULL, 0444); MODULE_PARM_DESC(sbport, "Port # for CMI8330/CMI8329 SB driver."); -module_param_array(sbirq, int, NULL, 0444); +module_param_hw_array(sbirq, int, irq, NULL, 0444); MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330/CMI8329 SB driver."); -module_param_array(sbdma8, int, NULL, 0444); +module_param_hw_array(sbdma8, int, dma, NULL, 0444); MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330/CMI8329 SB driver."); -module_param_array(sbdma16, int, NULL, 0444); +module_param_hw_array(sbdma16, int, dma, NULL, 0444); MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330/CMI8329 SB driver."); -module_param_array(wssport, long, NULL, 0444); +module_param_hw_array(wssport, long, ioport, NULL, 0444); MODULE_PARM_DESC(wssport, "Port # for CMI8330/CMI8329 WSS driver."); -module_param_array(wssirq, int, NULL, 0444); +module_param_hw_array(wssirq, int, irq, NULL, 0444); MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330/CMI8329 WSS driver."); -module_param_array(wssdma, int, NULL, 0444); +module_param_hw_array(wssdma, int, dma, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330/CMI8329 WSS driver."); -module_param_array(fmport, long, NULL, 0444); +module_param_hw_array(fmport, long, ioport, NULL, 0444); MODULE_PARM_DESC(fmport, "FM port # for CMI8330/CMI8329 driver."); -module_param_array(mpuport, long, NULL, 0444); +module_param_hw_array(mpuport, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330/CMI8329 driver."); -module_param_array(mpuirq, int, NULL, 0444); +module_param_hw_array(mpuirq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330/CMI8329 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ef7448e9f813..e8edd9017a2f 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -55,17 +55,17 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " CRD_NAME " driver."); static int snd_cs4231_match(struct device *dev, unsigned int n) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 9d7582c90a95..1f9a3b2be7a1 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -98,23 +98,23 @@ MODULE_PARM_DESC(enable, "Enable " IDENT " soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " IDENT " driver."); -module_param_array(cport, long, NULL, 0444); +module_param_hw_array(cport, long, ioport, NULL, 0444); MODULE_PARM_DESC(cport, "Control port # for " IDENT " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " IDENT " driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for " IDENT " driver."); -module_param_array(sb_port, long, NULL, 0444); +module_param_hw_array(sb_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(sb_port, "SB port # for " IDENT " driver (optional)."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " IDENT " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " IDENT " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " IDENT " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 1901c2bb6c3b..36320e7f2789 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -71,17 +71,17 @@ module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver."); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); #ifdef CONFIG_PNP diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5094b62d8f77..0cabe2b8974f 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1999,17 +1999,17 @@ MODULE_PARM_DESC(enable, "Enable ES18xx soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for ES18xx driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for ES18xx driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for ES18xx driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for ES18xx driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA 1 # for ES18xx driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA 2 # for ES18xx driver."); #ifdef CONFIG_PNP diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index 379abe2cbeb2..b9994cc9f5fb 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -53,21 +53,21 @@ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(wss_port, long, NULL, 0444); +module_param_hw_array(wss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(wss_port, "WSS port # for " CRD_NAME " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "Playback DMA # for " CRD_NAME " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "Capture DMA # for " CRD_NAME " driver."); /* diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index c169be49ed71..92a997ab1229 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -58,13 +58,13 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " CRD_NAME " driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for " CRD_NAME " driver."); diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 77ac2fd723b4..beb52c0f70ea 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -66,21 +66,21 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(gf1_port, long, NULL, 0444); +module_param_hw_array(gf1_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(gf1_port, "GF1 port # for " CRD_NAME " driver (optional)."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(gf1_irq, int, NULL, 0444); +module_param_hw_array(gf1_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(gf1_irq, "GF1 IRQ # for " CRD_NAME " driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "GF1 DMA # for " CRD_NAME " driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for " CRD_NAME " driver."); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index dd88c9d33492..63309a453140 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -56,13 +56,13 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for GUS MAX soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable GUS MAX soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for GUS MAX driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for GUS MAX driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for GUS MAX driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for GUS MAX driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for GUS MAX driver."); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 70d0040484c8..0687b7ef3e53 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -92,17 +92,17 @@ MODULE_PARM_DESC(enable, "Enable InterWave soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for InterWave driver."); #ifdef SNDRV_STB -module_param_array(port_tc, long, NULL, 0444); +module_param_hw_array(port_tc, long, ioport, NULL, 0444); MODULE_PARM_DESC(port_tc, "Tone control (TEA6330T - i2c bus) port # for InterWave driver."); #endif -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for InterWave driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for InterWave driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for InterWave driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for InterWave driver."); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 4c072666115d..ad4897337df5 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -800,22 +800,22 @@ MODULE_LICENSE("GPL"); MODULE_FIRMWARE(INITCODEFILE); MODULE_FIRMWARE(PERMCODEFILE); -module_param_array(io, long, NULL, S_IRUGO); +module_param_hw_array(io, long, ioport, NULL, S_IRUGO); MODULE_PARM_DESC(io, "IO port #"); -module_param_array(irq, int, NULL, S_IRUGO); -module_param_array(mem, long, NULL, S_IRUGO); +module_param_hw_array(irq, int, irq, NULL, S_IRUGO); +module_param_hw_array(mem, long, iomem, NULL, S_IRUGO); module_param_array(write_ndelay, int, NULL, S_IRUGO); module_param(calibrate_signal, int, S_IRUGO); #ifndef MSND_CLASSIC module_param_array(digital, int, NULL, S_IRUGO); -module_param_array(cfg, long, NULL, S_IRUGO); +module_param_hw_array(cfg, long, ioport, NULL, S_IRUGO); module_param_array(reset, int, 0, S_IRUGO); -module_param_array(mpu_io, long, NULL, S_IRUGO); -module_param_array(mpu_irq, int, NULL, S_IRUGO); -module_param_array(ide_io0, long, NULL, S_IRUGO); -module_param_array(ide_io1, long, NULL, S_IRUGO); -module_param_array(ide_irq, int, NULL, S_IRUGO); -module_param_array(joystick_io, long, NULL, S_IRUGO); +module_param_hw_array(mpu_io, long, ioport, NULL, S_IRUGO); +module_param_hw_array(mpu_irq, int, irq, NULL, S_IRUGO); +module_param_hw_array(ide_io0, long, ioport, NULL, S_IRUGO); +module_param_hw_array(ide_io1, long, ioport, NULL, S_IRUGO); +module_param_hw_array(ide_irq, int, irq, NULL, S_IRUGO); +module_param_hw_array(joystick_io, long, ioport, NULL, S_IRUGO); #endif diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index ae133633a420..4098e3e0353d 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -69,21 +69,21 @@ MODULE_PARM_DESC(enable, "Enable OPL3-SA soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for OPL3-SA driver."); -module_param_array(sb_port, long, NULL, 0444); +module_param_hw_array(sb_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(sb_port, "SB port # for OPL3-SA driver."); -module_param_array(wss_port, long, NULL, 0444); +module_param_hw_array(wss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(wss_port, "WSS port # for OPL3-SA driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for OPL3-SA driver."); -module_param_array(midi_port, long, NULL, 0444); +module_param_hw_array(midi_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(midi_port, "MIDI port # for OPL3-SA driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for OPL3-SA driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for OPL3-SA driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for OPL3-SA driver."); module_param_array(opl3sa3_ymode, int, NULL, 0444); MODULE_PARM_DESC(opl3sa3_ymode, "Speaker size selection for 3D Enhancement mode: Desktop/Large Notebook/Small Notebook/HiFi."); diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 3a9067db1a84..bcbff56f060d 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -69,19 +69,19 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for miro soundcard."); -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "WSS port # for miro driver."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for miro driver."); -module_param(fm_port, long, 0444); +module_param_hw(fm_port, long, ioport, 0444); MODULE_PARM_DESC(fm_port, "FM Port # for miro driver."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "WSS irq # for miro driver."); -module_param(mpu_irq, int, 0444); +module_param_hw(mpu_irq, int, irq, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 irq # for miro driver."); -module_param(dma1, int, 0444); +module_param_hw(dma1, int, dma, 0444); MODULE_PARM_DESC(dma1, "1st dma # for miro driver."); -module_param(dma2, int, 0444); +module_param_hw(dma2, int, dma, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for miro driver."); module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0a5266003786..ceddb392b1e3 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -88,20 +88,20 @@ MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard."); module_param(isapnp, bool, 0444); MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); #endif -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "WSS port # for opti9xx driver."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for opti9xx driver."); -module_param(fm_port, long, 0444); +module_param_hw(fm_port, long, ioport, 0444); MODULE_PARM_DESC(fm_port, "FM port # for opti9xx driver."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "WSS irq # for opti9xx driver."); -module_param(mpu_irq, int, 0444); +module_param_hw(mpu_irq, int, irq, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 irq # for opti9xx driver."); -module_param(dma1, int, 0444); +module_param_hw(dma1, int, dma, 0444); MODULE_PARM_DESC(dma1, "1st dma # for opti9xx driver."); #if defined(CS4231) || defined(OPTi93X) -module_param(dma2, int, 0444); +module_param_hw(dma2, int, dma, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 4d909971eedb..bfa0055e1fd6 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -50,17 +50,17 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for jazz16 driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); -module_param_array(dma16, int, NULL, 0444); +module_param_hw_array(dma16, int, dma, NULL, 0444); MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); #define SB_JAZZ16_WAKEUP 0xaf diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 4a7d7c89808f..3b2e4f405ff2 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -99,21 +99,21 @@ MODULE_PARM_DESC(enable, "Enable SoundBlaster 16 soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SB16 driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for SB16 driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for SB16 PnP driver."); #ifdef SNDRV_SBAWE_EMU8000 -module_param_array(awe_port, long, NULL, 0444); +module_param_hw_array(awe_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(awe_port, "AWE port # for SB16 PnP driver."); #endif -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SB16 driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for SB16 driver."); -module_param_array(dma16, int, NULL, 0444); +module_param_hw_array(dma16, int, dma, NULL, 0444); MODULE_PARM_DESC(dma16, "16-bit DMA # for SB16 driver."); module_param_array(mic_agc, int, NULL, 0444); MODULE_PARM_DESC(mic_agc, "Mic Auto-Gain-Control switch."); diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index ad42d2364199..d77dcba276b5 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -47,11 +47,11 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Sound Blaster soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Sound Blaster soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SB8 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SB8 driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for SB8 driver."); struct snd_sb8 { diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index b61a6633d8f2..c09d9b914efe 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -64,17 +64,17 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); -module_param_array(mss_port, long, NULL, 0444); +module_param_hw_array(mss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); -module_param_array(dma, int, NULL, 0444); +module_param_hw_array(dma, int, dma, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); module_param_array(joystick, bool, NULL, 0444); MODULE_PARM_DESC(joystick, "Enable gameport."); diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index fdcfa29e2205..54f5758a1bb3 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -63,22 +63,22 @@ MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "Description for SoundScape card"); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SoundScape driver."); -module_param_array(wss_port, long, NULL, 0444); +module_param_hw_array(wss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(wss_port, "WSS Port # for SoundScape driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SoundScape driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU401 IRQ # for SoundScape driver."); -module_param_array(dma, int, NULL, 0444); +module_param_hw_array(dma, int, dma, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); module_param_array(joystick, bool, NULL, 0444); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index a0987a57c8a9..da4e9a85f0af 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -63,23 +63,23 @@ MODULE_PARM_DESC(enable, "Enable WaveFront soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for WaveFront soundcards."); #endif -module_param_array(cs4232_pcm_port, long, NULL, 0444); +module_param_hw_array(cs4232_pcm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(cs4232_pcm_port, "Port # for CS4232 PCM interface."); -module_param_array(cs4232_pcm_irq, int, NULL, 0444); +module_param_hw_array(cs4232_pcm_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(cs4232_pcm_irq, "IRQ # for CS4232 PCM interface."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for CS4232 PCM interface."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for CS4232 PCM interface."); -module_param_array(cs4232_mpu_port, long, NULL, 0444); +module_param_hw_array(cs4232_mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(cs4232_mpu_port, "port # for CS4232 MPU-401 interface."); -module_param_array(cs4232_mpu_irq, int, NULL, 0444); +module_param_hw_array(cs4232_mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(cs4232_mpu_irq, "IRQ # for CS4232 MPU-401 interface."); -module_param_array(ics2115_irq, int, NULL, 0444); +module_param_hw_array(ics2115_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(ics2115_irq, "IRQ # for ICS2115."); -module_param_array(ics2115_port, long, NULL, 0444); +module_param_hw_array(ics2115_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(ics2115_port, "Port # for ICS2115."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port #."); module_param_array(use_cs4232_midi, bool, NULL, 0444); MODULE_PARM_DESC(use_cs4232_midi, "Use CS4232 MPU-401 interface (inaccessibly located inside your computer)");

1 0

[alsa-devel] external mic not recognized on Acer Aspire E 15 (E5-573-55U8)
by Kevin Bloch (né Field) 05 Apr '17

05 Apr '17

Hi wonderful ALSA devs, Wondering if you could help me out. I inherited this laptop and put Linux Mint 18.1 on it, and would really appreciate it if there were some way to eventually get an external mic working on it, because the internal one is naturally pretty noisy, which makes it hard to use this machine for professional meetings. Hopefully it's not against list policy to simply link elsewhere: the details are all available at http://unix.stackexchange.com/questions/355552/external-headset-mic-not-lis… . I tried getting into patching the drivers myself, but I didn't make it too far. BTW I would just report a bug with the bug tracker, but the Bug Tracker link in the sidebar of http://www.alsa-project.org/main/index.php/Main_Page leads to https://bugtrack.alsa-project.org/alsa-bug which gives a Not Found error page. Anyway please let me know what I can do to help facilitate by getting further info from my machine here. Thanks very much, Kev

1 0

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Fix channel swap on playback start
by Fabio Estevam 05 Apr '17

05 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit f8fdf5375e2005f2 ("ASoC: fsl-ssi: add SSIEN errata work around") implemented the workaround for the following erratum found on i.MX35 errata document: ENGcm06222: SSI:Transmission does not take place in bit length early frame sync configuration and also for ENGcm06222 from the same document. However it has been only applied for AC97 mode. Apply it to I2S mode as well so that it can fix audio channel swap during playback start. The channel swap can be noticed in about 10% of the times an audio track starts. With the recommended workaround in place no more channel swap happened after running audio start/stop sequence in more than 2000 times. Tested on a mx6dl-wandboard. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Do not impact 61fcf10a0ee44763e0 ("ASoC: fsl_ssi: Fix channel slipping in Playback at startup") sound/soc/fsl/fsl_ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 17f92b8..549b2a5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -575,7 +575,9 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, "Timeout waiting TX FIFO filling\n"); } } - regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr); + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); } } -- 2.7.4

7 29

[alsa-devel] [PATCH v2 0/2] ASoC: codec: wm8960: Relax bit clock computation when using PLL
by Daniel Baluta 05 Apr '17

05 Apr '17

This is a follow up of commit 3c01b9ee2ab ("ASoC: codec: wm8960: Relax bit clock computation") where we relaxed bitclk when sysclk was derived from MCLK. Now, we do the same thing for sysclk derived using PLL. Changes since v1: * fixed "may be used uninitialized" warnings [i strongly believe they were compiler false positive - see full warning log here (1)] * fixed break for inner loop [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119513.html Daniel Baluta (2): ASoC: codec: wm9860: Refactor PLL out freq search ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 109 ++++++++++++++++++++++++++++++++++------------ 1 file changed, 80 insertions(+), 29 deletions(-) -- 2.7.4

1 2

[alsa-devel] Question about hw_param with Ctrl-Z + fg
by Kuninori Morimoto 05 Apr '17

05 Apr '17

Hi ALSA SoC ML We noticed that 2nd hw_param on ALSA SoC isn't called if we used Ctrl-Z + fg on aplay. hw_param is necessary especially if we use sampling rate converter (it depends on .be_hw_params_fixup). Ctrl-Z will use notmal STOP, and fg will use normal START, thus it is impossible to distinction Ctrl-C (stop) or Ctrl-Z (temporary suspension). Is this aplay issue ? or kernel issue ? start/stop case - aplay xxx.wav => - hw_param - SNDRV_PCM_TRIGGER_START - Ctrl-C (stop) - SNDRV_PCM_TRIGGER_STOP Ctrl-Z case - aplay xxx.wav => - hw_param - SNDRV_PCM_TRIGGER_START - Ctrl-Z - SNDRV_PCM_TRIGGER_STOP - fg => - ## no hw_param here - SNDRV_PCM_TRIGGER_START - Ctrl-C (stop) - SNDRV_PCM_TRIGGER_STOP Best regards --- Kuninori Morimoto

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Re: [alsa-devel] mono/stereo translation & number of frames per period
by Paul Stone 05 Apr '17

05 Apr '17

If my driver reports channels_min of '1', then the following command produces silence rather than the expected tone. Maybe, there is a problem due to the nonstandard frequency of 16 kHz? # speaker-test -D plughw:3,0 -c1 -r16000 -l4 -t wav -w ~/1kmono.wav After rewriting my driver to support having only a single channel, I found that I have to be very careful about the parameters I provide in the hardware definition, otherwise ALSA will tend to ask for a copy operation outside of the bounds of my buffer space. This surprises me, because I'm providing the size of the buffer with the buffer_bytes_max. The buffer overrun in the copy doesn't happen all the time, and my ALSA is pretty old, so I thought maybe my ALSA has a race condition, and sometimes doesn't reset to the start of the buffer for the copy. The immediate prior pointer callback returned zero, so it seems like the problem is ALSA's. I modified my copy function to detect this situation and position to the start of the buffer, and it seems to work better now. -Paul Stone

1 0

Re: [alsa-devel] mono/stereo translation & number of frames per period
by Paul Stone 05 Apr '17

05 Apr '17

The driver should report only those sample formats that the hardware actually supports. The ALSA library can do automatic sample rate/format conversions (with the "default" or "plughw" devices). However, this does not work if you bypass it with the "hw" device. Thanks Clemens. One problem I am having is that my driver doesn't always work with the plughw interface. For instance, the following stereo pipeline works for 'hw:3,0' but outputs silence for 'plughw:3,0': gst-launch playbin2 uri=file:///home/root/Iguazu.mp3 audio-sink="alsasink device=plughw:3,0" I am using older ALSA libraries, so maybe there is a bug there? alsa-lib-1.0.27.2-r1 My hw definition looks like this: static struct snd_pcm_hardware snd_imx_audio_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER ), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_16000, .rate_min = 16000, .rate_max = 16000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = PLAYBACK_TOTAL_BUFFER_SIZE, // total num buffers * buffer size .period_bytes_min = PLAYBACK_STEREO_BUFFER_SIZE, .period_bytes_max = PLAYBACK_STEREO_BUFFER_SIZE, .periods_min = 1, .periods_max = PLAYBACK_TOTAL_NUM_BUFFERS, };

1 0

[alsa-devel] Question re: static-y sound on an Asus Prime Z270-A motherboard...
by Graham TerMarsch 05 Apr '17

05 Apr '17

I recently purchased a new computer w/an Asus Prime Z270-A motherboard in it, which uses a Realtek ALC S1220-A chipset (according to the specs at asus.com) Although the "snd_hda_codec_generic" module does load and produce sound, the "left speaker" outputs static along with the sound. Quieter the sound, the louder the static. If I use the headphone jack the sound is perfect, but using the speaker output it has static (and _only_ on the left side). I'm currently running Fedora-25, and the 4.10.5 kernel release. I tried patching in a535ad57d0e6b ("hda/realtek - New codec support of ALC1220") but with no luck; upon rebooting it does not load that module. Loading it with "modprobe" directly also does not generate any output to indicate that it detected the card and is in use. I'm not sure what the exact status is for this motherboard at the moment and whether or not it is currently supported, but I thought I would reach out and ask. Would be happy to test patches and/or provide any info I can from this motherboard, if that would prove useful. Much appreciated, -- Graham TerMarsch

1 0

[alsa-devel] PROBLEM: Internal microphone (& combined headset jack, in default config) not working on Acer VN7-592G (Skylake)
by Ralf Jung 04 Apr '17

04 Apr '17

Hi all, (This is my first bugreport to a mailing list, so I hope I'm filling out everything properly here...) [1.] Internal microphone (& combined headset jack, in default config) not working on Acer VN7-592G (Skylake) [2.] The internal microphone of my new laptop, an Acer Aspire V Nitro Black Edition (VN7-592G) with Skylake, is not working. The machine has a single "headset" jack which is supposed to support both speakers, microphones, and "combined jacks" that carry both audio-in and audio-out (like they are common on smartphones). Without any further configuration, the jack works only for external speakers. A plugged-in microphone doesn't seem to do anything, for a combined headset (i.e. sth. with both microphone and speaker in one jack), only the speaker part works. I tried all combinations of settings in pavucontrol, i.e., "Speakers" (that's the internal ones) vs. "Headphones" for audio output and "Microphone" vs. "Headset Microhpone" for the audio input. I was able to improve the situation by adding this to modprobe.d: options snd-hda-intel model=dell-headset-multi Now, mics and external speakers plugged in work fine (where for mics, I need to manually configure them in pacuvontrol: "Speakers" and "Microphone"). Combined headsets also work (with "Headphones" and "Headset Microphone"), i.e. I get audio in and audio out, but the audio in from the microphone is fairly noisy. I am not sure whether this is due to the headset or the laptop; I am using the same headset with my phone and people I called did not complain about noise. Furthermore, there is a weird effect: I have a headset with *separate* plugs for speaker and mic (which I used above to test the speaker vs. mic functionality). If I plug in the speaker part, but configure it as a microhpone (out: "Speakers", in: "Microphone"), then I actually get a (rather noisy) input signal. Not sure what is going on there, but I'm reasonably sure it is getting that signal from the headset, not the internal mic of the laptop. Even with this configuration, the internal microphone of the laptop dos not work. The system seems entirely unaware that there even is an internal microphone. Here's a Ubuntu bugreport with other folks having the same problem: <https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1523100> [3.] Keywords: audio microphone skylake combined-jack [4.] Kernel information [4.1.] Kernel version (from /proc/version): Linux version 4.9.1 (r@r-schwarzschild) (gcc version 6.2.1 20161124 (Debian 6.2.1-5) ) #1 SMP Sun Jan 8 11:40:58 CET 2017 This is a vanilla kernel compiled myself; I'm otherwise on Debian testing amd64. [4.2.] Kernel .config file: attached [8.] Environment [8.1.] Software > Linux r-schwarzschild 4.9.1 #1 SMP Sun Jan 8 11:40:58 CET 2017 x86_64 GNU/Linux > > GNU C 6.2.1 > GNU Make 4.1 > Binutils 2.27.51.20161220 > Util-linux 2.29 > Mount 2.29 > Linux C Library 2.24 > Dynamic linker (ldd) 2.24 > readlink: missing operand > Try 'readlink --help' for more information. > Procps 3.3.12 > Kbd 2.0.3 > Console-tools 2.0.3 > Sh-utils 8.26 > Udev 232 > Modules Loaded ablk_helper ac acer_wmi acpi_pad aesni_intel aes_x86_64 af_alg ahci algif_skcipher arc4 ath10k_core ath10k_pci ath autofs4 battery bbswitch bluetooth bnep bridge btbcm btintel btrtl btusb button cfg80211 cmac coretemp crc16 crc32c_intel crc32_pclmul crct10dif_pclmul cryptd cuse dm_crypt dm_mod drm drm_kms_helper ebtable_broute ebtable_filter ebtable_nat ebtables efi_pstore efivarfs efivars evdev ext4 fat fjes fscrypto fuse gf128mul ghash_clmulni_intel glue_helper hid hid_generic hid_multitouch i2c_algo_bit i2c_designware_core i2c_designware_platform i2c_hid i2c_i801 i2c_smbus i915 idma64 intel_lpss intel_lpss_acpi intel_lpss_pci intel_powerclamp intel_rapl ip6table_filter ip6table_mangle ip6table_nat ip6table_raw ip6_tables ip6table_security ip6t_REJECT ip6t_rpfilter ip_set iptable_filter iptable_mangle iptable_nat iptable_raw ip_tables iptable_security ipt_REJECT irqbypass iTCO_vendor_support iTCO_wdt jbd2 joydev kvm kvm_intel libahci libata llc lp lrw mac80211 mbcache media mei mei_me memstick mfd_core mii mmc_core nf_conntrack nf_conntrack_ipv4 nf_conntrack_ipv6 nf_defrag_ipv4 nf_defrag_ipv6 nf_nat nf_nat_ipv4 nf_nat_ipv6 nfnetlink nf_reject_ipv4 nf_reject_ipv6 nls_ascii nls_cp437 parport parport_pc pci_stub pcspkr ppdev r8169 rfkill rtsx_usb rtsx_usb_ms rtsx_usb_sdmmc scsi_mod sd_mod serio_raw sg shpchp snd snd_hda_codec snd_hda_codec_generic snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_core snd_hda_intel snd_hwdep snd_pcm snd_timer soundcore sparse_keymap stp thermal tpm tpm_crb tpm_tis tpm_tis_core usbcore usbhid uvcvideo vboxdrv vboxnetadp vboxnetflt vboxpci vfat video videobuf2_core videobuf2_memops videobuf2_v4l2 videobuf2_vmalloc videodev virt_dma wmi x86_pkg_temp_thermal xhci_hcd xhci_pci x_tables xt_conntrack xt_tcpudp [8.2.] Processor information (from /proc/cpuinfo): > vendor_id : GenuineIntel > cpu family : 6 > model : 94 > model name : Intel(R) Core(TM) i7-6700HQ CPU @ 2.60GHz > stepping : 3 > microcode : 0x9e [8.3.] Module information (from /proc/modules): > pci_stub 16384 1 - Live 0xffffffffc0d1a000 (E) > vboxpci 24576 0 - Live 0xffffffffc0d0f000 (OE) > vboxnetadp 28672 0 - Live 0xffffffffc0d07000 (OE) > vboxnetflt 28672 0 - Live 0xffffffffc0cfb000 (OE) > vboxdrv 454656 3 vboxpci,vboxnetadp,vboxnetflt, Live 0xffffffffc0c6d000 (OE) > arc4 16384 2 - Live 0xffffffffc0c68000 (E) > xt_tcpudp 16384 3 - Live 0xffffffffc0c63000 (E) > ip6t_rpfilter 16384 1 - Live 0xffffffffc0c5e000 (E) > ip6t_REJECT 16384 2 - Live 0xffffffffc0c59000 (E) > nf_reject_ipv6 16384 1 ip6t_REJECT, Live 0xffffffffc0c54000 (E) > ipt_REJECT 16384 2 - Live 0xffffffffc0c44000 (E) > nf_reject_ipv4 16384 1 ipt_REJECT, Live 0xffffffffc0c3f000 (E) > xt_conntrack 16384 11 - Live 0xffffffffc08cc000 (E) > ip_set 45056 0 - Live 0xffffffffc08d1000 (E) > nfnetlink 16384 1 ip_set, Live 0xffffffffc08a7000 (E) > ebtable_nat 16384 1 - Live 0xffffffffc0c4f000 (E) > ebtable_broute 16384 1 - Live 0xffffffffc0c4a000 (E) > bridge 131072 1 ebtable_broute, Live 0xffffffffc0b50000 (E) > stp 16384 1 bridge, Live 0xffffffffc0b4b000 (E) > llc 16384 2 bridge,stp, Live 0xffffffffc0b42000 (E) > ip6table_raw 16384 1 - Live 0xffffffffc0b3d000 (E) > ip6table_nat 16384 1 - Live 0xffffffffc0b38000 (E) > nf_conntrack_ipv6 20480 7 - Live 0xffffffffc0b1d000 (E) > nf_defrag_ipv6 36864 1 nf_conntrack_ipv6, Live 0xffffffffc0b2a000 (E) > nf_nat_ipv6 16384 1 ip6table_nat, Live 0xffffffffc0b14000 (E) > ip6table_mangle 16384 1 - Live 0xffffffffc0b0f000 (E) > ip6table_security 16384 1 - Live 0xffffffffc0b0a000 (E) > iptable_raw 16384 1 - Live 0xffffffffc0b05000 (E) > iptable_nat 16384 1 - Live 0xffffffffc0b00000 (E) > nf_conntrack_ipv4 16384 6 - Live 0xffffffffc0af7000 (E) > nf_defrag_ipv4 16384 1 nf_conntrack_ipv4, Live 0xffffffffc0a37000 (E) > nf_nat_ipv4 16384 1 iptable_nat, Live 0xffffffffc08b4000 (E) > nf_nat 28672 2 nf_nat_ipv6,nf_nat_ipv4, Live 0xffffffffc0a2f000 (E) > nf_conntrack 114688 6 xt_conntrack,nf_conntrack_ipv6,nf_nat_ipv6,nf_conntrack_ipv4,nf_nat_ipv4,nf_nat, Live 0xffffffffc0930000 (E) > iptable_mangle 16384 1 - Live 0xffffffffc0890000 (E) > iptable_security 16384 1 - Live 0xffffffffc088b000 (E) > ebtable_filter 16384 1 - Live 0xffffffffc070e000 (E) > ebtables 36864 3 ebtable_nat,ebtable_broute,ebtable_filter, Live 0xffffffffc0746000 (E) > ip6table_filter 16384 1 - Live 0xffffffffc06b1000 (E) > ip6_tables 28672 5 ip6table_raw,ip6table_nat,ip6table_mangle,ip6table_security,ip6table_filter, Live 0xffffffffc0719000 (E) > iptable_filter 16384 1 - Live 0xffffffffc0698000 (E) > cmac 16384 1 - Live 0xffffffffc0693000 (E) > bbswitch 16384 0 - Live 0xffffffffc0642000 (OE) > bnep 20480 2 - Live 0xffffffffc062e000 (E) > hid_multitouch 20480 0 - Live 0xffffffffc063c000 (E) > joydev 20480 0 - Live 0xffffffffc07a3000 (E) > snd_hda_codec_hdmi 45056 1 - Live 0xffffffffc0924000 (E) > nls_ascii 16384 1 - Live 0xffffffffc0663000 (E) > ath10k_pci 45056 0 - Live 0xffffffffc08e7000 (E) > nls_cp437 20480 1 - Live 0xffffffffc07f9000 (E) > ath10k_core 245760 1 ath10k_pci, Live 0xffffffffc09e2000 (E) > vfat 20480 1 - Live 0xffffffffc07f3000 (E) > ath 32768 1 ath10k_core, Live 0xffffffffc07aa000 (E) > acer_wmi 20480 0 - Live 0xffffffffc06d5000 (E) > i2c_designware_platform 16384 0 - Live 0xffffffffc052b000 (E) > fat 69632 1 vfat, Live 0xffffffffc08ba000 (E) > snd_hda_codec_realtek 86016 1 - Live 0xffffffffc090e000 (E) > efi_pstore 16384 0 - Live 0xffffffffc0497000 (E) > i2c_designware_core 20480 1 i2c_designware_platform, Live 0xffffffffc08ae000 (E) > snd_hda_codec_generic 69632 1 snd_hda_codec_realtek, Live 0xffffffffc0895000 (E) > intel_rapl 20480 0 - Live 0xffffffffc069f000 (E) > sparse_keymap 16384 1 acer_wmi, Live 0xffffffffc0526000 (E) > mac80211 663552 1 ath10k_core, Live 0xffffffffc0b9c000 (E) > x86_pkg_temp_thermal 16384 0 - Live 0xffffffffc0b25000 (E) > cfg80211 552960 3 ath10k_core,ath,mac80211, Live 0xffffffffc0a6f000 (E) > snd_hda_intel 36864 6 - Live 0xffffffffc08dd000 (E) > intel_powerclamp 16384 0 - Live 0xffffffffc06ac000 (E) > btusb 45056 0 - Live 0xffffffffc0a23000 (E) > coretemp 16384 0 - Live 0xffffffffc0647000 (E) > btrtl 16384 1 btusb, Live 0xffffffffc07ee000 (E) > snd_hda_codec 135168 4 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_codec_generic,snd_hda_intel, Live 0xffffffffc0b7a000 (E) > uvcvideo 90112 0 - Live 0xffffffffc08f7000 (E) > snd_hda_core 81920 5 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_codec_generic,snd_hda_intel,snd_hda_codec, Live 0xffffffffc07d9000 (E) > btbcm 16384 1 btusb, Live 0xffffffffc07d4000 (E) > videobuf2_vmalloc 16384 1 uvcvideo, Live 0xffffffffc0764000 (E) > btintel 16384 1 btusb, Live 0xffffffffc075f000 (E) > videobuf2_memops 16384 1 videobuf2_vmalloc, Live 0xffffffffc07cf000 (E) > snd_hwdep 16384 1 snd_hda_codec, Live 0xffffffffc0b71000 (E) > videobuf2_v4l2 24576 1 uvcvideo, Live 0xffffffffc06a5000 (E) > kvm_intel 192512 0 - Live 0xffffffffc0a3f000 (E) > videobuf2_core 36864 2 uvcvideo,videobuf2_v4l2, Live 0xffffffffc0668000 (E) > kvm 589824 1 kvm_intel, Live 0xffffffffc0951000 (E) > bluetooth 552960 26 bnep,btusb,btrtl,btbcm,btintel, Live 0xffffffffc0803000 (E) > snd_pcm 110592 6 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hda_core, Live 0xffffffffc07b3000 (E) > videodev 176128 3 uvcvideo,videobuf2_v4l2,videobuf2_core, Live 0xffffffffc0777000 (E) > rtsx_usb_ms 20480 0 - Live 0xffffffffc076d000 (E) > irqbypass 16384 1 kvm, Live 0xffffffffc0677000 (E) > efivars 20480 1 efi_pstore, Live 0xffffffffc0680000 (E) > media 40960 2 uvcvideo,videodev, Live 0xffffffffc073b000 (E) > snd_timer 32768 1 snd_pcm, Live 0xffffffffc0756000 (E) > mei_me 36864 0 - Live 0xffffffffc06ec000 (E) > pcspkr 16384 0 - Live 0xffffffffc0629000 (E) > serio_raw 16384 0 - Live 0xffffffffc0624000 (E) > snd 86016 18 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_codec_generic,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer, Live 0xffffffffc0725000 (E) > idma64 20480 0 - Live 0xffffffffc0510000 (E) > iTCO_wdt 16384 0 - Live 0xffffffffc0751000 (E) > memstick 20480 1 rtsx_usb_ms, Live 0xffffffffc0713000 (E) > rfkill 24576 8 acer_wmi,cfg80211,bluetooth, Live 0xffffffffc0707000 (E) > iTCO_vendor_support 16384 1 iTCO_wdt, Live 0xffffffffc0702000 (E) > soundcore 16384 1 snd, Live 0xffffffffc06f9000 (E) > sg 32768 0 - Live 0xffffffffc06df000 (E) > mei 102400 1 mei_me, Live 0xffffffffc06bb000 (E) > virt_dma 16384 1 idma64, Live 0xffffffffc06b6000 (E) > shpchp 36864 0 - Live 0xffffffffc0659000 (E) > intel_lpss_pci 16384 0 - Live 0xffffffffc0650000 (E) > battery 16384 0 - Live 0xffffffffc0519000 (E) > ac 16384 0 - Live 0xffffffffc010e000 (E) > intel_lpss_acpi 16384 0 - Live 0xffffffffc068e000 (E) > wmi 16384 1 acer_wmi, Live 0xffffffffc0672000 (E) > intel_lpss 16384 2 intel_lpss_pci,intel_lpss_acpi, Live 0xffffffffc031f000 (E) > evdev 24576 18 - Live 0xffffffffc061d000 (E) > acpi_pad 24576 0 - Live 0xffffffffc0616000 (E) > tpm_crb 16384 0 - Live 0xffffffffc0637000 (E) > tpm_tis 16384 0 - Live 0xffffffffc0492000 (E) > tpm_tis_core 20480 1 tpm_tis, Live 0xffffffffc0688000 (E) > tpm 45056 3 tpm_crb,tpm_tis,tpm_tis_core, Live 0xffffffffc0532000 (E) > cuse 16384 3 - Live 0xffffffffc02b5000 (E) > fuse 98304 2 cuse, Live 0xffffffffc04b7000 (E) > parport_pc 28672 0 - Live 0xffffffffc048a000 (E) > ppdev 20480 0 - Live 0xffffffffc0484000 (E) > lp 20480 0 - Live 0xffffffffc0335000 (E) > parport 49152 3 parport_pc,ppdev,lp, Live 0xffffffffc02fe000 (E) > efivarfs 16384 1 - Live 0xffffffffc02f9000 (E) > ip_tables 24576 5 iptable_raw,iptable_nat,iptable_mangle,iptable_security,iptable_filter, Live 0xffffffffc02ba000 (E) > x_tables 36864 16 xt_tcpudp,ip6t_rpfilter,ip6t_REJECT,ipt_REJECT,xt_conntrack,ip6table_raw,ip6table_mangle,ip6table_security,iptable_raw,iptable_mangle,iptable_security,ebtables,ip6table_filter,ip6_tables,iptable_filter,ip_tables, Live 0xffffffffc025a000 (E) > autofs4 40960 2 - Live 0xffffffffc02aa000 (E) > ext4 581632 4 - Live 0xffffffffc0587000 (E) > crc16 16384 2 bluetooth,ext4, Live 0xffffffffc0265000 (E) > jbd2 106496 1 ext4, Live 0xffffffffc0469000 (E) > fscrypto 28672 1 ext4, Live 0xffffffffc01cf000 (E) > mbcache 16384 5 ext4, Live 0xffffffffc01c6000 (E) > algif_skcipher 20480 0 - Live 0xffffffffc0108000 (E) > af_alg 16384 1 algif_skcipher, Live 0xffffffffc00f5000 (E) > dm_crypt 24576 2 - Live 0xffffffffc011d000 (E) > dm_mod 114688 14 dm_crypt, Live 0xffffffffc056a000 (E) > rtsx_usb_sdmmc 28672 0 - Live 0xffffffffc0115000 (E) > mmc_core 143360 1 rtsx_usb_sdmmc, Live 0xffffffffc04e5000 (E) > rtsx_usb 24576 2 rtsx_usb_ms,rtsx_usb_sdmmc, Live 0xffffffffc00ee000 (E) > mfd_core 16384 2 intel_lpss,rtsx_usb, Live 0xffffffffc00e9000 (E) > hid_generic 16384 0 - Live 0xffffffffc00df000 (E) > usbhid 53248 0 - Live 0xffffffffc00fa000 (E) > sd_mod 45056 6 - Live 0xffffffffc009a000 (E) > crct10dif_pclmul 16384 0 - Live 0xffffffffc0521000 (E) > crc32_pclmul 16384 0 - Live 0xffffffffc04b2000 (E) > crc32c_intel 24576 8 - Live 0xffffffffc0509000 (E) > ghash_clmulni_intel 16384 0 - Live 0xffffffffc04d7000 (E) > aesni_intel 167936 8 - Live 0xffffffffc0540000 (E) > aes_x86_64 20480 1 aesni_intel, Live 0xffffffffc04df000 (E) > lrw 16384 1 aesni_intel, Live 0xffffffffc04aa000 (E) > gf128mul 16384 1 lrw, Live 0xffffffffc04d2000 (E) > glue_helper 16384 1 aesni_intel, Live 0xffffffffc00e4000 (E) > ablk_helper 16384 1 aesni_intel, Live 0xffffffffc00b5000 (E) > cryptd 24576 5 ghash_clmulni_intel,aesni_intel,ablk_helper, Live 0xffffffffc049f000 (E) > i915 1228800 29 - Live 0xffffffffc033c000 (E) > ahci 36864 4 - Live 0xffffffffc032b000 (E) > i2c_algo_bit 16384 1 i915, Live 0xffffffffc0324000 (E) > libahci 32768 1 ahci, Live 0xffffffffc0316000 (E) > i2c_i801 24576 0 - Live 0xffffffffc030b000 (E) > drm_kms_helper 155648 1 i915, Live 0xffffffffc02d2000 (E) > xhci_pci 16384 0 - Live 0xffffffffc02c9000 (E) > i2c_smbus 16384 1 i2c_i801, Live 0xffffffffc02c1000 (E) > libata 249856 2 ahci,libahci, Live 0xffffffffc026c000 (E) > xhci_hcd 188416 1 xhci_pci, Live 0xffffffffc0216000 (E) > r8169 81920 0 - Live 0xffffffffc0245000 (E) > mii 16384 1 r8169, Live 0xffffffffc020f000 (E) > scsi_mod 225280 3 sg,sd_mod,libata, Live 0xffffffffc01d7000 (E) > usbcore 249856 6 btusb,uvcvideo,rtsx_usb,usbhid,xhci_pci,xhci_hcd, Live 0xffffffffc0184000 (E) > drm 360448 9 i915,drm_kms_helper, Live 0xffffffffc012b000 (E) > thermal 20480 0 - Live 0xffffffffc00d9000 (E) > i2c_hid 20480 0 - Live 0xffffffffc0125000 (E) > hid 122880 4 hid_multitouch,hid_generic,usbhid,i2c_hid, Live 0xffffffffc00ba000 (E) > video 40960 2 acer_wmi,i915, Live 0xffffffffc00aa000 (E) > fjes 28672 0 - Live 0xffffffffc0092000 (E) > button 16384 1 i915, Live 0xffffffffc0089000 (E) [8.4.] Loaded driver and hardware information /proc/ioports: > 0000-0000 : PCI Bus 0000:00 > 0000-0000 : dma1 > 0000-0000 : pic1 > 0000-0000 : timer0 > 0000-0000 : timer1 > 0000-0000 : keyboard > 0000-0000 : PNP0C09:00 > 0000-0000 : EC data > 0000-0000 : keyboard > 0000-0000 : PNP0C09:00 > 0000-0000 : EC cmd > 0000-0000 : rtc0 > 0000-0000 : dma page reg > 0000-0000 : pic2 > 0000-0000 : dma2 > 0000-0000 : fpu > 0000-0000 : PNP0C04:00 > 0000-0000 : iTCO_wdt > 0000-0000 : iTCO_wdt > 0000-0000 : pnp 00:03 > 0000-0000 : PCI conf1 > 0000-0000 : PCI Bus 0000:00 > 0000-0000 : pnp 00:02 > 0000-0000 : ACPI PM1a_EVT_BLK > 0000-0000 : ACPI PM1a_CNT_BLK > 0000-0000 : ACPI PM_TMR > 0000-0000 : ACPI CPU throttle > 0000-0000 : iTCO_wdt > 0000-0000 : iTCO_wdt > 0000-0000 : ACPI PM2_CNT_BLK > 0000-0000 : pnp 00:05 > 0000-0000 : ACPI GPE0_BLK > 0000-0000 : pnp 00:01 > 0000-0000 : PCI Bus 0000:07 > 0000-0000 : 0000:07:00.0 > 0000-0000 : r8169 > 0000-0000 : PCI Bus 0000:01 > 0000-0000 : 0000:01:00.0 > 0000-0000 : 0000:00:02.0 > 0000-0000 : 0000:00:1f.4 > 0000-0000 : i801_smbus > 0000-0000 : 0000:00:17.0 > 0000-0000 : ahci > 0000-0000 : 0000:00:17.0 > 0000-0000 : ahci > 0000-0000 : 0000:00:17.0 > 0000-0000 : ahci > 0000-0000 : PCI Bus 0000:08 > 0000-0000 : pnp 00:02 > 0000-0000 : pnp 00:02 > 0000-0000 : pnp 00:02 /proc/iomem: > 00000000-00000000 : reserved > 00000000-00000000 : System RAM > 00000000-00000000 : reserved > 00000000-00000000 : System RAM > 00000000-00000000 : reserved > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : System ROM > 00000000-00000000 : System RAM > 00000000-00000000 : ACPI Non-volatile Storage > 00000000-00000000 : reserved > 00000000-00000000 : System RAM > 00000000-00000000 : reserved > 00000000-00000000 : System RAM > 00000000-00000000 : reserved > 00000000-00000000 : reserved > 00000000-00000000 : ACPI Non-volatile Storage > 00000000-00000000 : ACPI Tables > 00000000-00000000 : System RAM > 00000000-00000000 : reserved > 00000000-00000000 : Graphics Stolen Memory > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : pnp 00:08 > 00000000-00000000 : PCI Bus 0000:01 > 00000000-00000000 : 0000:01:00.0 > 00000000-00000000 : 0000:01:00.0 > 00000000-00000000 : PCI Bus 0000:08 > 00000000-00000000 : 0000:00:02.0 > 00000000-00000000 : PCI Bus 0000:01 > 00000000-00000000 : 0000:01:00.0 > 00000000-00000000 : PCI Bus 0000:06 > 00000000-00000000 : 0000:06:00.0 > 00000000-00000000 : ath > 00000000-00000000 : PCI Bus 0000:07 > 00000000-00000000 : 0000:07:00.0 > 00000000-00000000 : r8169 > 00000000-00000000 : 0000:07:00.0 > 00000000-00000000 : r8169 > 00000000-00000000 : 0000:00:14.0 > 00000000-00000000 : xhci-hcd > 00000000-00000000 : 0000:00:1f.3 > 00000000-00000000 : ICH HD audio > 00000000-00000000 : 0000:00:1f.3 > 00000000-00000000 : ICH HD audio > 00000000-00000000 : 0000:00:1f.2 > 00000000-00000000 : 0000:00:17.0 > 00000000-00000000 : ahci > 00000000-00000000 : 0000:00:15.0 > 00000000-00000000 : lpss_dev > 00000000-00000000 : lpss_dev > 00000000-00000000 : lpss_priv > 00000000-00000000 : idma64.0 > 00000000-00000000 : idma64.0 > 00000000-00000000 : 0000:00:15.1 > 00000000-00000000 : lpss_dev > 00000000-00000000 : lpss_dev > 00000000-00000000 : lpss_priv > 00000000-00000000 : idma64.1 > 00000000-00000000 : idma64.1 > 00000000-00000000 : 0000:00:16.0 > 00000000-00000000 : mei_me > 00000000-00000000 : 0000:00:17.0 > 00000000-00000000 : ahci > 00000000-00000000 : 0000:00:1f.4 > 00000000-00000000 : 0000:00:17.0 > 00000000-00000000 : ahci > 00000000-00000000 : 0000:00:02.0 > 00000000-00000000 : PCI Bus 0000:08 > 00000000-00000000 : PCI MMCONFIG 0000 [bus 00-ff] > 00000000-00000000 : reserved > 00000000-00000000 : pnp 00:08 > 00000000-00000000 : reserved > 00000000-00000000 : PCI Bus 0000:00 > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : pnp 00:07 > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : pnp 00:07 > 00000000-00000000 : pnp 00:07 > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : iTCO_wdt > 00000000-00000000 : iTCO_wdt > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : pnp 00:00 > 00000000-00000000 : pnp 00:02 > 00000000-00000000 : reserved > 00000000-00000000 : reserved > 00000000-00000000 : IOAPIC 0 > 00000000-00000000 : reserved > 00000000-00000000 : HPET 0 > 00000000-00000000 : PNP0103:00 > 00000000-00000000 : reserved > 00000000-00000000 : pnp 00:08 > 00000000-00000000 : pnp 00:08 > 00000000-00000000 : pnp 00:08 > 00000000-00000000 : pnp 00:08 > 00000000-00000000 : MSFT0101:00 > 00000000-00000000 : reserved > 00000000-00000000 : dmar0 > 00000000-00000000 : dmar1 > 00000000-00000000 : Local APIC > 00000000-00000000 : reserved > 00000000-00000000 : INT0800:00 > 00000000-00000000 : reserved > 00000000-00000000 : System RAM > 00000000-00000000 : Kernel code > 00000000-00000000 : Kernel data > 00000000-00000000 : Kernel bss > 00000000-00000000 : RAM buffer [8.5.] PCI information: attached I would be very grateful for your assistance. Let me know if you need any further information. Kind regards, Ralf

2 3

[alsa-devel] ESI Maya22 init issues
by Rabit 04 Apr '17

04 Apr '17

Hi AlsaDevs, Seems like something wrong with maya22 usb initialization. I'm trying to use my control application after lin init (fail) and win init (working): https://github.com/rabits/esi-maya22-linux - it's working fine on ubuntu 14.04 ~yr ago, when I'm done this app, but now it's broken... Could you please help me with figuring out what's wrong? I can attach required info and try another ubuntu versions. Attached alsainfo. Also can provide pcapng with init process for lin & win. -- Best regards

1 1

[alsa-devel] [RFC][PATCH] ASoC: soc-core: verify Sound Card normality
by Kuninori Morimoto 04 Apr '17

04 Apr '17

Current ALSA SoC Sound Card basically consists of CPU/Codec/Platform drivers. If system uses Kernel modules, we can disable these drivers by using rmmod command. In such case, we can't disable CPU/Codec/Platform driver without disabling Sound Card driver. But on the other hand, we can disable these drivers by using unbind command. In such case, we can disable these drivers randomly. In this case, we can create dirty Sound Card which is missing necessary components. (1) If user disabled Sound Card first, but did nothing to other drivers, user can't use Sound because Sound Card is no longer exists. (2) If user disabled CPU/Codec/Platform driver randomly, but did nothing to Sound Card, user still be able to use Sound Card, because dirty Sound Card which is missing necessary components still exists. In this case, Sound system will be crashed if user started sound playback/capture. But we can't block such random unbind now. To avoid Sound Card crash in (2) case, what we can do now is, add dirty flag on Sound Card, and avoid to open Sound Card. This patch solved this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Lars-Peter This is still RFC, but I think we need to consider about this issue. I tested this patch on my system, and it can block to sound card crash for me include/sound/soc.h | 6 ++++-- sound/soc/soc-core.c | 12 +++++++++++- sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 3 files changed, 35 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 316fdce..4813fc5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1106,8 +1106,6 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; - bool instantiated; - int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); int (*remove)(struct snd_soc_card *card); @@ -1197,6 +1195,10 @@ struct snd_soc_card { u32 pop_time; void *drvdata; + + /* bit field */ + u32 instantiated:1; + u32 dirty:1; }; /* SoC machine DAI configuration, glues a codec and cpu DAI together */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 07e4eec..a18e106 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2882,7 +2882,15 @@ int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) { card->instantiated = false; - snd_soc_dapm_shutdown(card); + if (!card->dirty) { + /* + * In case of unbind, CPU/Codec/Platform component can + * be unbinded *before* Card unbind. In such case + * (= card->dirty) Card connected DAPMs are already + * doesn't exist. + */ + snd_soc_dapm_shutdown(card); + } soc_cleanup_card_resources(card); dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } @@ -3236,6 +3244,8 @@ static void snd_soc_component_cleanup(struct snd_soc_component *component) static void snd_soc_component_del_unlocked(struct snd_soc_component *component) { + if (component->card) + component->card->dirty = 1; list_del(&component->list); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index efc5831..c861aba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -455,6 +455,16 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) const char *codec_dai_name = "multicodec"; int i, ret = 0; + if (rtd->card->dirty) { + /* + * In case of unbind, each CPU/Codec/Platform component can + * be unbinded randomly. In such case (= card->dirty) + * Sound Card is no longer safety. Don't open it. + */ + dev_warn(cpu_dai->dev, "Sound SoC Card is dirty, Recheck your system\n"); + return -ENXIO; + } + pinctrl_pm_select_default_state(cpu_dai->dev); for (i = 0; i < rtd->num_codecs; i++) pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev); @@ -2577,6 +2587,16 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) int ret; int stream = fe_substream->stream; + if (fe->card->dirty) { + /* + * In case of unbind, each CPU/Codec/Platform component can + * be unbinded randomly. In such case (= card->dirty) + * Sound Card is no longer safety. Don't open it. + */ + dev_warn(fe->dev, "Sound SoC Card is dirty, Recheck your system\n"); + return -ENXIO; + } + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); fe->dpcm[stream].runtime = fe_substream->runtime; -- 1.9.1

4 11

[alsa-devel] Token
by David Hedlund 04 Apr '17

04 Apr '17

Hi, I'm a volunteer for h-node.org. You can check my free software contributions at https://directory.fsf.org/wiki/User:David_Hedlund#Free_software_contributio… to make sure I'm a serious. I'm trying to make an account but http://www.alsa-project.org/main/index.php?title=Special:UserLogin&type=sig… says: Token to prevent spam? Please, ask on mailing lists... What is it?

1 0

[alsa-devel] mono/stereo translation & number of frames per period
by Paul Stone 04 Apr '17

04 Apr '17

I thought that simply by reporting that my driver only supports two channels, that I could simply utilize a mixer or that ALSA would automatically support sending and receiving monaural data to/from my stereo-only driver. Ultimately, I wasn't able to make it work. Is this a mode of operation that ALSA supports, or does ALSA require the driver to support single-channel operation in order to send/receive monaural data? So, I decided that I should support both mono and stereo in the driver. It appears that ALSA looks at the minimum period size and decides to send that much audio data for playback. The minimum period size is in bytes, which means that the number of mono and stereo frames per period will differ. This introduces a bit of additional unwanted complexity. I'm wondering if there is a way for ALSA to support a fixed number of frames per period. I see that there is a bit field SNDRV_PCM_INFO_FIFO_IN_FRAMES, and a fifo_size h/w parameter, but I'm not sure how this relates to the period size. I feel like this is a common mode of operation (fixed number of frames per period, regardless of the number of channels), and that it's likely supported by ALSA, but I don't know how it's supposed to work. Should I set up a rule to alter the minimum period size, based on the number of channels? Best Regards, Paul Stone

2 1

[alsa-devel] Playback formats; does aplay pad S16_LE, or does the hardware?
by Rob Nertney 04 Apr '17

04 Apr '17

I can successfully play 32b@32KHz and 32b@16KHz to my I2S DAC. I can also, with the same settings as 32b@16KHz, play 16b@32KHz, but I don't think this is the way ALSA wants its data.... I've learned that hw_params is responsible for describing only what the hardware is supposed to be capable of. Does this mean the hardware needs to find a way to pad 0's on lower quality data for any given rate? For example, when I play 32b @32 KHz, everything works. If I play 16b@32KHz, I shouldn't change any output clocks, but will also present the data on a 32bit data bus as 16bits padded with 0? Or does it pack the data such that the hardware needs to determine that it's in 16b mode, and pad the data itself? Thanks, Rob

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Apr '17

04 Apr '17

From: Zidan Wang <b50113(a)freescale.com> When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK. When current bias_level is SND_SOC_BIAS_ON, it is preparing for a transition away from ON. In this case, disable the codec mclk. When current bias_level is not SND_SOC_BIAS_ON, it preparing for a transition to ON. In this case, enable the codec mclk. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 51 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 48 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 75cd7de..fd25973 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +118,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -618,14 +620,38 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - /* Set VMID to 2x50k */ - snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_STANDBY: + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } + + /* Set VMID to 2x50k */ + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + break; + + case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + break; + + default: + break; + } + break; case SND_SOC_BIAS_STANDBY: @@ -674,7 +700,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; switch (level) { case SND_SOC_BIAS_ON: @@ -715,9 +741,22 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } break; case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -1002,6 +1041,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); -- 1.9.1

5 10

[alsa-devel] [PATCH v5] ASoC: Add support for Maxim Integrated MAX98927 Amplifier
by Ryan Lee 03 Apr '17

03 Apr '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v5 * Changed mode to 644. Changes since v4: * Removed support for SND_SOC_DAIFMT_CBS_CFM. * Fixed coding style for indention. * Removed variables if it has only one user. * Assigned ch_size directly. * Removed oring. * Put the return false in the switch statement. * Removed 'Monomix Output' and 'Speaker Source' controls. * Modified control names per control-names.rst. * Moved Revision ID check code to i2c_probe function. * Added 'Current Limit' control. * Removed 'devm__kfree' function. Changes since v3: * Combined MAX98926 and MAX98927 binding. Kept existing property name. Changes since v2: * Removed local register read/write function to avoid duplication of ASoC core function. .../devicetree/bindings/sound/max98925.txt | 22 - .../devicetree/bindings/sound/max98926.txt | 32 - .../devicetree/bindings/sound/max9892x.txt | 41 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98927.c | 841 +++++++++++++++++++++ sound/soc/codecs/max98927.h | 272 +++++++ 7 files changed, 1161 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max98925.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 sound/soc/codecs/max98927.c create mode 100644 sound/soc/codecs/max98927.h diff --git a/Documentation/devicetree/bindings/sound/max98925.txt b/Documentation/devicetree/bindings/sound/max98925.txt deleted file mode 100644 index 27be63e..0000000 --- a/Documentation/devicetree/bindings/sound/max98925.txt +++ /dev/null @@ -1,22 +0,0 @@ -max98925 audio CODEC - -This device supports I2C. - -Required properties: - - - compatible : "maxim,max98925" - - - vmon-slot-no : slot number used to send voltage information - - - imon-slot-no : slot number used to send current information - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98925@1a { - compatible = "maxim,max98925"; - vmon-slot-no = <0>; - imon-slot-no = <2>; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt deleted file mode 100644 index 0b7f4e4..0000000 --- a/Documentation/devicetree/bindings/sound/max98926.txt +++ /dev/null @@ -1,32 +0,0 @@ -max98926 audio CODEC - -This device supports I2C. - -Required properties: - - - compatible : "maxim,max98926" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - - - imon-slot-no : slot number used to send current information - - - interleave-mode : When using two MAX98926 in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98926@1a { - compatible = "maxim,max98926"; - vmon-slot-no = <0>; - imon-slot-no = <2>; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt new file mode 100644 index 0000000..f617159 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -0,0 +1,41 @@ +Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : should be one of the following + - "maxim,max98925" + - "maxim,max98926" + - "maxim,max98927" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - imon-slot-no : slot number used to send current information + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - interleave-mode : When using two MAX9892X in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + Range : 0 (off), 1 (on), Default : 0 + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98927@3a { + compatible = "maxim,max98927"; + vmon-slot-no = <0>; + imon-slot-no = <1>; + interleave-mode = <0>; + reg = <0x3a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e49e9da..cbdc8aa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -89,6 +89,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C + select SND_SOC_MAX98927 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C @@ -588,6 +589,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX98926 tristate +config SND_SOC_MAX98927 + tristate "Maxim Integrated MAX98927 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1796cb9..187a639 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o +snd-soc-max98927-objs := max98927.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o @@ -313,6 +314,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o +obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c new file mode 100644 index 0000000..b5ee294 --- /dev/null +++ b/sound/soc/codecs/max98927.c @@ -0,0 +1,841 @@ +/* + * max98927.c -- MAX98927 ALSA Soc Audio driver + * + * Copyright (C) 2016 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/acpi.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98927.h" + +static struct reg_default max98927_reg[] = { + {MAX98927_R0001_INT_RAW1, 0x00}, + {MAX98927_R0002_INT_RAW2, 0x00}, + {MAX98927_R0003_INT_RAW3, 0x00}, + {MAX98927_R0004_INT_STATE1, 0x00}, + {MAX98927_R0005_INT_STATE2, 0x00}, + {MAX98927_R0006_INT_STATE3, 0x00}, + {MAX98927_R0007_INT_FLAG1, 0x00}, + {MAX98927_R0008_INT_FLAG2, 0x00}, + {MAX98927_R0009_INT_FLAG3, 0x00}, + {MAX98927_R000A_INT_EN1, 0x00}, + {MAX98927_R000B_INT_EN2, 0x00}, + {MAX98927_R000C_INT_EN3, 0x00}, + {MAX98927_R000D_INT_FLAG_CLR1, 0x00}, + {MAX98927_R000E_INT_FLAG_CLR2, 0x00}, + {MAX98927_R000F_INT_FLAG_CLR3, 0x00}, + {MAX98927_R0010_IRQ_CTRL, 0x00}, + {MAX98927_R0011_CLK_MON, 0x00}, + {MAX98927_R0012_WDOG_CTRL, 0x00}, + {MAX98927_R0013_WDOG_RST, 0x00}, + {MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH, 0x00}, + {MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH, 0x00}, + {MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS, 0x00}, + {MAX98927_R0017_PIN_CFG, 0x55}, + {MAX98927_R0018_PCM_RX_EN_A, 0x00}, + {MAX98927_R0019_PCM_RX_EN_B, 0x00}, + {MAX98927_R001A_PCM_TX_EN_A, 0x00}, + {MAX98927_R001B_PCM_TX_EN_B, 0x00}, + {MAX98927_R001C_PCM_TX_HIZ_CTRL_A, 0x00}, + {MAX98927_R001D_PCM_TX_HIZ_CTRL_B, 0x00}, + {MAX98927_R001E_PCM_TX_CH_SRC_A, 0x00}, + {MAX98927_R001F_PCM_TX_CH_SRC_B, 0x00}, + {MAX98927_R0020_PCM_MODE_CFG, 0x40}, + {MAX98927_R0021_PCM_MASTER_MODE, 0x00}, + {MAX98927_R0022_PCM_CLK_SETUP, 0x22}, + {MAX98927_R0023_PCM_SR_SETUP1, 0x00}, + {MAX98927_R0024_PCM_SR_SETUP2, 0x00}, + {MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, 0x00}, + {MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, 0x00}, + {MAX98927_R0027_ICC_RX_EN_A, 0x00}, + {MAX98927_R0028_ICC_RX_EN_B, 0x00}, + {MAX98927_R002B_ICC_TX_EN_A, 0x00}, + {MAX98927_R002C_ICC_TX_EN_B, 0x00}, + {MAX98927_R002E_ICC_HIZ_MANUAL_MODE, 0x00}, + {MAX98927_R002F_ICC_TX_HIZ_EN_A, 0x00}, + {MAX98927_R0030_ICC_TX_HIZ_EN_B, 0x00}, + {MAX98927_R0031_ICC_LNK_EN, 0x00}, + {MAX98927_R0032_PDM_TX_EN, 0x00}, + {MAX98927_R0033_PDM_TX_HIZ_CTRL, 0x00}, + {MAX98927_R0034_PDM_TX_CTRL, 0x00}, + {MAX98927_R0035_PDM_RX_CTRL, 0x00}, + {MAX98927_R0036_AMP_VOL_CTRL, 0x00}, + {MAX98927_R0037_AMP_DSP_CFG, 0x02}, + {MAX98927_R0038_TONE_GEN_DC_CFG, 0x00}, + {MAX98927_R0039_DRE_CTRL, 0x01}, + {MAX98927_R003A_AMP_EN, 0x00}, + {MAX98927_R003B_SPK_SRC_SEL, 0x00}, + {MAX98927_R003C_SPK_GAIN, 0x00}, + {MAX98927_R003D_SSM_CFG, 0x01}, + {MAX98927_R003E_MEAS_EN, 0x00}, + {MAX98927_R003F_MEAS_DSP_CFG, 0x04}, + {MAX98927_R0040_BOOST_CTRL0, 0x00}, + {MAX98927_R0041_BOOST_CTRL3, 0x00}, + {MAX98927_R0042_BOOST_CTRL1, 0x00}, + {MAX98927_R0043_MEAS_ADC_CFG, 0x00}, + {MAX98927_R0044_MEAS_ADC_BASE_MSB, 0x00}, + {MAX98927_R0045_MEAS_ADC_BASE_LSB, 0x00}, + {MAX98927_R0046_ADC_CH0_DIVIDE, 0x00}, + {MAX98927_R0047_ADC_CH1_DIVIDE, 0x00}, + {MAX98927_R0048_ADC_CH2_DIVIDE, 0x00}, + {MAX98927_R0049_ADC_CH0_FILT_CFG, 0x00}, + {MAX98927_R004A_ADC_CH1_FILT_CFG, 0x00}, + {MAX98927_R004B_ADC_CH2_FILT_CFG, 0x00}, + {MAX98927_R004C_MEAS_ADC_CH0_READ, 0x00}, + {MAX98927_R004D_MEAS_ADC_CH1_READ, 0x00}, + {MAX98927_R004E_MEAS_ADC_CH2_READ, 0x00}, + {MAX98927_R0051_BROWNOUT_STATUS, 0x00}, + {MAX98927_R0052_BROWNOUT_EN, 0x00}, + {MAX98927_R0053_BROWNOUT_INFINITE_HOLD, 0x00}, + {MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR, 0x00}, + {MAX98927_R0055_BROWNOUT_LVL_HOLD, 0x00}, + {MAX98927_R005A_BROWNOUT_LVL1_THRESH, 0x00}, + {MAX98927_R005B_BROWNOUT_LVL2_THRESH, 0x00}, + {MAX98927_R005C_BROWNOUT_LVL3_THRESH, 0x00}, + {MAX98927_R005D_BROWNOUT_LVL4_THRESH, 0x00}, + {MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS, 0x00}, + {MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL, 0x00}, + {MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL, 0x00}, + {MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE, 0x00}, + {MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT, 0x00}, + {MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1, 0x00}, + {MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2, 0x00}, + {MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3, 0x00}, + {MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT, 0x00}, + {MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1, 0x00}, + {MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2, 0x00}, + {MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3, 0x00}, + {MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT, 0x00}, + {MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1, 0x00}, + {MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2, 0x00}, + {MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3, 0x00}, + {MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT, 0x00}, + {MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, 0x00}, + {MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2, 0x00}, + {MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3, 0x00}, + {MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0x00}, + {MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY, 0x00}, + {MAX98927_R0084_ENV_TRACK_REL_RATE, 0x00}, + {MAX98927_R0085_ENV_TRACK_HOLD_RATE, 0x00}, + {MAX98927_R0086_ENV_TRACK_CTRL, 0x00}, + {MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, 0x00}, + {MAX98927_R00FF_GLOBAL_SHDN, 0x00}, + {MAX98927_R0100_SOFT_RESET, 0x00}, + {MAX98927_R01FF_REV_ID, 0x40}, +}; + +static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int mode = 0; + unsigned int format = 0; + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + mode = MAX98927_PCM_MASTER_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98927->master = true; + mode = MAX98927_PCM_MASTER_MODE_MASTER; + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MASK, + mode); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + max98927->iface |= SND_SOC_DAIFMT_I2S; + format = MAX98927_PCM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + max98927->iface |= SND_SOC_DAIFMT_LEFT_J; + format = MAX98927_PCM_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_PDM: + max98927->iface |= SND_SOC_DAIFMT_PDM; + break; + default: + return -EINVAL; + } + + /* pcm channel configuration */ + if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + regmap_update_bits(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN); + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_FORMAT_MASK, + format << MAX98927_PCM_MODE_CFG_FORMAT_SHIFT); + + regmap_update_bits(max98927->regmap, + MAX98927_R003B_SPK_SRC_SEL, + MAX98927_SPK_SRC_MASK, 0); + + } else + regmap_update_bits(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + /* pdm channel configuration */ + if (max98927->iface & SND_SOC_DAIFMT_PDM) { + regmap_update_bits(max98927->regmap, + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 1); + + regmap_update_bits(max98927->regmap, + MAX98927_R003B_SPK_SRC_SEL, + MAX98927_SPK_SRC_MASK, 3); + } else + regmap_update_bits(max98927->regmap, + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + return 0; +} + +/* codec MCLK rate in master mode */ +static const int rate_table[] = { + 5644800, 6000000, 6144000, 6500000, + 9600000, 11289600, 12000000, 12288000, + 13000000, 19200000, +}; + +static int max98927_set_clock(struct max98927_priv *max98927, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_codec *codec = max98927->codec; + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98927->ch_size; + int value; + + if (max98927->master) { + int i; + /* match rate to closest value */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i] >= max98927->sysclk) + break; + } + if (i == ARRAY_SIZE(rate_table)) { + dev_err(codec->dev, "failed to find proper clock rate.\n"); + return -EINVAL; + } + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MCLK_MASK, + i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); + } + + switch (blr_clk_ratio) { + case 32: + value = 2; + break; + case 48: + value = 3; + break; + case 64: + value = 4; + break; + default: + return -EINVAL; + } + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); + return 0; +} + +static int max98927_dai_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 max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d", + params_format(params)); + goto err; + } + + max98927->ch_size = snd_pcm_format_width(params_format(params)); + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(codec->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_8000; + break; + case 11025: + sampling_rate = MAX98927_PCM_SR_SET1_SR_11025; + break; + case 12000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_12000; + break; + case 16000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_16000; + break; + case 22050: + sampling_rate = MAX98927_PCM_SR_SET1_SR_22050; + break; + case 24000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_24000; + break; + case 32000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_32000; + break; + case 44100: + sampling_rate = MAX98927_PCM_SR_SET1_SR_44100; + break; + case 48000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_48000; + break; + default: + dev_err(codec->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98927->regmap, + MAX98927_R0023_PCM_SR_SETUP1, + MAX98927_PCM_SR_SET1_SR_MASK, + sampling_rate); + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_SR_MASK, + sampling_rate << MAX98927_PCM_SR_SET2_SR_SHIFT); + + /* set sampling rate of IV */ + if (max98927->interleave_mode && + sampling_rate > MAX98927_PCM_SR_SET1_SR_16000) + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate - 3); + else + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate); + return max98927_set_clock(max98927, params); +err: + return -EINVAL; +} + +#define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->sysclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops max98927_dai_ops = { + .set_sysclk = max98927_dai_set_sysclk, + .set_fmt = max98927_dai_set_fmt, + .hw_params = max98927_dai_hw_params, +}; + +static int max98927_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 1); + /* enable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN); + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 1); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 0); + /* disable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, 0); + break; + default: + return 0; + } + return 0; +} + +static const char * const max98927_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT, + 3, max98927_switch_text); + +static const struct snd_kcontrol_new max98927_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_soc_dapm_widget max98927_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98927_R003A_AMP_EN, + 0, 0, max98927_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, + &max98927_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static DECLARE_TLV_DB_SCALE(max98927_spk_tlv, 300, 300, 0); +static DECLARE_TLV_DB_SCALE(max98927_digital_tlv, -1600, 25, 0); + +static bool max98927_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0028_ICC_RX_EN_B: + case MAX98927_R002B_ICC_TX_EN_A ... MAX98927_R002C_ICC_TX_EN_B: + case MAX98927_R002E_ICC_HIZ_MANUAL_MODE + ... MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS + ... MAX98927_R0055_BROWNOUT_LVL_HOLD: + case MAX98927_R005A_BROWNOUT_LVL1_THRESH + ... MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE: + case MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT + ... MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R00FF_GLOBAL_SHDN: + case MAX98927_R0100_SOFT_RESET: + case MAX98927_R01FF_REV_ID: + return true; + default: + return false; + } +}; + +static bool max98927_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + return true; + default: + return false; + } +} + +static const char * const max98927_boost_voltage_text[] = { + "6.5V", "6.625V", "6.75V", "6.875V", "7V", "7.125V", "7.25V", "7.375V", + "7.5V", "7.625V", "7.75V", "7.875V", "8V", "8.125V", "8.25V", "8.375V", + "8.5V", "8.625V", "8.75V", "8.875V", "9V", "9.125V", "9.25V", "9.375V", + "9.5V", "9.625V", "9.75V", "9.875V", "10V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_boost_voltage, + MAX98927_R0040_BOOST_CTRL0, 0, + max98927_boost_voltage_text); + +static const char * const max98927_current_limit_text[] = { + "1.00A", "1.10A", "1.20A", "1.30A", "1.40A", "1.50A", "1.60A", "1.70A", + "1.80A", "1.90A", "2.00A", "2.10A", "2.20A", "2.30A", "2.40A", "2.50A", + "2.60A", "2.70A", "2.80A", "2.90A", "3.00A", "3.10A", "3.20A", "3.30A", + "3.40A", "3.50A", "3.60A", "3.70A", "3.80A", "3.90A", "4.00A", "4.10A" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_current_limit, + MAX98927_R0042_BOOST_CTRL1, 1, + max98927_current_limit_text); + +static const struct snd_kcontrol_new max98927_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, + 0, 6, 0, + max98927_spk_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98927_R0036_AMP_VOL_CTRL, + 0, (1<<MAX98927_AMP_VOL_WIDTH)-1, 0, + max98927_digital_tlv), + SOC_SINGLE("Amp DSP Switch", MAX98927_R0052_BROWNOUT_EN, + MAX98927_BROWNOUT_DSP_SHIFT, 1, 0), + SOC_SINGLE("Ramp Switch", MAX98927_R0037_AMP_DSP_CFG, + MAX98927_AMP_DSP_CFG_RMP_SHIFT, 1, 0), + SOC_SINGLE("DRE Switch", MAX98927_R0039_DRE_CTRL, + MAX98927_DRE_EN_SHIFT, 1, 0), + SOC_SINGLE("Volume Location Switch", MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), + SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), + SOC_ENUM("Current Limit", max98927_current_limit), +}; + +static const struct snd_soc_dapm_route max98927_audio_map[] = { + {"Amp Enable", NULL, "DAI_OUT"}, + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98927_dai[] = { + { + .name = "max98927-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .ops = &max98927_dai_ops, + } +}; + +static int max98927_probe(struct snd_soc_codec *codec) +{ + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->codec = codec; + codec->control_data = max98927->regmap; + codec->cache_bypass = 1; + + /* Software Reset */ + regmap_write(max98927->regmap, + MAX98927_R0100_SOFT_RESET, MAX98927_SOFT_RESET); + + /* IV default slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + 0x80); + regmap_write(max98927->regmap, + MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, + 0x1); + /* Set inital volume (+13dB) */ + regmap_write(max98927->regmap, + MAX98927_R0036_AMP_VOL_CTRL, + 0x38); + regmap_write(max98927->regmap, + MAX98927_R003C_SPK_GAIN, + 0x05); + /* Enable DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R0037_AMP_DSP_CFG, + 0x03); + /* Enable IMON VMON DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R003F_MEAS_DSP_CFG, + 0xF7); + /* Boost Output Voltage & Current limit */ + regmap_write(max98927->regmap, + MAX98927_R0040_BOOST_CTRL0, + 0x1C); + regmap_write(max98927->regmap, + MAX98927_R0042_BOOST_CTRL1, + 0x3E); + /* Measurement ADC config */ + regmap_write(max98927->regmap, + MAX98927_R0043_MEAS_ADC_CFG, + 0x04); + regmap_write(max98927->regmap, + MAX98927_R0044_MEAS_ADC_BASE_MSB, + 0x00); + regmap_write(max98927->regmap, + MAX98927_R0045_MEAS_ADC_BASE_LSB, + 0x24); + /* Brownout Level */ + regmap_write(max98927->regmap, + MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, + 0x06); + /* Envelope Tracking configuration */ + regmap_write(max98927->regmap, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, + 0x08); + regmap_write(max98927->regmap, + MAX98927_R0086_ENV_TRACK_CTRL, + 0x01); + regmap_write(max98927->regmap, + MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, + 0x10); + + /* voltage, current slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001E_PCM_TX_CH_SRC_A, + (max98927->i_l_slot<<MAX98927_PCM_TX_CH_SRC_A_I_SHIFT| + max98927->v_l_slot)&0xFF); + + if (max98927->v_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->v_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->v_l_slot, + 1 << max98927->v_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->v_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->v_l_slot - 8), + 1 << (max98927->v_l_slot - 8)); + } + + if (max98927->i_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->i_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->i_l_slot, + 1 << max98927->i_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->i_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->i_l_slot - 8), + 1 << (max98927->i_l_slot - 8)); + } + + /* Set interleave mode */ + if (max98927->interleave_mode) + regmap_update_bits(max98927->regmap, + MAX98927_R001F_PCM_TX_CH_SRC_B, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK); + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_dev_max98927 = { + .probe = max98927_probe, + .component_driver = { + .controls = max98927_snd_controls, + .num_controls = ARRAY_SIZE(max98927_snd_controls), + .dapm_widgets = max98927_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98927_dapm_widgets), + .dapm_routes = max98927_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98927_audio_map), + }, +}; + +static const struct regmap_config max98927_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98927_R01FF_REV_ID, + .reg_defaults = max98927_reg, + .num_reg_defaults = ARRAY_SIZE(max98927_reg), + .readable_reg = max98927_readable_register, + .volatile_reg = max98927_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static void max98927_slot_config(struct i2c_client *i2c, + struct max98927_priv *max98927) +{ + int value; + + if (!of_property_read_u32(i2c->dev.of_node, + "vmon-slot-no", &value)) + max98927->v_l_slot = value & 0xF; + else + max98927->v_l_slot = 0; + if (!of_property_read_u32(i2c->dev.of_node, + "imon-slot-no", &value)) + max98927->i_l_slot = value & 0xF; + else + max98927->i_l_slot = 1; +} + +static int max98927_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + + int ret = 0, value; + int reg = 0; + struct max98927_priv *max98927 = NULL; + + max98927 = devm_kzalloc(&i2c->dev, + sizeof(*max98927), GFP_KERNEL); + + if (!max98927) { + ret = -ENOMEM; + return ret; + } + i2c_set_clientdata(i2c, max98927); + + /* update interleave mode info */ + if (!of_property_read_u32(i2c->dev.of_node, + "interleave_mode", &value)) { + if (value > 0) + max98927->interleave_mode = 1; + else + max98927->interleave_mode = 0; + } else + max98927->interleave_mode = 0; + + /* regmap initialization */ + max98927->regmap + = devm_regmap_init_i2c(i2c, &max98927_regmap); + if (IS_ERR(max98927->regmap)) { + ret = PTR_ERR(max98927->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + /* Check Revision ID */ + ret = regmap_read(max98927->regmap, + MAX98927_R01FF_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, + "Failed to read: 0x%02X\n", MAX98927_R01FF_REV_ID); + return ret; + } + dev_info(&i2c->dev, "MAX98927 revisionID: 0x%02X\n", reg); + + /* voltage/current slot configuration */ + max98927_slot_config(i2c, max98927); + + /* codec registeration */ + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98927, + max98927_dai, ARRAY_SIZE(max98927_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int max98927_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98927_i2c_id[] = { + { "max98927", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98927_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98927_of_match[] = { + { .compatible = "maxim,max98927", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98927_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98927_acpi_match[] = { + { "MX98927", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98927_acpi_match); +#endif + +static struct i2c_driver max98927_i2c_driver = { + .driver = { + .name = "max98927", + .of_match_table = of_match_ptr(max98927_of_match), + .acpi_match_table = ACPI_PTR(max98927_acpi_match), + .pm = NULL, + }, + .probe = max98927_i2c_probe, + .remove = max98927_i2c_remove, + .id_table = max98927_i2c_id, +}; + +module_i2c_driver(max98927_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98927 driver"); +MODULE_AUTHOR("Ryan Lee <ryans.lee(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h new file mode 100644 index 0000000..ece6a60 --- /dev/null +++ b/sound/soc/codecs/max98927.h @@ -0,0 +1,272 @@ +/* + * max98927.h -- MAX98927 ALSA Soc Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ +#ifndef _MAX98927_H +#define _MAX98927_H + +/* Register Values */ +#define MAX98927_R0001_INT_RAW1 0x0001 +#define MAX98927_R0002_INT_RAW2 0x0002 +#define MAX98927_R0003_INT_RAW3 0x0003 +#define MAX98927_R0004_INT_STATE1 0x0004 +#define MAX98927_R0005_INT_STATE2 0x0005 +#define MAX98927_R0006_INT_STATE3 0x0006 +#define MAX98927_R0007_INT_FLAG1 0x0007 +#define MAX98927_R0008_INT_FLAG2 0x0008 +#define MAX98927_R0009_INT_FLAG3 0x0009 +#define MAX98927_R000A_INT_EN1 0x000A +#define MAX98927_R000B_INT_EN2 0x000B +#define MAX98927_R000C_INT_EN3 0x000C +#define MAX98927_R000D_INT_FLAG_CLR1 0x000D +#define MAX98927_R000E_INT_FLAG_CLR2 0x000E +#define MAX98927_R000F_INT_FLAG_CLR3 0x000F +#define MAX98927_R0010_IRQ_CTRL 0x0010 +#define MAX98927_R0011_CLK_MON 0x0011 +#define MAX98927_R0012_WDOG_CTRL 0x0012 +#define MAX98927_R0013_WDOG_RST 0x0013 +#define MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH 0x0014 +#define MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH 0x0015 +#define MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS 0x0016 +#define MAX98927_R0017_PIN_CFG 0x0017 +#define MAX98927_R0018_PCM_RX_EN_A 0x0018 +#define MAX98927_R0019_PCM_RX_EN_B 0x0019 +#define MAX98927_R001A_PCM_TX_EN_A 0x001A +#define MAX98927_R001B_PCM_TX_EN_B 0x001B +#define MAX98927_R001C_PCM_TX_HIZ_CTRL_A 0x001C +#define MAX98927_R001D_PCM_TX_HIZ_CTRL_B 0x001D +#define MAX98927_R001E_PCM_TX_CH_SRC_A 0x001E +#define MAX98927_R001F_PCM_TX_CH_SRC_B 0x001F +#define MAX98927_R0020_PCM_MODE_CFG 0x0020 +#define MAX98927_R0021_PCM_MASTER_MODE 0x0021 +#define MAX98927_R0022_PCM_CLK_SETUP 0x0022 +#define MAX98927_R0023_PCM_SR_SETUP1 0x0023 +#define MAX98927_R0024_PCM_SR_SETUP2 0x0024 +#define MAX98927_R0025_PCM_TO_SPK_MONOMIX_A 0x0025 +#define MAX98927_R0026_PCM_TO_SPK_MONOMIX_B 0x0026 +#define MAX98927_R0027_ICC_RX_EN_A 0x0027 +#define MAX98927_R0028_ICC_RX_EN_B 0x0028 +#define MAX98927_R002B_ICC_TX_EN_A 0x002B +#define MAX98927_R002C_ICC_TX_EN_B 0x002C +#define MAX98927_R002E_ICC_HIZ_MANUAL_MODE 0x002E +#define MAX98927_R002F_ICC_TX_HIZ_EN_A 0x002F +#define MAX98927_R0030_ICC_TX_HIZ_EN_B 0x0030 +#define MAX98927_R0031_ICC_LNK_EN 0x0031 +#define MAX98927_R0032_PDM_TX_EN 0x0032 +#define MAX98927_R0033_PDM_TX_HIZ_CTRL 0x0033 +#define MAX98927_R0034_PDM_TX_CTRL 0x0034 +#define MAX98927_R0035_PDM_RX_CTRL 0x0035 +#define MAX98927_R0036_AMP_VOL_CTRL 0x0036 +#define MAX98927_R0037_AMP_DSP_CFG 0x0037 +#define MAX98927_R0038_TONE_GEN_DC_CFG 0x0038 +#define MAX98927_R0039_DRE_CTRL 0x0039 +#define MAX98927_R003A_AMP_EN 0x003A +#define MAX98927_R003B_SPK_SRC_SEL 0x003B +#define MAX98927_R003C_SPK_GAIN 0x003C +#define MAX98927_R003D_SSM_CFG 0x003D +#define MAX98927_R003E_MEAS_EN 0x003E +#define MAX98927_R003F_MEAS_DSP_CFG 0x003F +#define MAX98927_R0040_BOOST_CTRL0 0x0040 +#define MAX98927_R0041_BOOST_CTRL3 0x0041 +#define MAX98927_R0042_BOOST_CTRL1 0x0042 +#define MAX98927_R0043_MEAS_ADC_CFG 0x0043 +#define MAX98927_R0044_MEAS_ADC_BASE_MSB 0x0044 +#define MAX98927_R0045_MEAS_ADC_BASE_LSB 0x0045 +#define MAX98927_R0046_ADC_CH0_DIVIDE 0x0046 +#define MAX98927_R0047_ADC_CH1_DIVIDE 0x0047 +#define MAX98927_R0048_ADC_CH2_DIVIDE 0x0048 +#define MAX98927_R0049_ADC_CH0_FILT_CFG 0x0049 +#define MAX98927_R004A_ADC_CH1_FILT_CFG 0x004A +#define MAX98927_R004B_ADC_CH2_FILT_CFG 0x004B +#define MAX98927_R004C_MEAS_ADC_CH0_READ 0x004C +#define MAX98927_R004D_MEAS_ADC_CH1_READ 0x004D +#define MAX98927_R004E_MEAS_ADC_CH2_READ 0x004E +#define MAX98927_R0051_BROWNOUT_STATUS 0x0051 +#define MAX98927_R0052_BROWNOUT_EN 0x0052 +#define MAX98927_R0053_BROWNOUT_INFINITE_HOLD 0x0053 +#define MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR 0x0054 +#define MAX98927_R0055_BROWNOUT_LVL_HOLD 0x0055 +#define MAX98927_R005A_BROWNOUT_LVL1_THRESH 0x005A +#define MAX98927_R005B_BROWNOUT_LVL2_THRESH 0x005B +#define MAX98927_R005C_BROWNOUT_LVL3_THRESH 0x005C +#define MAX98927_R005D_BROWNOUT_LVL4_THRESH 0x005D +#define MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS 0x005E +#define MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL 0x005F +#define MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL 0x0060 +#define MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE 0x0061 +#define MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT 0x0072 +#define MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1 0x0073 +#define MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2 0x0074 +#define MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3 0x0075 +#define MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT 0x0076 +#define MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1 0x0077 +#define MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2 0x0078 +#define MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3 0x0079 +#define MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT 0x007A +#define MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1 0x007B +#define MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2 0x007C +#define MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3 0x007D +#define MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT 0x007E +#define MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1 0x007F +#define MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2 0x0080 +#define MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3 0x0081 +#define MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM 0x0082 +#define MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY 0x0083 +#define MAX98927_R0084_ENV_TRACK_REL_RATE 0x0084 +#define MAX98927_R0085_ENV_TRACK_HOLD_RATE 0x0085 +#define MAX98927_R0086_ENV_TRACK_CTRL 0x0086 +#define MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ 0x0087 +#define MAX98927_R00FF_GLOBAL_SHDN 0x00FF +#define MAX98927_R0100_SOFT_RESET 0x0100 +#define MAX98927_R01FF_REV_ID 0x01FF + +/* MAX98927_R0018_PCM_RX_EN_A */ +#define MAX98927_PCM_RX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_RX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_RX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_RX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_RX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_RX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_RX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_RX_CH7_EN (0x1 << 7) + +/* MAX98927_R001A_PCM_TX_EN_A */ +#define MAX98927_PCM_TX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_TX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_TX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_TX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_TX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_TX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_TX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_TX_CH7_EN (0x1 << 7) + +/* MAX98927_R001E_PCM_TX_CH_SRC_A */ +#define MAX98927_PCM_TX_CH_SRC_A_V_SHIFT (0) +#define MAX98927_PCM_TX_CH_SRC_A_I_SHIFT (4) + +/* MAX98927_R001F_PCM_TX_CH_SRC_B */ +#define MAX98927_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 5) + +/* MAX98927_R0020_PCM_MODE_CFG */ +#define MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 2) +#define MAX98927_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) +#define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) +#define MAX98927_PCM_FORMAT_I2S (0x0 << 0) +#define MAX98927_PCM_FORMAT_LJ (0x1 << 0) + +#define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98927_R0021_PCM_MASTER_MODE */ +#define MAX98927_PCM_MASTER_MODE_MASK (0x3 << 0) +#define MAX98927_PCM_MASTER_MODE_SLAVE (0x0 << 0) +#define MAX98927_PCM_MASTER_MODE_MASTER (0x3 << 0) + +#define MAX98927_PCM_MASTER_MODE_MCLK_MASK (0xF << 2) +#define MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT (2) + +/* MAX98927_R0022_PCM_CLK_SETUP */ +#define MAX98927_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98927_R0023_PCM_SR_SETUP1 */ +#define MAX98927_PCM_SR_SET1_SR_MASK (0xF << 0) + +#define MAX98927_PCM_SR_SET1_SR_8000 (0x0 << 0) +#define MAX98927_PCM_SR_SET1_SR_11025 (0x1 << 0) +#define MAX98927_PCM_SR_SET1_SR_12000 (0x2 << 0) +#define MAX98927_PCM_SR_SET1_SR_16000 (0x3 << 0) +#define MAX98927_PCM_SR_SET1_SR_22050 (0x4 << 0) +#define MAX98927_PCM_SR_SET1_SR_24000 (0x5 << 0) +#define MAX98927_PCM_SR_SET1_SR_32000 (0x6 << 0) +#define MAX98927_PCM_SR_SET1_SR_44100 (0x7 << 0) +#define MAX98927_PCM_SR_SET1_SR_48000 (0x8 << 0) + +/* MAX98927_R0024_PCM_SR_SETUP2 */ +#define MAX98927_PCM_SR_SET2_SR_MASK (0xF << 4) +#define MAX98927_PCM_SR_SET2_SR_SHIFT (4) +#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xf << 0) + +/* MAX98927_R0025_PCM_TO_SPK_MONOMIX_A */ +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT (6) + +/* MAX98927_R0035_PDM_RX_CTRL */ +#define MAX98927_PDM_RX_EN_MASK (0x1 << 0) + +/* MAX98927_R0036_AMP_VOL_CTRL */ +#define MAX98927_AMP_VOL_SEL (0x1 << 7) +#define MAX98927_AMP_VOL_SEL_WIDTH (1) +#define MAX98927_AMP_VOL_SEL_SHIFT (7) +#define MAX98927_AMP_VOL_MASK (0x7f << 0) +#define MAX98927_AMP_VOL_WIDTH (7) +#define MAX98927_AMP_VOL_SHIFT (0) + +/* MAX98927_R0037_AMP_DSP_CFG */ +#define MAX98927_AMP_DSP_CFG_DCBLK_EN (0x1 << 0) +#define MAX98927_AMP_DSP_CFG_DITH_EN (0x1 << 1) +#define MAX98927_AMP_DSP_CFG_RMP_BYPASS (0x1 << 4) +#define MAX98927_AMP_DSP_CFG_DAC_INV (0x1 << 5) +#define MAX98927_AMP_DSP_CFG_RMP_SHIFT (4) + +/* MAX98927_R0039_DRE_CTRL */ +#define MAX98927_DRE_CTRL_DRE_EN (0x1 << 0) +#define MAX98927_DRE_EN_SHIFT 0x1 + +/* MAX98927_R003A_AMP_EN */ +#define MAX98927_AMP_EN_MASK (0x1 << 0) + +/* MAX98927_R003B_SPK_SRC_SEL */ +#define MAX98927_SPK_SRC_MASK (0x3 << 0) + +/* MAX98927_R003C_SPK_GAIN */ +#define MAX98927_SPK_PCM_GAIN_MASK (0x7 << 0) +#define MAX98927_SPK_PDM_GAIN_MASK (0x7 << 4) +#define MAX98927_SPK_GAIN_WIDTH (3) + +/* MAX98927_R003E_MEAS_EN */ +#define MAX98927_MEAS_V_EN (0x1 << 0) +#define MAX98927_MEAS_I_EN (0x1 << 1) + +/* MAX98927_R0040_BOOST_CTRL0 */ +#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1f << 0) +#define MAX98927_BOOST_CTRL0_PVDD_MASK (0x1 << 7) +#define MAX98927_BOOST_CTRL0_PVDD_EN_SHIFT (7) + +/* MAX98927_R0052_BROWNOUT_EN */ +#define MAX98927_BROWNOUT_BDE_EN (0x1 << 0) +#define MAX98927_BROWNOUT_AMP_EN (0x1 << 1) +#define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) +#define MAX98927_BROWNOUT_DSP_SHIFT (2) + +/* MAX98927_R0100_SOFT_RESET */ +#define MAX98927_SOFT_RESET (0x1 << 0) + +/* MAX98927_R00FF_GLOBAL_SHDN */ +#define MAX98927_GLOBAL_EN_MASK (0x1 << 0) + +struct max98927_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98927_pdata *pdata; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int v_l_slot; + unsigned int i_l_slot; + bool interleave_mode; + unsigned int ch_size; + unsigned int rate; + unsigned int iface; + unsigned int master; + unsigned int digital_gain; +}; +#endif -- 2.7.4

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[alsa-devel] [resend][PATCH v5 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 03 Apr '17

03 Apr '17

Hi Rob, Mark 2 weeks passed. I will resend this patch-set. These are v5 of OF-graph base audio card patch-set. It fixed small things which was pointed by Rob. 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH 0/3] Improve SKL+ position reporting for legacy HD-audio
by Takashi Iwai 03 Apr '17

03 Apr '17

Hi, this is a port of SKL-specific hacks to the legacy HD-audio driver for improving the PCM position reporting that has been applied to ASoC driver so far. Takashi === Takashi Iwai (3): ALSA: hda - Avoid tricky macros ALSA: hda - Move SKL+ vendor specific register definitions to hda_register.h ALSA: hda - Improved position reporting on SKL+ include/sound/hda_register.h | 22 ++++++++++++++++++++-- include/sound/hdaudio.h | 28 ++++++++++++++++++---------- sound/hda/hdac_controller.c | 2 +- sound/hda/hdac_stream.c | 4 ++-- sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl.h | 21 --------------------- 6 files changed, 77 insertions(+), 40 deletions(-) -- 2.11.1

2 7

[alsa-devel] [PATCH v2] ALSA: hda/ca0132: Limit values for chip addresses to 32-bit
by Matthias Kaehlcke 01 Apr '17

01 Apr '17

With the previous unsigned long value clang generates warnings like this: sound/pci/hda/patch_ca0132.c:860:37: error: implicit conversion from 'unsigned long' to 'u32' (aka 'unsigned int') changes value from 18446744073709551615 to 4294967295 [-Werror,-Wconstant-conversion] spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx; ~ ^~~~ Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - Remove unnecessary cast to u32 sound/pci/hda/patch_ca0132.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 07a9deb17477..2023fe472be1 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -857,7 +857,7 @@ static int chipio_write_address(struct hda_codec *codec, chip_addx >> 16); } - spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx; + spec->curr_chip_addx = (res < 0) ? ~0U : chip_addx; return res; } @@ -882,7 +882,7 @@ static int chipio_write_data(struct hda_codec *codec, unsigned int data) /*If no error encountered, automatically increment the address as per chip behaviour*/ spec->curr_chip_addx = (res != -EIO) ? - (spec->curr_chip_addx + 4) : ~0UL; + (spec->curr_chip_addx + 4) : ~0U; return res; } @@ -933,7 +933,7 @@ static int chipio_read_data(struct hda_codec *codec, unsigned int *data) /*If no error encountered, automatically increment the address as per chip behaviour*/ spec->curr_chip_addx = (res != -EIO) ? - (spec->curr_chip_addx + 4) : ~0UL; + (spec->curr_chip_addx + 4) : ~0U; return res; } -- 2.12.2.564.g063fe858b8-goog

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix channel swap on playback start
by Fabio Estevam 01 Apr '17

01 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit f8fdf5375e2005f2 ("ASoC: fsl-ssi: add SSIEN errata work around") implemented the workaround for the following erratum found on i.MX35 errata document: ENGcm06222: SSI:Transmission does not take place in bit length early frame sync configuration and also for ENGcm06222 from the same document. However it has been only applied for AC97 mode. Apply it to I2S mode as well so that it can fix audio channel swap during playback start. The channel swap can be noticed in about 10% of the times an audio track starts. With the recommended workaround in place no more channel swap happened after running audio start/stop sequence in more than 2000 times. Tested on a mx6dl-wandboard. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fde08660..17f92b8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -93,6 +93,9 @@ CCSR_SSI_SIER_TLS_EN | CCSR_SSI_SIER_TFS_EN | \ CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TFRC_EN) +#define FSLSSI_SSIEN_WORKAROUND (CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | \ + CCSR_SSI_SCR_RE) + enum fsl_ssi_type { FSL_SSI_MCP8610, FSL_SSI_MX21, @@ -559,7 +562,8 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, int i; int max_loop = 100; regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN); + FSLSSI_SSIEN_WORKAROUND, + FSLSSI_SSIEN_WORKAROUND); for (i = 0; i < max_loop; i++) { u32 sfcsr; regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr); @@ -650,8 +654,7 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) * codec before a stream is started. */ regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE, - CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE); + FSLSSI_SSIEN_WORKAROUND, FSLSSI_SSIEN_WORKAROUND); regmap_write(regs, CCSR_SSI_SOR, CCSR_SSI_SOR_WAIT(3)); } -- 2.7.4

2 3

[alsa-devel] [PATCH] ALSA: hda/ca0132: Limit values for chip addresses to 32-bit
by Matthias Kaehlcke 01 Apr '17

01 Apr '17

With the previous unsigned long value clang generates warnings like this: sound/pci/hda/patch_ca0132.c:860:37: error: implicit conversion from 'unsigned long' to 'u32' (aka 'unsigned int') changes value from 18446744073709551615 to 4294967295 [-Werror,-Wconstant-conversion] spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx; ~ ^~~~ Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/pci/hda/patch_ca0132.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 11b9b2f17a2e..7175e2b46fc4 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -857,7 +857,7 @@ static int chipio_write_address(struct hda_codec *codec, chip_addx >> 16); } - spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx; + spec->curr_chip_addx = (res < 0) ? (u32)~0U : chip_addx; return res; } @@ -882,7 +882,7 @@ static int chipio_write_data(struct hda_codec *codec, unsigned int data) /*If no error encountered, automatically increment the address as per chip behaviour*/ spec->curr_chip_addx = (res != -EIO) ? - (spec->curr_chip_addx + 4) : ~0UL; + (spec->curr_chip_addx + 4) : (u32)~0U; return res; } @@ -933,7 +933,7 @@ static int chipio_read_data(struct hda_codec *codec, unsigned int *data) /*If no error encountered, automatically increment the address as per chip behaviour*/ spec->curr_chip_addx = (res != -EIO) ? - (spec->curr_chip_addx + 4) : ~0UL; + (spec->curr_chip_addx + 4) : (u32)~0U; return res; } -- 2.12.0.367.g23dc2f6d3c-goog

2 2

Re: [alsa-devel] mono/stereo translation & number of frames per period
by Paul Stone 01 Apr '17

01 Apr '17

I set up the buffer_bytes_max to correspond with a buffer size of 4 kB. For S16_LE and stereo interleaved, that means 1024 frames in the buffer. When it goes above 512, I get a BUG warning from ALSA: [ 241.219184] ALSA sound/core/pcm_lib.c:352 BUG: pcmC3D0p:0, pos = 512, buffer size = 512, period size = 128 I guess I will have to drill down into ALSA and see where that buffer size is coming from. -Paul Stone

1 0

Re: [alsa-devel] mono/stereo translation & number of frames per period
by Paul Stone 01 Apr '17

01 Apr '17

This is becoming increasingly difficult and frustrating. I've tried to work around ALSA by setting up a different chunk size for mono vs. stereo. However, ALSA now wants the total stereo buffer size to be half as large, in accordance with the chunk size. This seems completely arbitrary to me. Why can I not specify the buffer size? I feel like I must be missing something very basic. -Paul Stone

1 0

Re: [alsa-devel] mono/stereo translation & number of frames per period
by Paul Stone 31 Mar '17

31 Mar '17

Hi, I've tried setting the runtime->hw.buffer_bytes_max in the prepare() callback, but that doesn't seem to have any effect. The copy callback continues to exceed the boundary of my monaural buffer. I've tried setting up rules based on the number of channels, but the channels interval seems to have no bearing on the actual number of channels for the substream. According to the documentation written by Takashi Iwai, it should be possible to modify the hw parameters based on the number of channels for the substream. In his example, he uses the channel minimum from the interval. I don't understand how an interval applies to a substream. I would think that an open substream would have a fixed number of channels. When I run speakertest with a single channel, I would think that is not an interval, but rather a fixed number of channels equal to '1'. So, I don't understand the purpose of accepting an interval for channels in a rules function which is intended to make changes based on the number of channels for the substream. Here is the interval structure: struct snd_interval { unsigned int min, max; unsigned int openmin:1, openmax:1, integer:1, empty:1; }; None of those fields provided to my rules function seem to contain the correct number of channels for the current substream. I am using alsa-lib-1.0.27.2-r1, which is fairly old. Best Regards, Paul Stone http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH 2/2] ALSA: timer: Info leak in snd_timer_user_tinterrupt()
by Dan Carpenter 31 Mar '17

31 Mar '17

The "r1" struct has memory holes. We clear it with memset on one path where it is used but not the other. Let's just memset it at the start of the function so it's always safe. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/timer.c b/sound/core/timer.c index 8b9e7943a83b..2f836ca09860 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1277,6 +1277,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, struct timespec tstamp; int prev, append = 0; + memset(&r1, 0, sizeof(r1)); memset(&tstamp, 0, sizeof(tstamp)); spin_lock(&tu->qlock); if ((tu->filter & ((1 << SNDRV_TIMER_EVENT_RESOLUTION) | @@ -1292,7 +1293,6 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, } if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { - memset(&r1, 0, sizeof(r1)); r1.event = SNDRV_TIMER_EVENT_RESOLUTION; r1.tstamp = tstamp; r1.val = resolution;

2 1

[alsa-devel] [PATCH 1/2] ALSA: timer: remove some dead code
by Dan Carpenter 31 Mar '17

31 Mar '17

We just checked "id.card < 0" on the lines before so we know it's not true here. We can delete that check. Also checkpatch.pl complains about some extra curly braces so we may as well fix that while we're at it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/timer.c b/sound/core/timer.c index 6d4fbc439246..8b9e7943a83b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1430,18 +1430,13 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) if (id.card < 0) { id.card = 0; } else { - if (id.card < 0) { - id.card = 0; + if (id.device < 0) { + id.device = 0; } else { - if (id.device < 0) { - id.device = 0; - } else { - if (id.subdevice < 0) { - id.subdevice = 0; - } else { - id.subdevice++; - } - } + if (id.subdevice < 0) + id.subdevice = 0; + else + id.subdevice++; } } list_for_each(p, &snd_timer_list) {

2 1

[alsa-devel] [PATCH] ALSA: emux: stop if copy_from_user() fails
by Dan Carpenter 31 Mar '17

31 Mar '17

If we can't fill the "patch" struct because "count" is too small (it can be as low as 4 bytes) or because copy_from_user() failed, then just return instead of using unintialized data. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index ac75816ada7c..850fab4a8f3b 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -225,9 +225,9 @@ snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, else if (format == SNDRV_OSS_SOUNDFONT_PATCH) { struct soundfont_patch_info patch; if (count < (int)sizeof(patch)) - rc = -EINVAL; + return -EINVAL; if (copy_from_user(&patch, buf, sizeof(patch))) - rc = -EFAULT; + return -EFAULT; if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) rc = snd_soundfont_load(emu->sflist, buf, count, SF_CLIENT_NO(p->chset.port));

2 1

[alsa-devel] [bug report] ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
by Dan Carpenter 31 Mar '17

31 Mar '17

Hello Takashi Sakamoto, The patch 5b59d8098d2a: "ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation" from Dec 9, 2014, leads to the following static checker warning: sound/firewire/oxfw/oxfw-command.c:96 avc_stream_get_format() warn: check that subtract can't underflow 'err - 10' sound/firewire/oxfw/oxfw-command.c 76 /* do transaction and check buf[1-7] are the same against command */ 77 err = fcp_avc_transaction(unit, buf, 12, buf, *len, 78 BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | 79 BIT(6) | BIT(7)); 80 if ((err > 0) && (err < 10)) Assume err is either 0 or 10. 81 err = -EIO; 82 else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ 83 err = -ENOSYS; 84 else if (buf[0] == 0x0a) /* REJECTED */ 85 err = -EINVAL; 86 else if (buf[0] == 0x0b) /* IN TRANSITION */ 87 err = -EAGAIN; 88 /* LIST subfunction has entry ID */ 89 else if ((subfunc == 0xc1) && (buf[10] != eid)) If "err" is 10, doesn't that mean we are reading beyond the end of the loop here? 90 err = -EIO; 91 if (err < 0) 92 goto end; 93 94 /* keep just stream format information */ 95 if (subfunc == 0xc0) { 96 memmove(buf, buf + 10, err - 10); If "err" is 0 then we are going to crash here. 97 *len = err - 10; 98 } else { 99 memmove(buf, buf + 11, err - 11); Basically if "err" is zero we're going to crash one way or another. But if err is 10 and we reach this line then we will crash as well. 100 *len = err - 11; 101 } 102 103 err = 0; 104 end: 105 return err; 106 } regards, dan carpenter

1 0

[alsa-devel] [bug report] ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3
by Dan Carpenter 31 Mar '17

31 Mar '17

Hello Takashi Sakamoto, The patch 5992e30034c4: "ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3" from Mar 22, 2017, leads to the following static checker warning: sound/firewire/motu/motu-protocol-v3.c:104 v3_get_clock_source() warn: mask and shift to zero sound/firewire/motu/motu-protocol-v3.c 90 static int v3_get_clock_source(struct snd_motu *motu, 91 enum snd_motu_clock_source *src) 92 { 93 __be32 reg; 94 u32 data; 95 unsigned int val; 96 int err; 97 98 err = snd_motu_transaction_read(motu, V3_CLOCK_STATUS_OFFSET, &reg, 99 sizeof(reg)); 100 if (err < 0) 101 return err; 102 data = be32_to_cpu(reg); 103 104 val = (data & V3_CLOCK_SOURCE_MASK) >> V3_CLOCK_SOURCE_SHIFT; 0xff >> 8 val is always zero. These are the only places we use V3_CLOCK_SOURCE_MASK and V3_CLOCK_SOURCE_SHIFT so I have no idea what it should be. 105 if (val == 0x00) { 106 *src = SND_MOTU_CLOCK_SOURCE_INTERNAL; 107 } else if (val == 0x01) { 108 *src = SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC; 109 } else if (val == 0x10) { 110 *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; 111 } else if (val == 0x18 || val == 0x19) { 112 err = snd_motu_transaction_read(motu, V3_OPT_IFACE_MODE_OFFSET, 113 &reg, sizeof(reg)); 114 if (err < 0) 115 return err; regards, dan carpenter

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fake also USB device id when alias is given
by Takashi Iwai 31 Mar '17

31 Mar '17

Recently snd-usb-audio driver received a new option, quirk_alias, to allow user to apply the existing quirk for a different device. This works for many quirks as is, but some still need more tune-ups: namely, some quirks check the USB vendor/device IDs in various places, thus it doesn't work as long as the ID is different from the expected one. With this patch, the driver stores the aliased USB ID, so that these rest quirks per device ID are applied. The transition to use the cached USB ID was already done in the past, so what we needed now is only to overwrite chip->usb_id. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index f36cb068dad3..6640277a725b 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -332,6 +332,7 @@ static int snd_usb_audio_dev_free(struct snd_device *device) static int snd_usb_audio_create(struct usb_interface *intf, struct usb_device *dev, int idx, const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id, struct snd_usb_audio **rchip) { struct snd_card *card; @@ -381,8 +382,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, atomic_set(&chip->usage_count, 0); atomic_set(&chip->shutdown, 0); - chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct)); + chip->usb_id = usb_id; INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->midi_list); @@ -569,7 +569,7 @@ static int usb_audio_probe(struct usb_interface *intf, (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { err = snd_usb_audio_create(intf, dev, i, quirk, - &chip); + id, &chip); if (err < 0) goto __error; chip->pm_intf = intf; -- 2.11.1

2 2

[alsa-devel] [PATCH] ALSA: hda - fix a problem for lineout on a Dell AIO machine
by Hui Wang 31 Mar '17

31 Mar '17

On this Dell AIO machine, the lineout jack does not work. We found the pin 0x1a is assigned to lineout on this machine, and in the past, we applied ALC298_FIXUP_DELL1_MIC_NO_PRESENCE to fix the heaset-set mic problem for this machine, this fixup will redefine the pin 0x1a to headphone-mic, as a result the lineout doesn't work anymore. After consulting with Dell, they told us this machine doesn't support microphone via headset jack, so we add a new fixup which only defines the pin 0x18 as the headset-mic. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 634e64a..1bf3d67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4878,6 +4878,7 @@ enum { ALC256_FIXUP_ASUS_AIO_GPIO2, ALC233_FIXUP_ASUS_MIC_NO_PRESENCE, ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, + ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5567,7 +5568,7 @@ enum { .type = HDA_FIXUP_FUNC, .v.func = alc298_fixup_speaker_volume, .chained = true, - .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, + .chain_id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, @@ -5638,6 +5639,15 @@ enum { .chained = true, .chain_id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE }, + [ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH 0/7] DisplayPort audio support on Cherrytrail
by Takashi Iwai 31 Mar '17

31 Mar '17

Hi, the following patches enable DisplayPort Audio on Cherrytrail machines when applied on top of my topic/intel-lpe-audio branch. Tests of DP audio were run on Dell Wyse 3040. The regression test were performed on Baytrail (Compute Stick) and Cherrytrail (Zotac PI330) in HDMI mode. On Cherrytrail there were no issues changing between HDMI and DP connectors dynamically. Could you i915 people review and give ACK if they are OK? The changes in drm/i915 side are fairly trivial, so I'd like to take them through sound git tree once after I receive your ACKs. Changes since RFC: - reordered and squashed patches - clean-up of register definitions and offsets (based on feedback from Jani/Ville) - unmute amp for both HDMI and DP unconditionally - mute amp on invalid ELD (unplug) - remove test for chicken bit which seems to have no effect in hardware - cosmetic edits to make checkpatch happy - change i915 notification argument to pass the plataform device instead Most of hard work in this patchset has been done by Pierre, so all credits go to him. thanks, Takashi === Pierre-Louis Bossart (4): drm/i915: add DP support in LPE audio mode drm/i915: add DisplayPort amp unmute for LPE audio mode ALSA: x86: intel_hdmi: add definitions and logic for DP audio ALSA: x86: Use config base depending on the pipe Takashi Iwai (3): drm/i915: Avoid MST pipe handling for LPE audio drm/i915: Pass pipe to LPE audio notification drm/i915: Pass platform device to LPE audio notifier drivers/gpu/drm/i915/i915_drv.h | 3 +- drivers/gpu/drm/i915/i915_reg.h | 10 ++ drivers/gpu/drm/i915/intel_audio.c | 38 +++++--- drivers/gpu/drm/i915/intel_lpe_audio.c | 28 +++++- include/drm/intel_lpe_audio.h | 7 +- sound/x86/intel_hdmi_audio.c | 173 ++++++++++++++++++++++++++++----- sound/x86/intel_hdmi_audio.h | 8 +- sound/x86/intel_hdmi_lpe_audio.c | 83 ++++++++++++---- sound/x86/intel_hdmi_lpe_audio.h | 29 ++++++ 9 files changed, 315 insertions(+), 64 deletions(-) -- 2.11.0

3 17

[alsa-devel] [PATCH v3] ASoC: Add support for Maxim Integrated MAX98927 Amplifier
by Ryan Lee 31 Mar '17

31 Mar '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v3: * Combined MAX98926 and MAX98927 binding. Kept existing property name. Changes since v2: * Removed local register read/write function to avoid duplication of ASoC core function. .../devicetree/bindings/sound/max98925.txt | 22 - .../devicetree/bindings/sound/max98926.txt | 32 - .../devicetree/bindings/sound/max9892x.txt | 41 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98927.c | 826 +++++++++++++++++++++ sound/soc/codecs/max98927.h | 267 +++++++ 7 files changed, 1141 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max98925.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100755 Documentation/devicetree/bindings/sound/max9892x.txt mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/max98927.c create mode 100755 sound/soc/codecs/max98927.h diff --git a/Documentation/devicetree/bindings/sound/max98925.txt b/Documentation/devicetree/bindings/sound/max98925.txt deleted file mode 100644 index 27be63e..0000000 --- a/Documentation/devicetree/bindings/sound/max98925.txt +++ /dev/null @@ -1,22 +0,0 @@ -max98925 audio CODEC - -This device supports I2C. - -Required properties: - - - compatible : "maxim,max98925" - - - vmon-slot-no : slot number used to send voltage information - - - imon-slot-no : slot number used to send current information - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98925@1a { - compatible = "maxim,max98925"; - vmon-slot-no = <0>; - imon-slot-no = <2>; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt deleted file mode 100644 index 0b7f4e4..0000000 --- a/Documentation/devicetree/bindings/sound/max98926.txt +++ /dev/null @@ -1,32 +0,0 @@ -max98926 audio CODEC - -This device supports I2C. - -Required properties: - - - compatible : "maxim,max98926" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - - - imon-slot-no : slot number used to send current information - - - interleave-mode : When using two MAX98926 in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98926@1a { - compatible = "maxim,max98926"; - vmon-slot-no = <0>; - imon-slot-no = <2>; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt new file mode 100755 index 0000000..f617159 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -0,0 +1,41 @@ +Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : should be one of the following + - "maxim,max98925" + - "maxim,max98926" + - "maxim,max98927" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - imon-slot-no : slot number used to send current information + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - interleave-mode : When using two MAX9892X in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + Range : 0 (off), 1 (on), Default : 0 + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98927@3a { + compatible = "maxim,max98927"; + vmon-slot-no = <0>; + imon-slot-no = <1>; + interleave-mode = <0>; + reg = <0x3a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index e49e9da..cbdc8aa --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -89,6 +89,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C + select SND_SOC_MAX98927 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C @@ -588,6 +589,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX98926 tristate +config SND_SOC_MAX98927 + tristate "Maxim Integrated MAX98927 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 1796cb9..187a639 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o +snd-soc-max98927-objs := max98927.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o @@ -313,6 +314,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o +obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c new file mode 100755 index 0000000..15f2192 --- /dev/null +++ b/sound/soc/codecs/max98927.c @@ -0,0 +1,826 @@ +/* + * max98927.c -- MAX98927 ALSA Soc Audio driver + * + * Copyright (C) 2016 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/acpi.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98927.h" + +static struct reg_default max98927_reg[] = { + {MAX98927_R0001_INT_RAW1, 0x00}, + {MAX98927_R0002_INT_RAW2, 0x00}, + {MAX98927_R0003_INT_RAW3, 0x00}, + {MAX98927_R0004_INT_STATE1, 0x00}, + {MAX98927_R0005_INT_STATE2, 0x00}, + {MAX98927_R0006_INT_STATE3, 0x00}, + {MAX98927_R0007_INT_FLAG1, 0x00}, + {MAX98927_R0008_INT_FLAG2, 0x00}, + {MAX98927_R0009_INT_FLAG3, 0x00}, + {MAX98927_R000A_INT_EN1, 0x00}, + {MAX98927_R000B_INT_EN2, 0x00}, + {MAX98927_R000C_INT_EN3, 0x00}, + {MAX98927_R000D_INT_FLAG_CLR1, 0x00}, + {MAX98927_R000E_INT_FLAG_CLR2, 0x00}, + {MAX98927_R000F_INT_FLAG_CLR3, 0x00}, + {MAX98927_R0010_IRQ_CTRL, 0x00}, + {MAX98927_R0011_CLK_MON, 0x00}, + {MAX98927_R0012_WDOG_CTRL, 0x00}, + {MAX98927_R0013_WDOG_RST, 0x00}, + {MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH, 0x00}, + {MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH, 0x00}, + {MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS, 0x00}, + {MAX98927_R0017_PIN_CFG, 0x55}, + {MAX98927_R0018_PCM_RX_EN_A, 0x00}, + {MAX98927_R0019_PCM_RX_EN_B, 0x00}, + {MAX98927_R001A_PCM_TX_EN_A, 0x00}, + {MAX98927_R001B_PCM_TX_EN_B, 0x00}, + {MAX98927_R001C_PCM_TX_HIZ_CTRL_A, 0x00}, + {MAX98927_R001D_PCM_TX_HIZ_CTRL_B, 0x00}, + {MAX98927_R001E_PCM_TX_CH_SRC_A, 0x00}, + {MAX98927_R001F_PCM_TX_CH_SRC_B, 0x00}, + {MAX98927_R0020_PCM_MODE_CFG, 0x40}, + {MAX98927_R0021_PCM_MASTER_MODE, 0x00}, + {MAX98927_R0022_PCM_CLK_SETUP, 0x22}, + {MAX98927_R0023_PCM_SR_SETUP1, 0x00}, + {MAX98927_R0024_PCM_SR_SETUP2, 0x00}, + {MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, 0x00}, + {MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, 0x00}, + {MAX98927_R0027_ICC_RX_EN_A, 0x00}, + {MAX98927_R0028_ICC_RX_EN_B, 0x00}, + {MAX98927_R002B_ICC_TX_EN_A, 0x00}, + {MAX98927_R002C_ICC_TX_EN_B, 0x00}, + {MAX98927_R002E_ICC_HIZ_MANUAL_MODE, 0x00}, + {MAX98927_R002F_ICC_TX_HIZ_EN_A, 0x00}, + {MAX98927_R0030_ICC_TX_HIZ_EN_B, 0x00}, + {MAX98927_R0031_ICC_LNK_EN, 0x00}, + {MAX98927_R0032_PDM_TX_EN, 0x00}, + {MAX98927_R0033_PDM_TX_HIZ_CTRL, 0x00}, + {MAX98927_R0034_PDM_TX_CTRL, 0x00}, + {MAX98927_R0035_PDM_RX_CTRL, 0x00}, + {MAX98927_R0036_AMP_VOL_CTRL, 0x00}, + {MAX98927_R0037_AMP_DSP_CFG, 0x02}, + {MAX98927_R0038_TONE_GEN_DC_CFG, 0x00}, + {MAX98927_R0039_DRE_CTRL, 0x01}, + {MAX98927_R003A_AMP_EN, 0x00}, + {MAX98927_R003B_SPK_SRC_SEL, 0x00}, + {MAX98927_R003C_SPK_GAIN, 0x00}, + {MAX98927_R003D_SSM_CFG, 0x01}, + {MAX98927_R003E_MEAS_EN, 0x00}, + {MAX98927_R003F_MEAS_DSP_CFG, 0x04}, + {MAX98927_R0040_BOOST_CTRL0, 0x00}, + {MAX98927_R0041_BOOST_CTRL3, 0x00}, + {MAX98927_R0042_BOOST_CTRL1, 0x00}, + {MAX98927_R0043_MEAS_ADC_CFG, 0x00}, + {MAX98927_R0044_MEAS_ADC_BASE_MSB, 0x00}, + {MAX98927_R0045_MEAS_ADC_BASE_LSB, 0x00}, + {MAX98927_R0046_ADC_CH0_DIVIDE, 0x00}, + {MAX98927_R0047_ADC_CH1_DIVIDE, 0x00}, + {MAX98927_R0048_ADC_CH2_DIVIDE, 0x00}, + {MAX98927_R0049_ADC_CH0_FILT_CFG, 0x00}, + {MAX98927_R004A_ADC_CH1_FILT_CFG, 0x00}, + {MAX98927_R004B_ADC_CH2_FILT_CFG, 0x00}, + {MAX98927_R004C_MEAS_ADC_CH0_READ, 0x00}, + {MAX98927_R004D_MEAS_ADC_CH1_READ, 0x00}, + {MAX98927_R004E_MEAS_ADC_CH2_READ, 0x00}, + {MAX98927_R0051_BROWNOUT_STATUS, 0x00}, + {MAX98927_R0052_BROWNOUT_EN, 0x00}, + {MAX98927_R0053_BROWNOUT_INFINITE_HOLD, 0x00}, + {MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR, 0x00}, + {MAX98927_R0055_BROWNOUT_LVL_HOLD, 0x00}, + {MAX98927_R005A_BROWNOUT_LVL1_THRESH, 0x00}, + {MAX98927_R005B_BROWNOUT_LVL2_THRESH, 0x00}, + {MAX98927_R005C_BROWNOUT_LVL3_THRESH, 0x00}, + {MAX98927_R005D_BROWNOUT_LVL4_THRESH, 0x00}, + {MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS, 0x00}, + {MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL, 0x00}, + {MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL, 0x00}, + {MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE, 0x00}, + {MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT, 0x00}, + {MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1, 0x00}, + {MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2, 0x00}, + {MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3, 0x00}, + {MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT, 0x00}, + {MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1, 0x00}, + {MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2, 0x00}, + {MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3, 0x00}, + {MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT, 0x00}, + {MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1, 0x00}, + {MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2, 0x00}, + {MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3, 0x00}, + {MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT, 0x00}, + {MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, 0x00}, + {MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2, 0x00}, + {MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3, 0x00}, + {MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0x00}, + {MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY, 0x00}, + {MAX98927_R0084_ENV_TRACK_REL_RATE, 0x00}, + {MAX98927_R0085_ENV_TRACK_HOLD_RATE, 0x00}, + {MAX98927_R0086_ENV_TRACK_CTRL, 0x00}, + {MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, 0x00}, + {MAX98927_R00FF_GLOBAL_SHDN, 0x00}, + {MAX98927_R0100_SOFT_RESET, 0x00}, + {MAX98927_R01FF_REV_ID, 0x40}, +}; + +static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int mode = 0; + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + mode = MAX98927_PCM_MASTER_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98927->master = true; + mode = MAX98927_PCM_MASTER_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + mode = MAX98927_PCM_MASTER_MODE_HYBRID; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MASK, + mode); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + max98927->iface |= SND_SOC_DAIFMT_I2S; + + break; + case SND_SOC_DAIFMT_LEFT_J: + max98927->iface |= SND_SOC_DAIFMT_LEFT_J; + break; + default: + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + max98927->iface, max98927->iface); + + /* pcm channel configuration */ + if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN| + MAX98927_PCM_RX_CH1_EN); + + return 0; +} + +/* codec MCLK rate in master mode */ +static const int rate_table[] = { + 5644800, 6000000, 6144000, 6500000, + 9600000, 11289600, 12000000, 12288000, + 13000000, 19200000, +}; + +static int max98927_set_clock(struct max98927_priv *max98927, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_codec *codec = max98927->codec; + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98927->ch_size; + int reg = MAX98927_R0022_PCM_CLK_SETUP; + int mask = MAX98927_PCM_CLK_SETUP_BSEL_MASK; + int value; + + if (max98927->master) { + int i; + /* match rate to closest value */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i] >= max98927->sysclk) + break; + } + if (i == ARRAY_SIZE(rate_table)) { + dev_err(codec->dev, "failed to find proper clock rate.\n"); + return -EINVAL; + } + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MCLK_MASK, + i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); + } + + switch (blr_clk_ratio) { + case 32: + value = 2; + break; + case 48: + value = 3; + break; + case 64: + value = 4; + break; + default: + return -EINVAL; + } + regmap_update_bits(max98927->regmap, reg, mask, value); + return 0; +} + +static int max98927_dai_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 max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + max98927->ch_size = 16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + max98927->ch_size = 24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + max98927->ch_size = 32; + break; + default: + dev_err(codec->dev, "format unsupported %d", + params_format(params)); + goto err; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(codec->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_8000; + break; + case 11025: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_11025; + break; + case 12000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_12000; + break; + case 16000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_16000; + break; + case 22050: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_22050; + break; + case 24000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_24000; + break; + case 32000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_32000; + break; + case 44100: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_44100; + break; + case 48000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_48000; + break; + default: + dev_err(codec->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98927->regmap, + MAX98927_R0023_PCM_SR_SETUP1, + MAX98927_PCM_SR_SET1_SR_MASK, + sampling_rate); + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_SR_MASK, + sampling_rate << MAX98927_PCM_SR_SET2_SR_SHIFT); + + /* set sampling rate of IV */ + if (max98927->interleave_mode && + sampling_rate > MAX98927_PCM_SR_SET1_SR_16000) + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate - 3); + else + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate); + return max98927_set_clock(max98927, params); +err: + return -EINVAL; +} + +#define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->sysclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops max98927_dai_ops = { + .set_sysclk = max98927_dai_set_sysclk, + .set_fmt = max98927_dai_set_fmt, + .hw_params = max98927_dai_hw_params, +}; + +static int max98927_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 1); + /* enable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN); + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 1); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 0); + /* disable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, 0); + break; + default: + return 0; + } + return 0; +} + +static const char * const max98927_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT, + 3, max98927_switch_text); + +static const struct snd_kcontrol_new max98927_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_soc_dapm_widget max98927_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98927_R003A_AMP_EN, + 0, 0, max98927_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, + &max98927_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static DECLARE_TLV_DB_SCALE(max98927_spk_tlv, 300, 300, 0); +static DECLARE_TLV_DB_SCALE(max98927_digital_tlv, -1600, 25, 0); + +static bool max98927_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0028_ICC_RX_EN_B: + case MAX98927_R002B_ICC_TX_EN_A ... MAX98927_R002C_ICC_TX_EN_B: + case MAX98927_R002E_ICC_HIZ_MANUAL_MODE + ... MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS + ... MAX98927_R0055_BROWNOUT_LVL_HOLD: + case MAX98927_R005A_BROWNOUT_LVL1_THRESH + ... MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE: + case MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT + ... MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R00FF_GLOBAL_SHDN: + case MAX98927_R0100_SOFT_RESET: + case MAX98927_R01FF_REV_ID: + return true; + } + return false; +}; + +static bool max98927_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + return true; + default: + return false; + } +} + +static const char * const max98927_boost_voltage_text[] = { + "6.5V", "6.625V", "6.75V", "6.875V", "7V", "7.125V", "7.25V", "7.375V", + "7.5V", "7.625V", "7.75V", "7.875V", "8V", "8.125V", "8.25V", "8.375V", + "8.5V", "8.625V", "8.75V", "8.875V", "9V", "9.125V", "9.25V", "9.375V", + "9.5V", "9.625V", "9.75V", "9.875V", "10V" +}; + +static const char * const max98927_speaker_source_text[] = { + "i2s", "reserved", "tone", "pdm" +}; + +static const char * const max98927_monomix_output_text[] = { + "ch_0", "ch_1", "ch_1_2_div" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_boost_voltage, + MAX98927_R0040_BOOST_CTRL0, 0, + max98927_boost_voltage_text); + +static SOC_ENUM_SINGLE_DECL(max98927_spk_source, + MAX98927_R003B_SPK_SRC_SEL, 0, + max98927_speaker_source_text); + +static SOC_ENUM_SINGLE_DECL(max98927_monomix_output, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT, + max98927_monomix_output_text); + +static const struct snd_kcontrol_new max98927_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, + 0, 6, 0, + max98927_spk_tlv), + SOC_SINGLE_TLV("Digital Gain", MAX98927_R0036_AMP_VOL_CTRL, + 0, (1<<MAX98927_AMP_VOL_WIDTH)-1, 0, + max98927_digital_tlv), + SOC_SINGLE("Amp DSP Enable", MAX98927_R0052_BROWNOUT_EN, + MAX98927_BROWNOUT_DSP_SHIFT, 1, 0), + SOC_SINGLE("Ramp Switch", MAX98927_R0037_AMP_DSP_CFG, + MAX98927_AMP_DSP_CFG_RMP_SHIFT, 1, 0), + SOC_SINGLE("DRE EN", MAX98927_R0039_DRE_CTRL, + MAX98927_DRE_EN_SHIFT, 1, 0), + SOC_SINGLE("Amp Volume Location", MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), + SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), + SOC_ENUM("Speaker Source", max98927_spk_source), + SOC_ENUM("Monomix Output", max98927_monomix_output), +}; + +static const struct snd_soc_dapm_route max98927_audio_map[] = { + {"Amp Enable", NULL, "DAI_OUT"}, + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98927_dai[] = { + { + .name = "max98927-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .ops = &max98927_dai_ops, + } +}; + +static int max98927_probe(struct snd_soc_codec *codec) +{ + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int ret = 0, reg = 0; + + max98927->codec = codec; + codec->control_data = max98927->regmap; + codec->cache_bypass = 1; + + /* Software Reset */ + regmap_write(max98927->regmap, + MAX98927_R0100_SOFT_RESET, MAX98927_SOFT_RESET); + + /* Check Revision ID */ + ret = regmap_read(max98927->regmap, + MAX98927_R01FF_REV_ID, &reg); + if (ret < 0) + dev_err(codec->dev, + "Failed to read: 0x%02X\n", MAX98927_R01FF_REV_ID); + else + dev_info(codec->dev, + "MAX98927 revisionID: 0x%02X\n", reg); + + /* IV default slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + 0x80); + regmap_write(max98927->regmap, + MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, + 0x1); + /* Set inital volume (+13dB) */ + regmap_write(max98927->regmap, + MAX98927_R0036_AMP_VOL_CTRL, + 0x38); + regmap_write(max98927->regmap, + MAX98927_R003C_SPK_GAIN, + 0x05); + /* Enable DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R0037_AMP_DSP_CFG, + 0x03); + /* Enable IMON VMON DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R003F_MEAS_DSP_CFG, + 0xF7); + /* Boost Output Voltage & Current limit */ + regmap_write(max98927->regmap, + MAX98927_R0040_BOOST_CTRL0, + 0x1C); + regmap_write(max98927->regmap, + MAX98927_R0042_BOOST_CTRL1, + 0x3E); + /* Measurement ADC config */ + regmap_write(max98927->regmap, + MAX98927_R0043_MEAS_ADC_CFG, + 0x04); + regmap_write(max98927->regmap, + MAX98927_R0044_MEAS_ADC_BASE_MSB, + 0x00); + regmap_write(max98927->regmap, + MAX98927_R0045_MEAS_ADC_BASE_LSB, + 0x24); + /* Brownout Level */ + regmap_write(max98927->regmap, + MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, + 0x06); + /* Envelope Tracking configuration */ + regmap_write(max98927->regmap, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, + 0x08); + regmap_write(max98927->regmap, + MAX98927_R0086_ENV_TRACK_CTRL, + 0x01); + regmap_write(max98927->regmap, + MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, + 0x10); + + /* voltage, current slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001E_PCM_TX_CH_SRC_A, + (max98927->i_l_slot<<MAX98927_PCM_TX_CH_SRC_A_I_SHIFT| + max98927->v_l_slot)&0xFF); + + if (max98927->v_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->v_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->v_l_slot, + 1 << max98927->v_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->v_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->v_l_slot - 8), + 1 << (max98927->v_l_slot - 8)); + } + + if (max98927->i_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->i_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->i_l_slot, + 1 << max98927->i_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->i_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->i_l_slot - 8), + 1 << (max98927->i_l_slot - 8)); + } + + /* Set interleave mode */ + if (max98927->interleave_mode) + regmap_update_bits(max98927->regmap, + MAX98927_R001F_PCM_TX_CH_SRC_B, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK); + return ret; +} + +static const struct snd_soc_codec_driver soc_codec_dev_max98927 = { + .probe = max98927_probe, + .component_driver = { + .controls = max98927_snd_controls, + .num_controls = ARRAY_SIZE(max98927_snd_controls), + .dapm_widgets = max98927_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98927_dapm_widgets), + .dapm_routes = max98927_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98927_audio_map), + }, +}; + +static const struct regmap_config max98927_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98927_R01FF_REV_ID, + .reg_defaults = max98927_reg, + .num_reg_defaults = ARRAY_SIZE(max98927_reg), + .readable_reg = max98927_readable_register, + .volatile_reg = max98927_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static void max98927_slot_config(struct i2c_client *i2c, + struct max98927_priv *max98927) +{ + int value; + + if (!of_property_read_u32(i2c->dev.of_node, + "vmon-slot-no", &value)) + max98927->v_l_slot = value & 0xF; + else + max98927->v_l_slot = 0; + if (!of_property_read_u32(i2c->dev.of_node, + "imon-slot-no", &value)) + max98927->i_l_slot = value & 0xF; + else + max98927->i_l_slot = 1; +} + +static int max98927_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + + int ret = 0, value; + struct max98927_priv *max98927 = NULL; + + max98927 = devm_kzalloc(&i2c->dev, + sizeof(*max98927), GFP_KERNEL); + + if (!max98927) { + ret = -ENOMEM; + goto err; + } + i2c_set_clientdata(i2c, max98927); + + /* update interleave mode info */ + if (!of_property_read_u32(i2c->dev.of_node, + "interleave_mode", &value)) { + if (value > 0) + max98927->interleave_mode = 1; + else + max98927->interleave_mode = 0; + } else + max98927->interleave_mode = 0; + + /* regmap initialization */ + max98927->regmap + = devm_regmap_init_i2c(i2c, &max98927_regmap); + if (IS_ERR(max98927->regmap)) { + ret = PTR_ERR(max98927->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err; + } + + /* voltage/current slot configuration */ + max98927_slot_config(i2c, max98927); + + /* codec registeration */ + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98927, + max98927_dai, ARRAY_SIZE(max98927_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; + +err: + if (max98927) + devm_kfree(&i2c->dev, max98927); + return ret; +} + +static int max98927_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98927_i2c_id[] = { + { "max98927", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98927_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98927_of_match[] = { + { .compatible = "maxim,max98927", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98927_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98927_acpi_match[] = { + { "MX98927", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98927_acpi_match); +#endif + +static struct i2c_driver max98927_i2c_driver = { + .driver = { + .name = "max98927", + .of_match_table = of_match_ptr(max98927_of_match), + .acpi_match_table = ACPI_PTR(max98927_acpi_match), + .pm = NULL, + }, + .probe = max98927_i2c_probe, + .remove = max98927_i2c_remove, + .id_table = max98927_i2c_id, +}; + +module_i2c_driver(max98927_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98927 driver"); +MODULE_AUTHOR("Ryan Lee <ryans.lee(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h new file mode 100755 index 0000000..e13e18a --- /dev/null +++ b/sound/soc/codecs/max98927.h @@ -0,0 +1,267 @@ +/* + * max98927.h -- MAX98927 ALSA Soc Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ +#ifndef _MAX98927_H +#define _MAX98927_H + +/* Register Values */ +#define MAX98927_R0001_INT_RAW1 0x0001 +#define MAX98927_R0002_INT_RAW2 0x0002 +#define MAX98927_R0003_INT_RAW3 0x0003 +#define MAX98927_R0004_INT_STATE1 0x0004 +#define MAX98927_R0005_INT_STATE2 0x0005 +#define MAX98927_R0006_INT_STATE3 0x0006 +#define MAX98927_R0007_INT_FLAG1 0x0007 +#define MAX98927_R0008_INT_FLAG2 0x0008 +#define MAX98927_R0009_INT_FLAG3 0x0009 +#define MAX98927_R000A_INT_EN1 0x000A +#define MAX98927_R000B_INT_EN2 0x000B +#define MAX98927_R000C_INT_EN3 0x000C +#define MAX98927_R000D_INT_FLAG_CLR1 0x000D +#define MAX98927_R000E_INT_FLAG_CLR2 0x000E +#define MAX98927_R000F_INT_FLAG_CLR3 0x000F +#define MAX98927_R0010_IRQ_CTRL 0x0010 +#define MAX98927_R0011_CLK_MON 0x0011 +#define MAX98927_R0012_WDOG_CTRL 0x0012 +#define MAX98927_R0013_WDOG_RST 0x0013 +#define MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH 0x0014 +#define MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH 0x0015 +#define MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS 0x0016 +#define MAX98927_R0017_PIN_CFG 0x0017 +#define MAX98927_R0018_PCM_RX_EN_A 0x0018 +#define MAX98927_R0019_PCM_RX_EN_B 0x0019 +#define MAX98927_R001A_PCM_TX_EN_A 0x001A +#define MAX98927_R001B_PCM_TX_EN_B 0x001B +#define MAX98927_R001C_PCM_TX_HIZ_CTRL_A 0x001C +#define MAX98927_R001D_PCM_TX_HIZ_CTRL_B 0x001D +#define MAX98927_R001E_PCM_TX_CH_SRC_A 0x001E +#define MAX98927_R001F_PCM_TX_CH_SRC_B 0x001F +#define MAX98927_R0020_PCM_MODE_CFG 0x0020 +#define MAX98927_R0021_PCM_MASTER_MODE 0x0021 +#define MAX98927_R0022_PCM_CLK_SETUP 0x0022 +#define MAX98927_R0023_PCM_SR_SETUP1 0x0023 +#define MAX98927_R0024_PCM_SR_SETUP2 0x0024 +#define MAX98927_R0025_PCM_TO_SPK_MONOMIX_A 0x0025 +#define MAX98927_R0026_PCM_TO_SPK_MONOMIX_B 0x0026 +#define MAX98927_R0027_ICC_RX_EN_A 0x0027 +#define MAX98927_R0028_ICC_RX_EN_B 0x0028 +#define MAX98927_R002B_ICC_TX_EN_A 0x002B +#define MAX98927_R002C_ICC_TX_EN_B 0x002C +#define MAX98927_R002E_ICC_HIZ_MANUAL_MODE 0x002E +#define MAX98927_R002F_ICC_TX_HIZ_EN_A 0x002F +#define MAX98927_R0030_ICC_TX_HIZ_EN_B 0x0030 +#define MAX98927_R0031_ICC_LNK_EN 0x0031 +#define MAX98927_R0032_PDM_TX_EN 0x0032 +#define MAX98927_R0033_PDM_TX_HIZ_CTRL 0x0033 +#define MAX98927_R0034_PDM_TX_CTRL 0x0034 +#define MAX98927_R0035_PDM_RX_CTRL 0x0035 +#define MAX98927_R0036_AMP_VOL_CTRL 0x0036 +#define MAX98927_R0037_AMP_DSP_CFG 0x0037 +#define MAX98927_R0038_TONE_GEN_DC_CFG 0x0038 +#define MAX98927_R0039_DRE_CTRL 0x0039 +#define MAX98927_R003A_AMP_EN 0x003A +#define MAX98927_R003B_SPK_SRC_SEL 0x003B +#define MAX98927_R003C_SPK_GAIN 0x003C +#define MAX98927_R003D_SSM_CFG 0x003D +#define MAX98927_R003E_MEAS_EN 0x003E +#define MAX98927_R003F_MEAS_DSP_CFG 0x003F +#define MAX98927_R0040_BOOST_CTRL0 0x0040 +#define MAX98927_R0041_BOOST_CTRL3 0x0041 +#define MAX98927_R0042_BOOST_CTRL1 0x0042 +#define MAX98927_R0043_MEAS_ADC_CFG 0x0043 +#define MAX98927_R0044_MEAS_ADC_BASE_MSB 0x0044 +#define MAX98927_R0045_MEAS_ADC_BASE_LSB 0x0045 +#define MAX98927_R0046_ADC_CH0_DIVIDE 0x0046 +#define MAX98927_R0047_ADC_CH1_DIVIDE 0x0047 +#define MAX98927_R0048_ADC_CH2_DIVIDE 0x0048 +#define MAX98927_R0049_ADC_CH0_FILT_CFG 0x0049 +#define MAX98927_R004A_ADC_CH1_FILT_CFG 0x004A +#define MAX98927_R004B_ADC_CH2_FILT_CFG 0x004B +#define MAX98927_R004C_MEAS_ADC_CH0_READ 0x004C +#define MAX98927_R004D_MEAS_ADC_CH1_READ 0x004D +#define MAX98927_R004E_MEAS_ADC_CH2_READ 0x004E +#define MAX98927_R0051_BROWNOUT_STATUS 0x0051 +#define MAX98927_R0052_BROWNOUT_EN 0x0052 +#define MAX98927_R0053_BROWNOUT_INFINITE_HOLD 0x0053 +#define MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR 0x0054 +#define MAX98927_R0055_BROWNOUT_LVL_HOLD 0x0055 +#define MAX98927_R005A_BROWNOUT_LVL1_THRESH 0x005A +#define MAX98927_R005B_BROWNOUT_LVL2_THRESH 0x005B +#define MAX98927_R005C_BROWNOUT_LVL3_THRESH 0x005C +#define MAX98927_R005D_BROWNOUT_LVL4_THRESH 0x005D +#define MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS 0x005E +#define MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL 0x005F +#define MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL 0x0060 +#define MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE 0x0061 +#define MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT 0x0072 +#define MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1 0x0073 +#define MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2 0x0074 +#define MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3 0x0075 +#define MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT 0x0076 +#define MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1 0x0077 +#define MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2 0x0078 +#define MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3 0x0079 +#define MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT 0x007A +#define MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1 0x007B +#define MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2 0x007C +#define MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3 0x007D +#define MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT 0x007E +#define MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1 0x007F +#define MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2 0x0080 +#define MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3 0x0081 +#define MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM 0x0082 +#define MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY 0x0083 +#define MAX98927_R0084_ENV_TRACK_REL_RATE 0x0084 +#define MAX98927_R0085_ENV_TRACK_HOLD_RATE 0x0085 +#define MAX98927_R0086_ENV_TRACK_CTRL 0x0086 +#define MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ 0x0087 +#define MAX98927_R00FF_GLOBAL_SHDN 0x00FF +#define MAX98927_R0100_SOFT_RESET 0x0100 +#define MAX98927_R01FF_REV_ID 0x01FF + +/* MAX98927_R0018_PCM_RX_EN_A */ +#define MAX98927_PCM_RX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_RX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_RX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_RX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_RX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_RX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_RX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_RX_CH7_EN (0x1 << 7) + +/* MAX98927_R001A_PCM_TX_EN_A */ +#define MAX98927_PCM_TX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_TX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_TX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_TX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_TX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_TX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_TX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_TX_CH7_EN (0x1 << 7) + +/* MAX98927_R001E_PCM_TX_CH_SRC_A */ +#define MAX98927_PCM_TX_CH_SRC_A_V_SHIFT (0) +#define MAX98927_PCM_TX_CH_SRC_A_I_SHIFT (4) + +/* MAX98927_R001F_PCM_TX_CH_SRC_B */ +#define MAX98927_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 5) + +/* MAX98927_R0020_PCM_MODE_CFG */ +#define MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 2) +#define MAX98927_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) + +#define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98927_R0021_PCM_MASTER_MODE */ +#define MAX98927_PCM_MASTER_MODE_MASK (0x3 << 0) +#define MAX98927_PCM_MASTER_MODE_SLAVE (0x0 << 0) +#define MAX98927_PCM_MASTER_MODE_MASTER (0x3 << 0) +#define MAX98927_PCM_MASTER_MODE_HYBRID (0x1 << 0) + +#define MAX98927_PCM_MASTER_MODE_MCLK_MASK (0xF << 2) +#define MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT (2) + +/* MAX98927_R0022_PCM_CLK_SETUP */ +#define MAX98927_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98927_R0023_PCM_SR_SETUP1 */ +#define MAX98927_PCM_SR_SET1_SR_MASK (0xF << 0) + +#define MAX98927_PCM_SR_SET1_SR_8000 (0x0 << 0) +#define MAX98927_PCM_SR_SET1_SR_11025 (0x1 << 0) +#define MAX98927_PCM_SR_SET1_SR_12000 (0x2 << 0) +#define MAX98927_PCM_SR_SET1_SR_16000 (0x3 << 0) +#define MAX98927_PCM_SR_SET1_SR_22050 (0x4 << 0) +#define MAX98927_PCM_SR_SET1_SR_24000 (0x5 << 0) +#define MAX98927_PCM_SR_SET1_SR_32000 (0x6 << 0) +#define MAX98927_PCM_SR_SET1_SR_44100 (0x7 << 0) +#define MAX98927_PCM_SR_SET1_SR_48000 (0x8 << 0) + +/* MAX98927_R0024_PCM_SR_SETUP2 */ +#define MAX98927_PCM_SR_SET2_SR_MASK (0xF << 4) +#define MAX98927_PCM_SR_SET2_SR_SHIFT (4) +#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xf << 0) + +/* MAX98927_R0025_PCM_TO_SPK_MONOMIX_A */ +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT (6) + +/* MAX98927_R0036_AMP_VOL_CTRL */ +#define MAX98927_AMP_VOL_SEL (0x1 << 7) +#define MAX98927_AMP_VOL_SEL_WIDTH (1) +#define MAX98927_AMP_VOL_SEL_SHIFT (7) +#define MAX98927_AMP_VOL_MASK (0x7f << 0) +#define MAX98927_AMP_VOL_WIDTH (7) +#define MAX98927_AMP_VOL_SHIFT (0) + +/* MAX98927_R0037_AMP_DSP_CFG */ +#define MAX98927_AMP_DSP_CFG_DCBLK_EN (0x1 << 0) +#define MAX98927_AMP_DSP_CFG_DITH_EN (0x1 << 1) +#define MAX98927_AMP_DSP_CFG_RMP_BYPASS (0x1 << 4) +#define MAX98927_AMP_DSP_CFG_DAC_INV (0x1 << 5) +#define MAX98927_AMP_DSP_CFG_RMP_SHIFT (4) + +/* MAX98927_R0039_DRE_CTRL */ +#define MAX98927_DRE_CTRL_DRE_EN (0x1 << 0) +#define MAX98927_DRE_EN_SHIFT 0x1 + +/* MAX98927_R003A_AMP_EN */ +#define MAX98927_AMP_EN_MASK (0x1 << 0) + +/* MAX98927_R003B_SPK_SRC_SEL */ +#define MAX98927_SPK_SRC_MASK (0x3 << 0) + +/* MAX98927_R003C_SPK_GAIN */ +#define MAX98927_SPK_PCM_GAIN_MASK (0x7 << 0) +#define MAX98927_SPK_PDM_GAIN_MASK (0x7 << 4) +#define MAX98927_SPK_GAIN_WIDTH (3) + +/* MAX98927_R003E_MEAS_EN */ +#define MAX98927_MEAS_V_EN (0x1 << 0) +#define MAX98927_MEAS_I_EN (0x1 << 1) + +/* MAX98927_R0040_BOOST_CTRL0 */ +#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1f << 0) +#define MAX98927_BOOST_CTRL0_PVDD_MASK (0x1 << 7) +#define MAX98927_BOOST_CTRL0_PVDD_EN_SHIFT (7) + +/* MAX98927_R0052_BROWNOUT_EN */ +#define MAX98927_BROWNOUT_BDE_EN (0x1 << 0) +#define MAX98927_BROWNOUT_AMP_EN (0x1 << 1) +#define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) +#define MAX98927_BROWNOUT_DSP_SHIFT (2) + +/* MAX98927_R0100_SOFT_RESET */ +#define MAX98927_SOFT_RESET (0x1 << 0) + +/* MAX98927_R00FF_GLOBAL_SHDN */ +#define MAX98927_GLOBAL_EN_MASK (0x1 << 0) + +struct max98927_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98927_pdata *pdata; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int v_l_slot; + unsigned int i_l_slot; + bool interleave_mode; + unsigned int ch_size; + unsigned int rate; + unsigned int iface; + unsigned int master; + unsigned int digital_gain; +}; +#endif -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: bxtn: fix spelling mistake: "Timout" -> "Timeout"
by Colin King 31 Mar '17

31 Mar '17

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err error message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/bxt-sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 600d95891996..268bdaec8042 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -171,7 +171,7 @@ static int sst_bxt_prepare_fw(struct sst_dsp *ctx, SKL_ADSP_REG_HIPCIE_DONE, BXT_INIT_TIMEOUT, "HIPCIE Done"); if (ret < 0) { - dev_err(ctx->dev, "Timout for Purge Request%d\n", ret); + dev_err(ctx->dev, "Timeout for Purge Request%d\n", ret); goto base_fw_load_failed; } -- 2.11.0

3 2

[alsa-devel] [PATCH 0/2] ASoC: rcar: fixup of_clk_xxx_provider()
by Kuninori Morimoto 31 Mar '17

31 Mar '17

Hi Mark These are bugfix for Renesas R-Car sound clock. The issue will happen if I do bind/unbind many times. Kuninori Morimoto (2): ASoC: rcar: fixup of_clk_add_provider() usage for multi clkout ASoC: rcar: call missing of_clk_del_provider() when remove sound/soc/sh/rcar/adg.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: wm8903: add regulator handling
by Linus Walleij 30 Mar '17

30 Mar '17

The WM8903 has four different voltage inputs: AVDD, CPVDD, DBVDD and DCVDD. On the Qualcomm APQ8060 Dragonboard these are all supplied from proper regulators and thus need activating and binding. This is a quick-and-dirty solution just grabbing and enabling the regulator supplies on probe() and disabling them on remove() and the errorpath. More elaborate power management is likely possible. I assume the nVidia designs using this codec have some hard-wired always-on power and will be happy with using the dummy regulators for this. But someone from the nVidia camp should probably check whether they can bind these to proper regulators instead. We also amend the DT binding document. A small change like this does not warrant a separate patch for augmenting these. Cc: devicetree(a)vger.kernel.org Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Mark: I was thinking about adding runtime PM for disabling these regulators when unused, but I'm uncertain about the interaction with DAPM in that regard. This atleast gives us control over the supplies. --- Documentation/devicetree/bindings/sound/wm8903.txt | 13 +++++++++ sound/soc/codecs/wm8903.c | 31 ++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index 94ec32c194bb..afc51caf1137 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,14 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. + - AVDD-supply : Analog power supply regulator on the AVDD pin. + + - CPVDD-supply : Charge pump supply regulator on the CPVDD pin. + + - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin. + + - DCVDD-supply : Digital core supply regulator for the DCVDD pin. + Pins on the device (for linking into audio routes): * IN1L @@ -54,6 +62,11 @@ codec: wm8903@1a { reg = <0x1a>; interrupts = < 347 >; + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDC-supply = <&fooreg_d>; + gpio-controller; #gpio-cells = <2>; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 6e887c2c42b1..237eeb9a8b97 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -24,6 +24,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/irq.h> #include <linux/mutex.h> @@ -115,10 +116,19 @@ static const struct reg_default wm8903_reg_defaults[] = { { 172, 0x0000 }, /* R172 - Analogue Output Bias 0 */ }; +#define WM8903_NUM_SUPPLIES 4 +static const char *wm8903_supply_names[WM8903_NUM_SUPPLIES] = { + "AVDD", + "CPVDD", + "DBVDD", + "DCVDD", +}; + struct wm8903_priv { struct wm8903_platform_data *pdata; struct device *dev; struct regmap *regmap; + struct regulator_bulk_data supplies[WM8903_NUM_SUPPLIES]; int sysclk; int irq; @@ -2030,6 +2040,23 @@ static int wm8903_i2c_probe(struct i2c_client *i2c, pdata = wm8903->pdata; + for (i = 0; i < ARRAY_SIZE(wm8903->supplies); i++) + wm8903->supplies[i].supply = wm8903_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8903->supplies), + wm8903->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8903->supplies), + wm8903->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); @@ -2160,6 +2187,8 @@ static int wm8903_i2c_probe(struct i2c_client *i2c, return 0; err: + regulator_bulk_disable(ARRAY_SIZE(wm8903->supplies), + wm8903->supplies); return ret; } @@ -2167,6 +2196,8 @@ static int wm8903_i2c_remove(struct i2c_client *client) { struct wm8903_priv *wm8903 = i2c_get_clientdata(client); + regulator_bulk_disable(ARRAY_SIZE(wm8903->supplies), + wm8903->supplies); if (client->irq) free_irq(client->irq, wm8903); wm8903_free_gpio(wm8903); -- 2.9.3

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[alsa-devel] pulseaudio fails to start with kernel 4.11, caused by new snd_hdmi_lpe_audio module)
by Hans de Goede 30 Mar '17

30 Mar '17

Hi, Lately I've been working on fixing various issues people are having with baytrail / cherrytrail devices. One thing I noticed when moving my testing / development to 4.11 is that pulseaudio does not work (it aborts while starting) this seemed to be caused by the new snd_hdmi_lpe_audio module. If I blacklist that all is fine. Any idea what is causing this ? Regards, Hans

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Re: [alsa-devel] [Intel-gfx] [PATCH v2 2/2] drm/i915: Implement cdclk restrictions based on Azalia BCLK
by Ville Syrjälä 30 Mar '17

30 Mar '17

On Wed, Mar 29, 2017 at 09:16:36PM +0000, Pandiyan, Dhinakaran wrote: > On Wed, 2017-03-29 at 11:50 +0300, Ville Syrjälä wrote: > > On Tue, Mar 07, 2017 at 04:12:52PM -0800, Dhinakaran Pandiyan wrote: > > > According to BSpec, "The CD clock frequency must be at least twice the > > > frequency of the Azalia BCLK." and BCLK is configured to 96 MHz by > > > default. This check is needed because BXT and GLK support cdclk > > > frequencies less than 192 MHz. > > > > > > Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> > > > --- > > > drivers/gpu/drm/i915/intel_cdclk.c | 12 ++++++++++++ > > > 1 file changed, 12 insertions(+) > > > > > > diff --git a/drivers/gpu/drm/i915/intel_cdclk.c b/drivers/gpu/drm/i915/intel_cdclk.c > > > index e8c1181..7b1ac1d 100644 > > > --- a/drivers/gpu/drm/i915/intel_cdclk.c > > > +++ b/drivers/gpu/drm/i915/intel_cdclk.c > > > @@ -1458,6 +1458,18 @@ static int bdw_adjust_min_pipe_pixel_rate(struct intel_crtc_state *crtc_state, > > > pixel_rate = max(432000, pixel_rate); > > > } > > > > > > + /* According to BSpec, "The CD clock frequency must be at least twice > > > + * the frequency of the Azalia BCLK." and BCLK is 96 MHz by default. > > > + * The check for GLK has to be adjusted as the platform can output > > > + * two pixels per clock. > > > + */ > > > + if (crtc_state->has_audio) { > > > + if (IS_GEMINILAKE(dev_priv)) > > > + pixel_rate = max(2 * 2 * 96000, pixel_rate); > > > > BTW that x2 factor for GLK looks wrong. It should be /2. > > > > https://bugs.freedesktop.org/show_bug.cgi?id=100439 > > > > BSpec[1] says cdclk needs to be > 2*96 MHz (azalia bclk). To set a cdclk > > 192 MHz on GLK, we need to pass in max_pixclk = 2*192 MHz. But with a > factor of /2, we can end up picking 158.4 MHz or 79.2 MHz for cdclk, > which are lower than 192 MHz. > > > static int glk_calc_cdclk(int max_pixclk) > { > if (max_pixclk > 2 * 158400) > return 316800; > else if (max_pixclk > 2 * 79200) > return 158400; > else > return 79200; > } > > [1] Geminilake Clocks > "158.4 MHz CD (cannot be used when audio is enabled and Azalia BCLK is > 96 MHz) > 316.8 MHz CD > 79.2 MHz CD (exclusively for resolutions up to 1080p in low power single > pipe eDP/MIPI configurations, no audio support)" Argh. So it's really about the cdclk freq vs. bclk. Apparently the fact that the pipe can output two pixels per clock doesn't extend to the HDA side. OK, then the code looks correct. But how do we know the bclk is really 96Mhz? Or are we just making a worst case estimate here? Base on the hda spec I think bclk should typically be 24 MHz, but of course the hda spec is so old that it probably has little to do with today's realities. I guess ideally we'd like the audio driver to tell us what the frequency is, or we'd dig that up from somewhere ourselves. Hmm. AUD_FREQ_CNTRL seems to have something. 96 vs. 48 MHz. But I'm not sure we could trust that at boot time. Not sure if the audio driver will make its own choice if BCLK somewhere. Can any alsa folks help us? > > > > -DK > > > > + if (IS_BROXTON(dev_priv)) > > > + pixel_rate = max(2 * 96000, pixel_rate); > > > + } > > > + > > > return pixel_rate; > > > } > > > > > > -- > > > 2.7.4 > > > > > > _______________________________________________ > > > Intel-gfx mailing list > > > Intel-gfx(a)lists.freedesktop.org > > > https://lists.freedesktop.org/mailman/listinfo/intel-gfx > > > -- Ville Syrjälä Intel OTC

3 3

[alsa-devel] [GIT PULL] ASoC fixes for v4.11
by Mark Brown 30 Mar '17

30 Mar '17

The following changes since commit c1ae3cfa0e89fa1a7ecc4c99031f5e9ae99d9201: Linux 4.11-rc1 (2017-03-05 12:59:56 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.11-rc4 for you to fetch changes up to 2ddaa6762696db4bb35726063b421782a9e0436f: Merge remote-tracking branches 'asoc/fix/rt5665', 'asoc/fix/simple', 'asoc/fix/sti' and 'asoc/fix/sun8i' into asoc-linus (2017-03-29 12:55:08 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.11 A relatively large pile of fixes for mainline, the first since the merge window. The biggest block of changes here by volume is the sun8i-codec set, the driver was newly added in the merge window but it was realized that renaming some of the user visible controls was required so these are being pushed for v4.11 to avoid the original code appearing in a release. Otherwise it's all fairly standard bugfix stuff. ---------------------------------------------------------------- Arnaud Pouliquen (1): ASoC: STI: Fix reader substream pointer set Arnd Bergmann (1): ASoC: mediatek: add I2C dependency for CS42XX8 Bard Liao (6): ASoC: rt5665: fix getting wrong work handler container ASoC: rt5665: increase LDO level ASoC: rt5665: Vref3 is necessary for Mono Amp ASoC: rt5665: CLKDET is also a power of ASRC ASoC: rt5665: fix define of RT5665_HP_DRIVER_5X ASoC: rt5665: fix wrong shift rt5665_if2_1_adc_in_enum Brian Norris (1): ASoC: don't dereference NULL pcm_{new,free} Charles Keepax (2): ASoC: wm_adsp: Return an error on write to a disabled volatile control ASoC: wm_adsp: Acknowledge controls should also check the DSP is running Hiroyuki Yokoyama (1): ASoC: rsnd: fix sound route path when using SRC6/SRC9 Jeeja KP (2): ASoC: hdac_hdmi: avoid reference to invalid variable of the pin list ASoC: hdac_hdmi: don't update the iterator in pcm list remove Kuninori Morimoto (3): ASoC: rcar: avoid SSI_MODEx settings for SSI8 ASoC: rcar: clear DE bit only in PDMACHCR when it stops ASoC: rcar: dma: remove unnecessary "volatile" Lucas Stach (1): ASoC: simple-card: fix simple_dai clk lookup Mark Brown (4): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/atmel', 'asoc/fix/hdac-hdmi' and 'asoc/fix/mtk' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5665', 'asoc/fix/simple', 'asoc/fix/sti' and 'asoc/fix/sun8i' into asoc-linus Mylène Josserand (5): ASoC: sun8i-codec: Remove analog "HP" widget ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE ASoC: sun8i-codec: Fix space on audio-routing widget ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN ARM: dts: sun8i: Update audio-routing with renamed widgets Songjun Wu (1): ASoC: atmel-classd: fix audio clock rate Takashi Sakamoto (1): ASoC: Intel: Skylake: fix invalid memory access due to wrong reference of pointer arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- sound/soc/atmel/atmel-classd.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 16 ++++---- sound/soc/codecs/rt5665.c | 10 +++-- sound/soc/codecs/rt5665.h | 2 +- sound/soc/codecs/wm_adsp.c | 9 ++++- sound/soc/generic/simple-card-utils.c | 1 + sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/mediatek/Kconfig | 2 +- sound/soc/sh/rcar/cmd.c | 36 ++++++++++-------- sound/soc/sh/rcar/dma.c | 18 ++++++++- sound/soc/sh/rcar/ssiu.c | 6 ++- sound/soc/soc-core.c | 8 +++- sound/soc/sti/uniperif_reader.c | 3 ++ sound/soc/sunxi/sun8i-codec.c | 67 +++++++++++++++------------------- 15 files changed, 109 insertions(+), 77 deletions(-)

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[alsa-devel] New focusrite USB ID
by Gunter Königsmann 30 Mar '17

30 Mar '17

Dear all, The ardour-users mailing list mehtions that there is a new USB-ID to be found from focusrite: 1235:8004 is the 18i6. Source: http://lists.ardour.org/pipermail/ardour-users-ardour.org/2017-March/028365… As I don't have this specific hardware I cannot help with the testing but I thought I can at least report this thanking the alsa programmers as the focusrite scarlett I have at home works like a charm on linux. Kind regards, Gunter.

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[alsa-devel] No Jack sound on Lenovo dock with a T470 connected
by dmummenschanz＠web.de 30 Mar '17

30 Mar '17

Hello, I'm unable to get sound via the jack headphone connection on my lenovo dock when a t470 is connected. Some years ago I've experienced a similar issue with a t440p connected to the same docking station. Back than a patch helped to activate the jack sound: [codec] 0x10ec0292 0x17aa220e 0 [pincfg] 0x16 0x21211010 0x19 0x21a11010 So I've tried to match codec and pincfg to my t470 without success unfortunately: [codec] 0x10ec0298 0x17aa2245 0 [pincfg] 0x17 0x21211010 0x19 0x21a11010 /proc/asound/card0/codec#0: Codec: Realtek ALC298 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0298 Subsystem Id: 0x17aa2245 Revision Id: 0x100103 No Modem Function Group found Default PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=8, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Line Out Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x7f 0x7f] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x7f 0x7f] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0x20 0x20] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x25 Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Dock Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Dock Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 4 0x18 0x19 0x1a 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Connection: 1 0x03 Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x24 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x03 0x03] Pincap 0x00000020: IN Pin Default 0x90a60140: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40000000: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x14 [Pin Complex] wcaps 0x40050d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010010: OUT EAPD EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 3 0x0c* 0x0d 0x06 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a11030: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Dock Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001373c: IN OUT HP EAPD Detect Vref caps: HIZ 50 GRD 80 100 EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40648605: [N/A] Modem Line at Ext N/A Conn = RCA, Color = Purple DefAssociation = 0x0, Sequence = 0x5 Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1e [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=150 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Line Out Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x03211020: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a 0x1d 0x0b 0x12* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a* 0x1d 0x0b 0x13 Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x12* 0x13 Node 0x25 [Audio Selector] wcaps 0x300101: Stereo Connection: 1 0x1f Any Clues?

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[alsa-devel] [PATCH] ALSA: hda - set intel audio clock to a properly value
by libin.yang＠intel.com 30 Mar '17

30 Mar '17

From: Libin Yang <libin.yang(a)intel.com> On some Intel platforms, the audio clock may not be set correctly with initial setting. This will cause the audio playback/capture rates wrong. This patch checks the audio clock setting and will set it to a properly value if it is not set correct. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188411 Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/hda_register.h | 12 +++-- sound/hda/ext/hdac_ext_controller.c | 6 +-- sound/pci/hda/hda_intel.c | 91 +++++++++++++++++++++++++++++++++++++ 3 files changed, 103 insertions(+), 6 deletions(-) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0013063..7ea16cb 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -227,6 +227,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_PPLCLLPU 0xC /* registers for Multiple Links Capability Structure */ +/* Multiple Links Capability */ +#define AZX_REG_ML_CAP_BASE 0xc00 #define AZX_ML_CAP_ID 0x2 #define AZX_REG_ML_MLCH 0x00 #define AZX_REG_ML_MLCD 0x04 @@ -243,9 +245,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 -#define AZX_MLCTL_SPA (1<<16) -#define AZX_MLCTL_CPA 23 - +#define AZX_REG_ML_LCAPx(x) (AZX_REG_ML_CAP_BASE + (0x40 + 0x40 * x)) +#define AZX_REG_ML_LCTLx(x) (AZX_REG_ML_CAP_BASE + (0x44 + 0x40 * x)) +#define ML_LCTL_SCF_MASK 0xF +#define AZX_MLCTL_SPA (0x1 << 16) +#define AZX_MLCTL_CPA (0x1 << 23) +#define AZX_MLCTL_SPA_SHIFT 16 +#define AZX_MLCTL_CPA_SHIFT 23 /* registers for DMA Resume Capability Structure */ #define AZX_DRSM_CAP_ID 0x5 diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 2614691..84f3b81 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -171,7 +171,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) { int timeout; u32 val; - int mask = (1 << AZX_MLCTL_CPA); + int mask = (1 << AZX_MLCTL_CPA_SHIFT); udelay(3); timeout = 150; @@ -179,10 +179,10 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) do { val = readl(link->ml_addr + AZX_REG_ML_LCTL); if (enable) { - if (((val & mask) >> AZX_MLCTL_CPA)) + if (((val & mask) >> AZX_MLCTL_CPA_SHIFT)) return 0; } else { - if (!((val & mask) >> AZX_MLCTL_CPA)) + if (!((val & mask) >> AZX_MLCTL_CPA_SHIFT)) return 0; } udelay(3); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8256a8..017f64f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -539,6 +539,94 @@ static void bxt_reduce_dma_latency(struct azx *chip) azx_writel(chip, SKL_EM4L, val); } +/* + * ML_LCAP bits: + * bit 0: 6 MHz Supported + * bit 1: 12 MHz Supported + * bit 2: 24 MHz Supported + * bit 3: 48 MHz Supported + * bit 4: 96 MHz Supported + * bit 5: 192 MHz Supported + */ +static int intel_get_lctl_scf(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, ML_LCAPx(0)); + + if (val & (1 << 2)) + return 2; + else if (val & (1 << 3)) + return 3; + else if (val & (1 << 1)) + return 1; + else if (val & (1 << 4)) + return 4; + else if (val & (1 << 5)) + return 5; + + dev_warn(chip->card->dev, "set audio clock frequency to 6MHz"); + return 0; +} + +static void intel_init_lctl(struct azx *chip) +{ + u32 val; + int timeout; + + /* 0. check lctl register value is correct or not */ + /* the codecs are sharing the first link setting by default */ + val = azx_readl(chip, ML_LCTLx(0)); + /* if SCF is already set, let's use it */ + if ((val & ML_LCTL_SCF_MASK) != 0) + return; + + /* + * Before operatiing on SPA, CPA must match SPA. + * Any deviation may result in undefined behavior. + */ + if (((val & AZX_MLCTL_SPA) >> AZX_MLCTL_SPA_SHIFT) ^ + ((val & AZX_MLCTL_CPA) >> AZX_MLCTL_CPA_SHIFT)) + return; + + /* 1. turn link down: set SPA to 0 and wait CPA to 0 */ + val = azx_readl(chip, ML_LCTLx(0)); + val &= ~AZX_MLCTL_SPA; + azx_writel(chip, ML_LCTLx(0), val); + /* wait for CPA */ + timeout = 50; + while (timeout) { + if ((azx_readl(chip, ML_LCTLx(0)) & AZX_MLCTL_CPA) == 0) + break; + timeout--; + udelay(10); + } + if (!timeout) + goto SET_SPA; + /* need add 100ns delay for hardware ready */ + udelay(100); + + /* 2. update SCF to select a properly audio clock*/ + val &= ~ML_LCTL_SCF_MASK; + val |= intel_get_lctl_scf(chip); + azx_writel(chip, ML_LCTLx(0), val); + +SET_SPA: + /* 4. turn link up: set SPA to 1 and wait CPA to 1 */ + val = azx_readl(chip, ML_LCTLx(0)); + val |= AZX_MLCTL_SPA; + azx_writel(chip, ML_LCTLx(0), val); + timeout = 50; + while (timeout) { + if ((azx_readl(chip, ML_LCTLx(0)) & AZX_MLCTL_CPA) != 0) + break; + timeout--; + udelay(10); + } + /* need add 100ns delay for hardware ready */ + udelay(100); +} + static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); @@ -564,6 +652,9 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) /* reduce dma latency to avoid noise */ if (IS_BXT(pci)) bxt_reduce_dma_latency(chip); + + if (IS_SKL_PLUS(pci)) + intel_init_lctl(chip); } /* calculate runtime delay from LPIB */ -- 2.7.4

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[alsa-devel] [PATCH] ALSA: oxygen - Fix snd_oxygen module not loading for some (new?) Xonar DG SI cards.
by Eugene Ganeev 30 Mar '17

30 Mar '17

My Xonar DG SI card is showing up in lspci but no module is loaded for it. The patch just adds a new value with card's PCI ID to oxygen_ids array. Signed-off-by: Eugene Ganeev <easymodofrf(a)gmail.com> --- sound/pci/oxygen/oxygen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ada6c256378e..99ba0354d4cc 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -110,6 +110,7 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, /* Asus Xonar DG */ { OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG }, + { OXYGEN_PCI_SUBID(0x1043, 0x855e), .driver_data = MODEL_XONAR_DG }, /* Asus Xonar DGX */ { OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX }, /* PCI 2.0 HD Audio */ -- 2.11.1

2 3

[alsa-devel] [PATCH 0/3] clkdev: add devm_get_clk_from_child()
by Kuninori Morimoto 30 Mar '17

30 Mar '17

Hi Stephen This is v5 of "clkdev: add devm_of_clk_get()", but new series. I hope my understanding was correct with your idea. Kuninori Morimoto (3): 1) clkdev: add devm_get_clk_from_child() 2) ASoC: simple-card: use devm_get_clk_from_child() 3) ASoC: simple-card-utils: enable clocks/clock-names/clock-ranges .../devicetree/bindings/sound/simple-card.txt | 32 +++++++++++++++ drivers/clk/clk-devres.c | 21 ++++++++++ include/linux/clk.h | 29 ++++++++++++-- include/sound/simple_card_utils.h | 11 +++--- sound/soc/generic/simple-card-utils.c | 45 ++++++++++++++++++++-- sound/soc/generic/simple-card.c | 4 +- sound/soc/generic/simple-scu-card.c | 4 +- 7 files changed, 129 insertions(+), 17 deletions(-) -- 1.9.1

4 20

[alsa-devel] [PATCH] sound: oxygen: simply setting of the shortname for Xonar DG cards
by Clemens Ladisch 29 Mar '17

29 Mar '17

We don't need to manually set the card name; with an entry in the names[] array, this happens automatically. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/pci/oxygen/oxygen.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 74afb6b75976..e36ed8af55ad 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -767,6 +767,8 @@ static int get_oxygen_model(struct oxygen *chip, [MODEL_FANTASIA] = "TempoTec HiFier Fantasia", [MODEL_SERENADE] = "TempoTec HiFier Serenade", [MODEL_HG2PCI] = "CMI8787-HG2PCI", + [MODEL_XONAR_DG] = "Xonar DG", + [MODEL_XONAR_DGX] = "Xonar DGX", }; chip->model = model_generic; @@ -829,12 +831,8 @@ static int get_oxygen_model(struct oxygen *chip, chip->model.dac_channels_mixer = 2; break; case MODEL_XONAR_DG: - chip->model = model_xonar_dg; - chip->model.shortname = "Xonar DG"; - break; case MODEL_XONAR_DGX: chip->model = model_xonar_dg; - chip->model.shortname = "Xonar DGX"; break; } if (id->driver_data == MODEL_MERIDIAN ||

2 1

[alsa-devel] dmix, snd_pcm_drain(), SND_PCM_NONBLOCK - bug?
by Max Kellermann 29 Mar '17

29 Mar '17

Hi, I have received a bug report for the Music Player Daemon which complains about a freezing MPD process, after I switched MPD's ALSA output to non-blocking mode: https://bugs.musicpd.org/view.php?id=4662 This happens when snd_pcm_drain() is used on a dmix PCM which is in SND_PCM_NONBLOCK mode. snd_pcm_drain() will be called repeatedly, because ALSA's file descriptor is always ready. According to my research, the bug is in __snd_pcm_dmix_drain(). This function always returns -EAGAIN when SND_PCM_NONBLOCK is set: if (pcm->mode & SND_PCM_NONBLOCK) return -EAGAIN; This comes very early in the function, before any other relevant checks. It can never finish that way, and calling it again at any time will never do anything. Thus, snd_pcm_drain() appears to be broken in non-blocking mode. Or am I misunderstanding how snd_pcm_drain() is supposed to be used in non-blocking mode? Max

3 4

[alsa-devel] [PATCH] ASoC: WM8962: Let codec driver enable/disable its MCLK
by Daniel Baluta 29 Mar '17

29 Mar '17

From: Nicolin Chen <b42378(a)freescale.com> WM8962 needs its MCLK when powerup in wm8962_resume(). Thus it's better to control the MCLK in codec driver. Thus remove the clock enable in machine dirver accordingly. While at it, get rid of imx_wm8962_remove function since it is now empty. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/imx-wm8962.c | 40 ++++++++-------------------------------- 1 file changed, 8 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 1b60958..3d894d9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -33,7 +33,6 @@ struct imx_wm8962_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - struct clk *codec_clk; unsigned int clk_frequency; }; @@ -163,6 +162,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_wm8962_data *data; + struct clk *codec_clk; int int_port, ext_port; int ret; @@ -231,19 +231,14 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - ret = PTR_ERR(data->codec_clk); + codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(codec_clk)) { + ret = PTR_ERR(codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } - data->clk_frequency = clk_get_rate(data->codec_clk); - ret = clk_prepare_enable(data->codec_clk); - if (ret) { - dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); - goto fail; - } + data->clk_frequency = clk_get_rate(codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -258,10 +253,10 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -277,16 +272,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } - of_node_put(ssi_np); - of_node_put(codec_np); - - return 0; - -clk_fail: - clk_disable_unprepare(data->codec_clk); fail: of_node_put(ssi_np); of_node_put(codec_np); @@ -294,17 +282,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) return ret; } -static int imx_wm8962_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - - if (!IS_ERR(data->codec_clk)) - clk_disable_unprepare(data->codec_clk); - - return 0; -} - static const struct of_device_id imx_wm8962_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8962", }, { /* sentinel */ } @@ -318,7 +295,6 @@ static struct platform_driver imx_wm8962_driver = { .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, - .remove = imx_wm8962_remove, }; module_platform_driver(imx_wm8962_driver); -- 2.7.4

3 5

[alsa-devel] [PATCH v3 0/2] imx-wm8962: Let codec driver enable/disable its MCLK
by Daniel Baluta 29 Mar '17

29 Mar '17

This is a small cleanup patch series which removes MLCK enable/disable from imx-wm8962 machine driver and lets the codec handle that. While reviewing this, Nicolin noticed that codec_clk cleanup is not correct so the second patch fixes this. Changes since v2: * added 2/2 patch to fix codec_clk cleanup * fixed prefix subject Daniel Baluta (2): ASoC: imx-wm8962: Let codec driver enable/disable its MCLK ASoC: imx-wm8962: Fix codec_clk cleanup sound/soc/fsl/imx-wm8962.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) -- 2.7.4

4 10

[alsa-devel] [PATCH v2 00/11] ASoC: Intel: Skylake: More driver updates
by jeeja.kp＠intel.com 29 Mar '17

29 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides updates on the following: o Fix LLCH register read. o Add 16-bit constraint to FE bxt_rt298 machine. o Fix DMA position reporting for capture stream. o Use the sig_bits to define dai bps caps. o Update sig_bits based on converter capability. o Rearrangement of code to cleanup SKL SST library. o Add support for deferred DSP module bind. Changes in v2: - Addressed Pierre's comments for "Fix DMA position reporting for capture stream" patch. B, Jayachandran (1): ALSA: hda: Fix LLCH register read G Kranthi (1): ASoC: Intel: Skylake: Add 16-bit constraint to FE bxt_rt298 machine Hardik T Shah (1): ASoC: Intel: Skylake: Fix DMA position reporting for capture stream Jeeja KP (5): ASoC: Intel: Skylake: Use the sig_bits to define dai bps capability ASoC: hdac_hdmi: Update sig_bits based on converter capability ASoC: Intel: Skylake: Rearrangement of code to cleanup SKL SST library ASoC: Intel: Skylake: Fix module state after unbind and delete ASoC: Intel: Skylake: Add support for deferred DSP module bind Vinod Koul (3): ASoC: Intel: Skylake: Don't unload module when in use ASoC: Intel: Skylake: Remove redundant vmixer handler ASoC: Intel: Skylake: remove hard coded ACPI path sound/hda/hdac_controller.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 3 + sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-nhlt.c | 7 +- sound/soc/intel/skylake/skl-pcm.c | 95 ++++++++++++++++--- sound/soc/intel/skylake/skl-sst-dsp.h | 24 +++-- sound/soc/intel/skylake/skl-sst-ipc.h | 8 ++ sound/soc/intel/skylake/skl-sst-utils.c | 60 ++---------- sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 159 +++++++++++++++++++++++++------- sound/soc/intel/skylake/skl-topology.h | 17 ++-- sound/soc/intel/skylake/skl.h | 1 + 13 files changed, 263 insertions(+), 124 deletions(-) -- 2.5.0

5 21

[alsa-devel] [PATCH] ASoC: cs35l35: Clear reset_gpio on the error path in probe
by Charles Keepax 29 Mar '17

29 Mar '17

The error path in probe attempts to put the device back into reset. Should we fail to get the reset_gpio (such as a probe defer) we will leave the error value in there, which the gpiod_set_value_cansleep on the error path will attempt to deference. Fix this issue by clearing reset_gpio before we head into the error path. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 1d9f332..9688274 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1412,10 +1412,10 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, GPIOD_OUT_LOW); if (IS_ERR(cs35l35->reset_gpio)) { ret = PTR_ERR(cs35l35->reset_gpio); + cs35l35->reset_gpio = NULL; if (ret == -EBUSY) { dev_info(dev, "Reset line busy, assuming shared reset\n"); - cs35l35->reset_gpio = NULL; } else { dev_err(dev, "Failed to get reset GPIO: %d\n", ret); goto err; -- 2.1.4

3 2

[alsa-devel] Applied "ASoC: mediatek: Add mt2701-wm8960 machine driver" to the asoc tree
by Mark Brown 29 Mar '17

29 Mar '17

The patch ASoC: mediatek: Add mt2701-wm8960 machine driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8625c1dbd87631572f8e2c05bc67736b73d6f02f Mon Sep 17 00:00:00 2001 From: Ryder Lee <ryder.lee(a)mediatek.com> Date: Wed, 29 Mar 2017 16:59:31 +0800 Subject: [PATCH] ASoC: mediatek: Add mt2701-wm8960 machine driver Add wm8960 machine driver and config option for MT2701. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/Kconfig | 10 ++ sound/soc/mediatek/mt2701/Makefile | 1 + sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 16 +++ sound/soc/mediatek/mt2701/mt2701-wm8960.c | 176 +++++++++++++++++++++++++++++ 4 files changed, 203 insertions(+) create mode 100644 sound/soc/mediatek/mt2701/mt2701-wm8960.c diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index d7013bde6f45..c6f39040f71f 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -22,6 +22,16 @@ config SND_SOC_MT2701_CS42448 Select Y if you have such device. If unsure select "N". +config SND_SOC_MT2701_WM8960 + tristate "ASoc Audio driver for MT2701 with WM8960 codec" + depends on SND_SOC_MT2701 + select SND_SOC_WM8960 + help + This adds ASoC driver for Mediatek MT2701 boards + with the WM8960 codecs. + Select Y if you have such device. + If unsure select "N". + config SND_SOC_MT8173 tristate "ASoC support for Mediatek MT8173 chip" depends on ARCH_MEDIATEK diff --git a/sound/soc/mediatek/mt2701/Makefile b/sound/soc/mediatek/mt2701/Makefile index 31c3d04d4942..c91deb6aca21 100644 --- a/sound/soc/mediatek/mt2701/Makefile +++ b/sound/soc/mediatek/mt2701/Makefile @@ -17,3 +17,4 @@ obj-$(CONFIG_SND_SOC_MT2701) += snd-soc-mt2701-afe.o # machine driver obj-$(CONFIG_SND_SOC_MT2701_CS42448) += mt2701-cs42448.o +obj-$(CONFIG_SND_SOC_MT2701_WM8960) += mt2701-wm8960.o diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index c7fa3e663463..bc5d4db94de6 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -604,6 +604,22 @@ static struct snd_soc_dai_ops mt2701_btmrg_ops = { static struct snd_soc_dai_driver mt2701_afe_pcm_dais[] = { /* FE DAIs: memory intefaces to CPU */ { + .name = "PCMO0", + .id = MT2701_MEMIF_DL1, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, + .playback = { + .stream_name = "DL1", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE + | SNDRV_PCM_FMTBIT_S32_LE) + }, + .ops = &mt2701_single_memif_dai_ops, + }, + { .name = "PCM_multi", .id = MT2701_MEMIF_DLM, .suspend = mtk_afe_dai_suspend, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c new file mode 100644 index 000000000000..a08ce2323bdc --- /dev/null +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -0,0 +1,176 @@ +/* + * mt2701-wm8960.c -- MT2701 WM8960 ALSA SoC machine driver + * + * Copyright (c) 2017 MediaTek Inc. + * Author: Ryder Lee <ryder.lee(a)mediatek.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 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <sound/soc.h> + +#include "mt2701-afe-common.h" + +static const struct snd_soc_dapm_widget mt2701_wm8960_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), +}; + +static const struct snd_kcontrol_new mt2701_wm8960_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("AMIC"), +}; + +static int mt2701_wm8960_be_ops_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 *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int mclk_rate; + unsigned int rate = params_rate(params); + unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4; + unsigned int div_bck_over_lrck = 64; + + mclk_rate = rate * div_bck_over_lrck * div_mclk_over_bck; + + snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, SND_SOC_CLOCK_OUT); + snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, SND_SOC_CLOCK_IN); + + return 0; +} + +static struct snd_soc_ops mt2701_wm8960_be_ops = { + .hw_params = mt2701_wm8960_be_ops_hw_params +}; + +static struct snd_soc_dai_link mt2701_wm8960_dai_links[] = { + /* FE */ + { + .name = "wm8960-playback", + .stream_name = "wm8960-playback", + .cpu_dai_name = "PCMO0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "wm8960-capture", + .stream_name = "wm8960-capture", + .cpu_dai_name = "PCM0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_capture = 1, + }, + /* BE */ + { + .name = "wm8960-codec", + .cpu_dai_name = "I2S0", + .no_pcm = 1, + .codec_dai_name = "wm8960-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS + | SND_SOC_DAIFMT_GATED, + .ops = &mt2701_wm8960_be_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static struct snd_soc_card mt2701_wm8960_card = { + .name = "mt2701-wm8960", + .owner = THIS_MODULE, + .dai_link = mt2701_wm8960_dai_links, + .num_links = ARRAY_SIZE(mt2701_wm8960_dai_links), + .controls = mt2701_wm8960_controls, + .num_controls = ARRAY_SIZE(mt2701_wm8960_controls), + .dapm_widgets = mt2701_wm8960_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt2701_wm8960_widgets), +}; + +static int mt2701_wm8960_machine_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt2701_wm8960_card; + struct device_node *platform_node, *codec_node; + int ret, i; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt2701_wm8960_dai_links[i].platform_name) + continue; + mt2701_wm8960_dai_links[i].platform_of_node = platform_node; + } + + card->dev = &pdev->dev; + + codec_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt2701_wm8960_dai_links[i].codec_name) + continue; + mt2701_wm8960_dai_links[i].codec_of_node = codec_node; + } + + ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + return ret; + } + + 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; +} + +#ifdef CONFIG_OF +static const struct of_device_id mt2701_wm8960_machine_dt_match[] = { + {.compatible = "mediatek,mt2701-wm8960-machine",}, + {} +}; +#endif + +static struct platform_driver mt2701_wm8960_machine = { + .driver = { + .name = "mt2701-wm8960", + .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = mt2701_wm8960_machine_dt_match, +#endif + }, + .probe = mt2701_wm8960_machine_probe, +}; + +module_platform_driver(mt2701_wm8960_machine); + +/* Module information */ +MODULE_DESCRIPTION("MT2701 WM8960 ALSA SoC machine driver"); +MODULE_AUTHOR("Ryder Lee <ryder.lee(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("mt2701 wm8960 soc card"); + -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: mediatek: add bindings for wm8960 codec machine driver" to the asoc tree
by Mark Brown 29 Mar '17

29 Mar '17

The patch ASoC: mediatek: add bindings for wm8960 codec machine driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 768abad46df941b6f921f358d3b17e51451b29bd Mon Sep 17 00:00:00 2001 From: Ryder Lee <ryder.lee(a)mediatek.com> Date: Wed, 29 Mar 2017 16:59:32 +0800 Subject: [PATCH] ASoC: mediatek: add bindings for wm8960 codec machine driver add device tree bindings for mt2701-wm8960. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/mt2701-wm8960.txt | 24 ++++++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt new file mode 100644 index 000000000000..809b609ea9d0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt @@ -0,0 +1,24 @@ +MT2701 with WM8960 CODEC + +Required properties: +- compatible: "mediatek,mt2701-wm8960-machine" +- mediatek,platform: the phandle of MT2701 ASoC platform +- audio-routing: a list of the connections between audio +- mediatek,audio-codec: the phandles of wm8960 codec +- pinctrl-names: Should contain only one value - "default" +- pinctrl-0: Should specify pin control groups used for this controller. + +Example: + + sound:sound { + compatible = "mediatek,mt2701-wm8960-machine"; + mediatek,platform = <&afe>; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + mediatek,audio-codec = <&wm8960>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + }; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: rcar: remove rsnd_kctrl_remove()
by Kuninori Morimoto 29 Mar '17

29 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rcar driver is trying to remove kctrl when remove time. But, 1) rcar driver can't/shouldn't remove before removing sound card driver, 2) sound card driver will call snd_ctl_dev_free() and removes all kctrls by snd_ctl_remove(). Thus, rsnd_kctrl_remove() is not necessary. Current implementation will get Oops when removing rcar driver after sound card. This patch fix this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 9 --------- sound/soc/sh/rcar/dvc.c | 16 ---------------- sound/soc/sh/rcar/rsnd.h | 3 --- 3 files changed, 28 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5f6e84f..10e5e69 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1086,15 +1086,6 @@ static int __rsnd_kctrl_new(struct rsnd_mod *mod, return 0; } -void _rsnd_kctrl_remove(struct rsnd_kctrl_cfg *cfg) -{ - if (cfg->card && cfg->kctrl) - snd_ctl_remove(cfg->card, cfg->kctrl); - - cfg->card = NULL; - cfg->kctrl = NULL; -} - int rsnd_kctrl_new_m(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index cf8f59c..994fdb7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -218,21 +218,6 @@ static int rsnd_dvc_probe_(struct rsnd_mod *mod, return rsnd_cmd_attach(io, rsnd_mod_id(mod)); } -static int rsnd_dvc_remove_(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) -{ - struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); - - rsnd_kctrl_remove(dvc->volume); - rsnd_kctrl_remove(dvc->mute); - rsnd_kctrl_remove(dvc->ren); - rsnd_kctrl_remove(dvc->rup); - rsnd_kctrl_remove(dvc->rdown); - - return 0; -} - static int rsnd_dvc_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -332,7 +317,6 @@ static struct dma_chan *rsnd_dvc_dma_req(struct rsnd_dai_stream *io, .name = DVC_NAME, .dma_req = rsnd_dvc_dma_req, .probe = rsnd_dvc_probe_, - .remove = rsnd_dvc_remove_, .init = rsnd_dvc_init, .quit = rsnd_dvc_quit, .pcm_new = rsnd_dvc_pcm_new, diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e59dec0..a0df634 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -613,9 +613,6 @@ struct rsnd_kctrl_cfg_s { u32 val; }; -void _rsnd_kctrl_remove(struct rsnd_kctrl_cfg *cfg); -#define rsnd_kctrl_remove(_cfg) _rsnd_kctrl_remove(&((_cfg).cfg)) - int rsnd_kctrl_new_m(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] Add new machine driver for mt2701
by Ryder Lee 29 Mar '17

29 Mar '17

Hi, This adds wm8960 machine driver and DT bindings document. Ryder Lee (2): ASoC: mediatek: Add mt2701-wm8960 machine driver ASoC: mediatek: add bindings for wm8960 codec .../devicetree/bindings/sound/mt2701-wm8960.txt | 24 +++ sound/soc/mediatek/Kconfig | 10 ++ sound/soc/mediatek/mt2701/Makefile | 1 + sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 16 ++ sound/soc/mediatek/mt2701/mt2701-wm8960.c | 176 +++++++++++++++++++++ 5 files changed, 227 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-wm8960.txt create mode 100644 sound/soc/mediatek/mt2701/mt2701-wm8960.c -- 1.9.1

1 2

[alsa-devel] [PATCH 1/1] pcm:file: delegate htimestamping to slave instead of always getting real_htimestamp
by sutar.mounesh＠gmail.com 29 Mar '17

29 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> purpose of this fix, is to read most accurate timestamps. From documentation of /src/pcm/pcm.c, we can see: """" \par Timestamp mode The timestamp mode specifies, if timestamps are activated. Currently, only #SND_PCM_TSTAMP_NONE and #SND_PCM_TSTAMP_MMAP modes are known. The mmap mode means that timestamp is taken on every period time boundary. Corresponding position in the ring buffer assigned to timestamp can be obtained using #snd_pcm_htimestamp() function. """" As snd_pcm_generic_htimestamp() internally calls snd_pcm_htimestamp() to read time, so accurate timestamp can be read from snd_pcm_generic_htimestamp(). Also, in case of pcm_file, if the underlying slave is hardware, then we would wish to read elapsed hardware time, as it will be the most accurate, as opposed to the elapsed wall time. This will provide pcm_file with the most accurate timestamps. Following are the timesamps read with timestamp enabled, for with fix and without fix scenarios: 1> With fix: :~#time aplay --enable-tstamp -Dhtstamp_test --period-time=5000 -v -fdat /dev/urandom Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo File PCM (file=/tmp/swarate_out.wav) Final file PCM (file=/tmp/swarate_out.wav) .. Slave: Hardware PCM card 0 'imx6q-sabresd-wm8962' device 0 subdevice 0 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 : 24000 period_size : 240 period_time : 5000 tstamp_mode : ENABLE . . Before sleep = 142:409.807623 After sleep = 142:409.807623 Before sleep = 142:414.806016 (calling snd_pcm_htimestamp(handle, &avail, &tstamp_before)) sleep of 2 milisec After sleep = 142:414.806016 (calling snd_pcm_htimestamp(handle, &avail, &tstamp_after) >From the above timestamps, we can see that slave has returned the same timestamps, as --period-time choosen is 5msec. 2> Without this fix: The timestamps are returned with realtime value. :~# time aplay --enable-tstamp -Dhtstamp_test --period-time=5000 -v -fdat /dev/urandom Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo File PCM (file=/tmp/swarate_out.wav) Final file PCM (file=/tmp/swarate_out.wav) . . Slave: Hardware PCM card 0 'imx6q-sabresd-wm8962' device 0 subdevice 0 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 : 24000 period_size : 240 period_time : 5000 tstamp_mode : ENABLE . . Before sleep = 241:136.875845 (calling snd_pcm_htimestamp(handle, &avail, &tstamp_before)) sleep of 2 milisec After sleep = 241:139.076376 (calling snd_pcm_htimestamp(handle, &avail, &tstamp_after) We can observe here, the timestamps shows time diff of ~2ms, which is the time gap of sleep duration. Before sleep = 241:139.617588 After sleep = 241:141.746845 Before sleep = 241:142.291618 After sleep = 241:144.406406 Before sleep = 241:144.951421 After sleep = 241:147.066118 Before sleep = 241:147.623421 After sleep = 241:149.740573 Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 8012251..fe36a1c 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -728,7 +728,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors_count = snd_pcm_generic_poll_descriptors_count, .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, - .htimestamp = snd_pcm_generic_real_htimestamp, + .htimestamp = snd_pcm_generic_htimestamp, }; /** -- 2.7.4

4 4

[alsa-devel] [PATCH 0/5] Convert to use devm_*() for amba attached modules
by Leo Yan 29 Mar '17

29 Mar '17

When review device driver modules which attach to amba bus, found several modules are not using devm_*() apis to manage resource. As result, some drivers have memory leakage or missing iomem unmapping when rmmod module. And the code has many "goto" tags to handle different failures. So this patch series is to convert to use devm_*() for moudules which are attached to amba bus to manage resource and get more robust and neat code. Patch 0003 "drivers/rtc/rtc-pl031.c: Convert to use devm_*()" has been verified on 96boards Hikey. Other patches can pass building but have not really tested on hardware. Leo Yan (5): Input: ambakmi - Convert to use devm_*() drivers/rtc/rtc-pl030.c: Convert to use devm_*() drivers/rtc/rtc-pl031.c: Convert to use devm_*() vfio: platform: Convert to use devm_*() ALSA: AACI: Convert to use devm_ioremap_resource() drivers/input/serio/ambakmi.c | 44 +++++++---------------------- drivers/rtc/rtc-pl030.c | 49 ++++++++------------------------ drivers/rtc/rtc-pl031.c | 59 ++++++++++----------------------------- drivers/vfio/platform/vfio_amba.c | 25 ++++++----------- sound/arm/aaci.c | 21 ++++---------- 5 files changed, 51 insertions(+), 147 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Add support for Maxim Integrated MAX98927 Amplifier
by Ryan Lee 29 Mar '17

29 Mar '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Resubmit the intial version of MAX98927 driver. Added all fixes into the initial patch. .../devicetree/bindings/sound/max98927.txt | 39 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98927.c | 1040 ++++++++++++++++++++ sound/soc/codecs/max98927.h | 267 +++++ 5 files changed, 1353 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/max98927.txt mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/max98927.c create mode 100755 sound/soc/codecs/max98927.h diff --git a/Documentation/devicetree/bindings/sound/max98927.txt b/Documentation/devicetree/bindings/sound/max98927.txt new file mode 100755 index 0000000..d610879 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98927.txt @@ -0,0 +1,39 @@ +max98927 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98927" + + - maxim,vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + This property can be set to values from 0 to 15 for slot 0 to 15. + Default value is 0. + + - maxim,imon-slot-no : slot number used to send current information + This property can be set to values from 0 to 15 for slot 0 to 15. + Default value is 0. + + - maxim,interleave-mode : When using two MAX98927 in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + This property can be set to values from 0 to 1 for off and on. + Default value is 0. + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98927@3a { + compatible = "maxim,max98927"; + maxim,vmon-slot-no = <0>; + maxim,imon-slot-no = <1>; + maxim,interleave-mode = <0>; + reg = <0x3a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index e49e9da..cbdc8aa --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -89,6 +89,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C + select SND_SOC_MAX98927 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C @@ -588,6 +589,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX98926 tristate +config SND_SOC_MAX98927 + tristate "Maxim Integrated MAX98927 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 1796cb9..187a639 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o +snd-soc-max98927-objs := max98927.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o @@ -313,6 +314,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o +obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c new file mode 100755 index 0000000..daf4af7 --- /dev/null +++ b/sound/soc/codecs/max98927.c @@ -0,0 +1,1040 @@ +/* + * max98927.c -- MAX98927 ALSA Soc Audio driver + * + * Copyright (C) 2016 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/acpi.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98927.h" + +static struct reg_default max98927_reg[] = { + {MAX98927_R0001_INT_RAW1, 0x00}, + {MAX98927_R0002_INT_RAW2, 0x00}, + {MAX98927_R0003_INT_RAW3, 0x00}, + {MAX98927_R0004_INT_STATE1, 0x00}, + {MAX98927_R0005_INT_STATE2, 0x00}, + {MAX98927_R0006_INT_STATE3, 0x00}, + {MAX98927_R0007_INT_FLAG1, 0x00}, + {MAX98927_R0008_INT_FLAG2, 0x00}, + {MAX98927_R0009_INT_FLAG3, 0x00}, + {MAX98927_R000A_INT_EN1, 0x00}, + {MAX98927_R000B_INT_EN2, 0x00}, + {MAX98927_R000C_INT_EN3, 0x00}, + {MAX98927_R000D_INT_FLAG_CLR1, 0x00}, + {MAX98927_R000E_INT_FLAG_CLR2, 0x00}, + {MAX98927_R000F_INT_FLAG_CLR3, 0x00}, + {MAX98927_R0010_IRQ_CTRL, 0x00}, + {MAX98927_R0011_CLK_MON, 0x00}, + {MAX98927_R0012_WDOG_CTRL, 0x00}, + {MAX98927_R0013_WDOG_RST, 0x00}, + {MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH, 0x00}, + {MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH, 0x00}, + {MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS, 0x00}, + {MAX98927_R0017_PIN_CFG, 0x55}, + {MAX98927_R0018_PCM_RX_EN_A, 0x00}, + {MAX98927_R0019_PCM_RX_EN_B, 0x00}, + {MAX98927_R001A_PCM_TX_EN_A, 0x00}, + {MAX98927_R001B_PCM_TX_EN_B, 0x00}, + {MAX98927_R001C_PCM_TX_HIZ_CTRL_A, 0x00}, + {MAX98927_R001D_PCM_TX_HIZ_CTRL_B, 0x00}, + {MAX98927_R001E_PCM_TX_CH_SRC_A, 0x00}, + {MAX98927_R001F_PCM_TX_CH_SRC_B, 0x00}, + {MAX98927_R0020_PCM_MODE_CFG, 0x40}, + {MAX98927_R0021_PCM_MASTER_MODE, 0x00}, + {MAX98927_R0022_PCM_CLK_SETUP, 0x22}, + {MAX98927_R0023_PCM_SR_SETUP1, 0x00}, + {MAX98927_R0024_PCM_SR_SETUP2, 0x00}, + {MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, 0x00}, + {MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, 0x00}, + {MAX98927_R0027_ICC_RX_EN_A, 0x00}, + {MAX98927_R0028_ICC_RX_EN_B, 0x00}, + {MAX98927_R002B_ICC_TX_EN_A, 0x00}, + {MAX98927_R002C_ICC_TX_EN_B, 0x00}, + {MAX98927_R002E_ICC_HIZ_MANUAL_MODE, 0x00}, + {MAX98927_R002F_ICC_TX_HIZ_EN_A, 0x00}, + {MAX98927_R0030_ICC_TX_HIZ_EN_B, 0x00}, + {MAX98927_R0031_ICC_LNK_EN, 0x00}, + {MAX98927_R0032_PDM_TX_EN, 0x00}, + {MAX98927_R0033_PDM_TX_HIZ_CTRL, 0x00}, + {MAX98927_R0034_PDM_TX_CTRL, 0x00}, + {MAX98927_R0035_PDM_RX_CTRL, 0x00}, + {MAX98927_R0036_AMP_VOL_CTRL, 0x00}, + {MAX98927_R0037_AMP_DSP_CFG, 0x02}, + {MAX98927_R0038_TONE_GEN_DC_CFG, 0x00}, + {MAX98927_R0039_DRE_CTRL, 0x01}, + {MAX98927_R003A_AMP_EN, 0x00}, + {MAX98927_R003B_SPK_SRC_SEL, 0x00}, + {MAX98927_R003C_SPK_GAIN, 0x00}, + {MAX98927_R003D_SSM_CFG, 0x01}, + {MAX98927_R003E_MEAS_EN, 0x00}, + {MAX98927_R003F_MEAS_DSP_CFG, 0x04}, + {MAX98927_R0040_BOOST_CTRL0, 0x00}, + {MAX98927_R0041_BOOST_CTRL3, 0x00}, + {MAX98927_R0042_BOOST_CTRL1, 0x00}, + {MAX98927_R0043_MEAS_ADC_CFG, 0x00}, + {MAX98927_R0044_MEAS_ADC_BASE_MSB, 0x00}, + {MAX98927_R0045_MEAS_ADC_BASE_LSB, 0x00}, + {MAX98927_R0046_ADC_CH0_DIVIDE, 0x00}, + {MAX98927_R0047_ADC_CH1_DIVIDE, 0x00}, + {MAX98927_R0048_ADC_CH2_DIVIDE, 0x00}, + {MAX98927_R0049_ADC_CH0_FILT_CFG, 0x00}, + {MAX98927_R004A_ADC_CH1_FILT_CFG, 0x00}, + {MAX98927_R004B_ADC_CH2_FILT_CFG, 0x00}, + {MAX98927_R004C_MEAS_ADC_CH0_READ, 0x00}, + {MAX98927_R004D_MEAS_ADC_CH1_READ, 0x00}, + {MAX98927_R004E_MEAS_ADC_CH2_READ, 0x00}, + {MAX98927_R0051_BROWNOUT_STATUS, 0x00}, + {MAX98927_R0052_BROWNOUT_EN, 0x00}, + {MAX98927_R0053_BROWNOUT_INFINITE_HOLD, 0x00}, + {MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR, 0x00}, + {MAX98927_R0055_BROWNOUT_LVL_HOLD, 0x00}, + {MAX98927_R005A_BROWNOUT_LVL1_THRESH, 0x00}, + {MAX98927_R005B_BROWNOUT_LVL2_THRESH, 0x00}, + {MAX98927_R005C_BROWNOUT_LVL3_THRESH, 0x00}, + {MAX98927_R005D_BROWNOUT_LVL4_THRESH, 0x00}, + {MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS, 0x00}, + {MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL, 0x00}, + {MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL, 0x00}, + {MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE, 0x00}, + {MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT, 0x00}, + {MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1, 0x00}, + {MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2, 0x00}, + {MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3, 0x00}, + {MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT, 0x00}, + {MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1, 0x00}, + {MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2, 0x00}, + {MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3, 0x00}, + {MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT, 0x00}, + {MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1, 0x00}, + {MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2, 0x00}, + {MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3, 0x00}, + {MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT, 0x00}, + {MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, 0x00}, + {MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2, 0x00}, + {MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3, 0x00}, + {MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0x00}, + {MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY, 0x00}, + {MAX98927_R0084_ENV_TRACK_REL_RATE, 0x00}, + {MAX98927_R0085_ENV_TRACK_HOLD_RATE, 0x00}, + {MAX98927_R0086_ENV_TRACK_CTRL, 0x00}, + {MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, 0x00}, + {MAX98927_R00FF_GLOBAL_SHDN, 0x00}, + {MAX98927_R0100_SOFT_RESET, 0x00}, + {MAX98927_R01FF_REV_ID, 0x40}, +}; + +static int max98927_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int data; + + regmap_read(max98927->regmap, reg, &data); + ucontrol->value.integer.value[0] = (data & mask) >> shift; + return 0; +} + +static int max98927_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + regmap_update_bits(max98927->regmap, reg, mask, sel << shift); + dev_dbg(codec->dev, "%s: register 0x%02X, value 0x%02X\n", + __func__, reg, sel); + return 0; +} + +static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int mode = 0; + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + mode = MAX98927_PCM_MASTER_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98927->master = true; + mode = MAX98927_PCM_MASTER_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + mode = MAX98927_PCM_MASTER_MODE_HYBRID; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MASK, + mode); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + max98927->iface |= SND_SOC_DAIFMT_I2S; + + break; + case SND_SOC_DAIFMT_LEFT_J: + max98927->iface |= SND_SOC_DAIFMT_LEFT_J; + break; + default: + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + max98927->iface, max98927->iface); + + /* pcm channel configuration */ + if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN| + MAX98927_PCM_RX_CH1_EN); + + return 0; +} + +/* codec MCLK rate in master mode */ +static const int rate_table[] = { + 5644800, 6000000, 6144000, 6500000, + 9600000, 11289600, 12000000, 12288000, + 13000000, 19200000, +}; + +static int max98927_set_clock(struct max98927_priv *max98927, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_codec *codec = max98927->codec; + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98927->ch_size; + int reg = MAX98927_R0022_PCM_CLK_SETUP; + int mask = MAX98927_PCM_CLK_SETUP_BSEL_MASK; + int value; + + if (max98927->master) { + int i; + /* match rate to closest value */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i] >= max98927->sysclk) + break; + } + if (i == ARRAY_SIZE(rate_table)) { + dev_err(codec->dev, "failed to find proper clock rate.\n"); + return -EINVAL; + } + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MCLK_MASK, + i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); + } + + switch (blr_clk_ratio) { + case 32: + value = 2; + break; + case 48: + value = 3; + break; + case 64: + value = 4; + break; + default: + return -EINVAL; + } + regmap_update_bits(max98927->regmap, reg, mask, value); + return 0; +} + +static int max98927_dai_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 max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + max98927->ch_size = 16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + max98927->ch_size = 24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + max98927->ch_size = 32; + break; + default: + dev_err(codec->dev, "format unsupported %d", + params_format(params)); + goto err; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(codec->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_8000; + break; + case 11025: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_11025; + break; + case 12000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_12000; + break; + case 16000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_16000; + break; + case 22050: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_22050; + break; + case 24000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_24000; + break; + case 32000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_32000; + break; + case 44100: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_44100; + break; + case 48000: + sampling_rate |= MAX98927_PCM_SR_SET1_SR_48000; + break; + default: + dev_err(codec->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98927->regmap, + MAX98927_R0023_PCM_SR_SETUP1, + MAX98927_PCM_SR_SET1_SR_MASK, + sampling_rate); + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_SR_MASK, + sampling_rate << MAX98927_PCM_SR_SET2_SR_SHIFT); + + /* set sampling rate of IV */ + if (max98927->interleave_mode && + sampling_rate > MAX98927_PCM_SR_SET1_SR_16000) + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate - 3); + else + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate); + return max98927_set_clock(max98927, params); +err: + return -EINVAL; +} + +#define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->sysclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops max98927_dai_ops = { + .set_sysclk = max98927_dai_set_sysclk, + .set_fmt = max98927_dai_set_fmt, + .hw_params = max98927_dai_hw_params, +}; + +static int max98927_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 1); + /* enable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN); + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 1); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 0); + /* disable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, 0); + break; + default: + return 0; + } + return 0; +} + +static const char * const max98927_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT, + 3, max98927_switch_text); + +static const struct snd_kcontrol_new max98927_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_soc_dapm_widget max98927_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98927_R003A_AMP_EN, + 0, 0, max98927_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, + &max98927_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static DECLARE_TLV_DB_SCALE(max98927_spk_tlv, 300, 300, 0); +static DECLARE_TLV_DB_SCALE(max98927_digital_tlv, -1600, 25, 0); + +static int max98927_spk_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98927->spk_gain; + dev_dbg(codec->dev, "%s: spk_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98927_spk_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + /* Active range 0 ~ 6 */ + if (sel > 6) + return -EINVAL; + + regmap_update_bits(max98927->regmap, + MAX98927_R003C_SPK_GAIN, + MAX98927_SPK_PCM_GAIN_MASK, sel); + max98927->spk_gain = sel; + return 0; +} + +static int max98927_digital_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98927->digital_gain; + dev_dbg(codec->dev, "%s: digital_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98927_digital_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + /* Active range 0 ~ 0x7F */ + if (sel < 128) { + regmap_update_bits(max98927->regmap, + MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_MASK, sel); + max98927->digital_gain = sel; + } + return 0; +} + +static int max98927_boost_voltage_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R0040_BOOST_CTRL0, + MAX98927_BOOST_CTRL0_VOUT_MASK, 0); +} + +static int max98927_boost_voltage_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R0040_BOOST_CTRL0, + MAX98927_BOOST_CTRL0_VOUT_MASK, 0); +} + +static int max98927_amp_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_SEL, + MAX98927_AMP_VOL_SHIFT); +} + +static int max98927_amp_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_SEL, + MAX98927_AMP_VOL_SHIFT); +} + +static int max98927_amp_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R0052_BROWNOUT_EN, + MAX98927_BROWNOUT_DSP_EN, + MAX98927_BROWNOUT_DSP_SHIFT); +} + +static int max98927_amp_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R0052_BROWNOUT_EN, + MAX98927_BROWNOUT_DSP_EN, + MAX98927_BROWNOUT_DSP_SHIFT); +} + +static int max98927_ramp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R0037_AMP_DSP_CFG, + MAX98927_AMP_DSP_CFG_RMP_BYPASS, + MAX98927_AMP_DSP_CFG_RMP_SHIFT); +} +static int max98927_ramp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R0037_AMP_DSP_CFG, + MAX98927_AMP_DSP_CFG_RMP_BYPASS, + MAX98927_AMP_DSP_CFG_RMP_SHIFT); +} + +static int max98927_dre_en_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R0039_DRE_CTRL, + MAX98927_DRE_CTRL_DRE_EN, 0); +} +static int max98927_dre_en_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R0039_DRE_CTRL, + MAX98927_DRE_CTRL_DRE_EN, 0); +} +static int max98927_spk_src_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R003B_SPK_SRC_SEL, + MAX98927_SPK_SRC_MASK, 0); +} + +static int max98927_spk_src_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R003B_SPK_SRC_SEL, + MAX98927_SPK_SRC_MASK, 0); +} + +static int max98927_mono_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT); +} + +static int max98927_mono_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT); +} + +static bool max98927_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0028_ICC_RX_EN_B: + case MAX98927_R002B_ICC_TX_EN_A ... MAX98927_R002C_ICC_TX_EN_B: + case MAX98927_R002E_ICC_HIZ_MANUAL_MODE + ... MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS + ... MAX98927_R0055_BROWNOUT_LVL_HOLD: + case MAX98927_R005A_BROWNOUT_LVL1_THRESH + ... MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE: + case MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT + ... MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R00FF_GLOBAL_SHDN: + case MAX98927_R0100_SOFT_RESET: + case MAX98927_R01FF_REV_ID: + return true; + } + return false; +}; + +static bool max98927_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + return true; + default: + return false; + } +} + +static const char * const max98927_boost_voltage_text[] = { + "6.5V", "6.625V", "6.75V", "6.875V", "7V", "7.125V", "7.25V", "7.375V", + "7.5V", "7.625V", "7.75V", "7.875V", "8V", "8.125V", "8.25V", "8.375V", + "8.5V", "8.625V", "8.75V", "8.875V", "9V", "9.125V", "9.25V", "9.375V", + "9.5V", "9.625V", "9.75V", "9.875V", "10V" +}; + +static const char * const max98927_speaker_source_text[] = { + "i2s", "reserved", "tone", "pdm" +}; + +static const char * const max98927_monomix_output_text[] = { + "ch_0", "ch_1", "ch_1_2_div" +}; + +static const struct soc_enum max98927_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98927_monomix_output_text), + max98927_monomix_output_text), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98927_speaker_source_text), + max98927_speaker_source_text), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98927_boost_voltage_text), + max98927_boost_voltage_text), +}; + +static const struct snd_kcontrol_new max98927_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Volume", + MAX98927_R003C_SPK_GAIN, + 0, 6, 0, + max98927_spk_gain_get, max98927_spk_gain_put, + max98927_spk_tlv), + SOC_SINGLE_EXT_TLV("Digital Gain", + MAX98927_R0036_AMP_VOL_CTRL, + 0, (1<<MAX98927_AMP_VOL_WIDTH)-1, 0, + max98927_digital_gain_get, max98927_digital_gain_put, + max98927_digital_tlv), + SOC_SINGLE_EXT("Amp DSP Enable", + MAX98927_R0052_BROWNOUT_EN, + MAX98927_BROWNOUT_DSP_SHIFT, 1, 0, + max98927_amp_dsp_get, max98927_amp_dsp_put), + SOC_SINGLE_EXT("Ramp Switch", + MAX98927_R0037_AMP_DSP_CFG, + MAX98927_AMP_DSP_CFG_RMP_SHIFT, 1, 1, + max98927_ramp_switch_get, max98927_ramp_switch_put), + SOC_SINGLE_EXT("DRE EN", + MAX98927_R0039_DRE_CTRL, + MAX98927_DRE_EN_SHIFT, 1, 0, + max98927_dre_en_get, max98927_dre_en_put), + SOC_SINGLE_EXT("Amp Volume Location", + MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_SEL_SHIFT, 1, 0, + max98927_amp_vol_get, max98927_amp_vol_put), + SOC_ENUM_EXT("Boost Output Voltage", max98927_enum[2], + max98927_boost_voltage_get, max98927_boost_voltage_put), + SOC_ENUM_EXT("Speaker Source", max98927_enum[1], + max98927_spk_src_get, max98927_spk_src_put), + SOC_ENUM_EXT("Monomix Output", max98927_enum[0], + max98927_mono_out_get, max98927_mono_out_put), +}; + +static const struct snd_soc_dapm_route max98927_audio_map[] = { + {"Amp Enable", NULL, "DAI_OUT"}, + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98927_dai[] = { + { + .name = "max98927-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .ops = &max98927_dai_ops, + } +}; + +static int max98927_probe(struct snd_soc_codec *codec) +{ + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int ret = 0, reg = 0; + + max98927->codec = codec; + codec->control_data = max98927->regmap; + codec->cache_bypass = 1; + + /* Software Reset */ + regmap_write(max98927->regmap, + MAX98927_R0100_SOFT_RESET, MAX98927_SOFT_RESET); + + /* Check Revision ID */ + ret = regmap_read(max98927->regmap, + MAX98927_R01FF_REV_ID, &reg); + if (ret < 0) + dev_err(codec->dev, + "Failed to read: 0x%02X\n", MAX98927_R01FF_REV_ID); + else + dev_info(codec->dev, + "MAX98927 revisionID: 0x%02X\n", reg); + + /* IV default slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + 0x80); + regmap_write(max98927->regmap, + MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, + 0x1); + /* Set inital volume (+13dB) */ + regmap_write(max98927->regmap, + MAX98927_R0036_AMP_VOL_CTRL, + 0x38); + regmap_write(max98927->regmap, + MAX98927_R003C_SPK_GAIN, + 0x05); + /* Enable DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R0037_AMP_DSP_CFG, + 0x03); + /* Enable IMON VMON DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R003F_MEAS_DSP_CFG, + 0xF7); + /* Boost Output Voltage & Current limit */ + regmap_write(max98927->regmap, + MAX98927_R0040_BOOST_CTRL0, + 0x1C); + regmap_write(max98927->regmap, + MAX98927_R0042_BOOST_CTRL1, + 0x3E); + /* Measurement ADC config */ + regmap_write(max98927->regmap, + MAX98927_R0043_MEAS_ADC_CFG, + 0x04); + regmap_write(max98927->regmap, + MAX98927_R0044_MEAS_ADC_BASE_MSB, + 0x00); + regmap_write(max98927->regmap, + MAX98927_R0045_MEAS_ADC_BASE_LSB, + 0x24); + /* Brownout Level */ + regmap_write(max98927->regmap, + MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, + 0x06); + /* Envelope Tracking configuration */ + regmap_write(max98927->regmap, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, + 0x08); + regmap_write(max98927->regmap, + MAX98927_R0086_ENV_TRACK_CTRL, + 0x01); + regmap_write(max98927->regmap, + MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, + 0x10); + + /* voltage, current slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001E_PCM_TX_CH_SRC_A, + (max98927->i_l_slot<<MAX98927_PCM_TX_CH_SRC_A_I_SHIFT| + max98927->v_l_slot)&0xFF); + + if (max98927->v_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->v_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->v_l_slot, + 1 << max98927->v_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->v_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->v_l_slot - 8), + 1 << (max98927->v_l_slot - 8)); + } + + if (max98927->i_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->i_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->i_l_slot, + 1 << max98927->i_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->i_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->i_l_slot - 8), + 1 << (max98927->i_l_slot - 8)); + } + + /* Set interleave mode */ + if (max98927->interleave_mode) + regmap_update_bits(max98927->regmap, + MAX98927_R001F_PCM_TX_CH_SRC_B, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK); + return ret; +} + +static const struct snd_soc_codec_driver soc_codec_dev_max98927 = { + .probe = max98927_probe, + .component_driver = { + .controls = max98927_snd_controls, + .num_controls = ARRAY_SIZE(max98927_snd_controls), + .dapm_widgets = max98927_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98927_dapm_widgets), + .dapm_routes = max98927_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98927_audio_map), + }, +}; + +static const struct regmap_config max98927_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98927_R01FF_REV_ID, + .reg_defaults = max98927_reg, + .num_reg_defaults = ARRAY_SIZE(max98927_reg), + .readable_reg = max98927_readable_register, + .volatile_reg = max98927_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static void max98927_slot_config(struct i2c_client *i2c, + struct max98927_priv *max98927) +{ + int value; + + if (!of_property_read_u32(i2c->dev.of_node, + "maxim,vmon-slot-no", &value)) + max98927->v_l_slot = value & 0xF; + else + max98927->v_l_slot = 0; + if (!of_property_read_u32(i2c->dev.of_node, + "maxim,imon-slot-no", &value)) + max98927->i_l_slot = value & 0xF; + else + max98927->i_l_slot = 1; +} + +static int max98927_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + + int ret = 0, value; + struct max98927_priv *max98927 = NULL; + + max98927 = devm_kzalloc(&i2c->dev, + sizeof(*max98927), GFP_KERNEL); + + if (!max98927) { + ret = -ENOMEM; + goto err; + } + i2c_set_clientdata(i2c, max98927); + + /* update interleave mode info */ + if (!of_property_read_u32(i2c->dev.of_node, + "maxim,interleave_mode", &value)) { + if (value > 0) + max98927->interleave_mode = 1; + else + max98927->interleave_mode = 0; + } else + max98927->interleave_mode = 0; + + /* regmap initialization */ + max98927->regmap + = devm_regmap_init_i2c(i2c, &max98927_regmap); + if (IS_ERR(max98927->regmap)) { + ret = PTR_ERR(max98927->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err; + } + + /* voltage/current slot configuration */ + max98927_slot_config(i2c, max98927); + + /* codec registeration */ + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98927, + max98927_dai, ARRAY_SIZE(max98927_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; + +err: + if (max98927) + devm_kfree(&i2c->dev, max98927); + return ret; +} + +static int max98927_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98927_i2c_id[] = { + { "max98927", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98927_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98927_of_match[] = { + { .compatible = "maxim,max98927", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98927_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98927_acpi_match[] = { + { "MX98927", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98927_acpi_match); +#endif + +static struct i2c_driver max98927_i2c_driver = { + .driver = { + .name = "max98927", + .of_match_table = of_match_ptr(max98927_of_match), + .acpi_match_table = ACPI_PTR(max98927_acpi_match), + .pm = NULL, + }, + .probe = max98927_i2c_probe, + .remove = max98927_i2c_remove, + .id_table = max98927_i2c_id, +}; + +module_i2c_driver(max98927_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98927 driver"); +MODULE_AUTHOR("Ryan Lee <ryans.lee(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h new file mode 100755 index 0000000..e13e18a --- /dev/null +++ b/sound/soc/codecs/max98927.h @@ -0,0 +1,267 @@ +/* + * max98927.h -- MAX98927 ALSA Soc Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ +#ifndef _MAX98927_H +#define _MAX98927_H + +/* Register Values */ +#define MAX98927_R0001_INT_RAW1 0x0001 +#define MAX98927_R0002_INT_RAW2 0x0002 +#define MAX98927_R0003_INT_RAW3 0x0003 +#define MAX98927_R0004_INT_STATE1 0x0004 +#define MAX98927_R0005_INT_STATE2 0x0005 +#define MAX98927_R0006_INT_STATE3 0x0006 +#define MAX98927_R0007_INT_FLAG1 0x0007 +#define MAX98927_R0008_INT_FLAG2 0x0008 +#define MAX98927_R0009_INT_FLAG3 0x0009 +#define MAX98927_R000A_INT_EN1 0x000A +#define MAX98927_R000B_INT_EN2 0x000B +#define MAX98927_R000C_INT_EN3 0x000C +#define MAX98927_R000D_INT_FLAG_CLR1 0x000D +#define MAX98927_R000E_INT_FLAG_CLR2 0x000E +#define MAX98927_R000F_INT_FLAG_CLR3 0x000F +#define MAX98927_R0010_IRQ_CTRL 0x0010 +#define MAX98927_R0011_CLK_MON 0x0011 +#define MAX98927_R0012_WDOG_CTRL 0x0012 +#define MAX98927_R0013_WDOG_RST 0x0013 +#define MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH 0x0014 +#define MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH 0x0015 +#define MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS 0x0016 +#define MAX98927_R0017_PIN_CFG 0x0017 +#define MAX98927_R0018_PCM_RX_EN_A 0x0018 +#define MAX98927_R0019_PCM_RX_EN_B 0x0019 +#define MAX98927_R001A_PCM_TX_EN_A 0x001A +#define MAX98927_R001B_PCM_TX_EN_B 0x001B +#define MAX98927_R001C_PCM_TX_HIZ_CTRL_A 0x001C +#define MAX98927_R001D_PCM_TX_HIZ_CTRL_B 0x001D +#define MAX98927_R001E_PCM_TX_CH_SRC_A 0x001E +#define MAX98927_R001F_PCM_TX_CH_SRC_B 0x001F +#define MAX98927_R0020_PCM_MODE_CFG 0x0020 +#define MAX98927_R0021_PCM_MASTER_MODE 0x0021 +#define MAX98927_R0022_PCM_CLK_SETUP 0x0022 +#define MAX98927_R0023_PCM_SR_SETUP1 0x0023 +#define MAX98927_R0024_PCM_SR_SETUP2 0x0024 +#define MAX98927_R0025_PCM_TO_SPK_MONOMIX_A 0x0025 +#define MAX98927_R0026_PCM_TO_SPK_MONOMIX_B 0x0026 +#define MAX98927_R0027_ICC_RX_EN_A 0x0027 +#define MAX98927_R0028_ICC_RX_EN_B 0x0028 +#define MAX98927_R002B_ICC_TX_EN_A 0x002B +#define MAX98927_R002C_ICC_TX_EN_B 0x002C +#define MAX98927_R002E_ICC_HIZ_MANUAL_MODE 0x002E +#define MAX98927_R002F_ICC_TX_HIZ_EN_A 0x002F +#define MAX98927_R0030_ICC_TX_HIZ_EN_B 0x0030 +#define MAX98927_R0031_ICC_LNK_EN 0x0031 +#define MAX98927_R0032_PDM_TX_EN 0x0032 +#define MAX98927_R0033_PDM_TX_HIZ_CTRL 0x0033 +#define MAX98927_R0034_PDM_TX_CTRL 0x0034 +#define MAX98927_R0035_PDM_RX_CTRL 0x0035 +#define MAX98927_R0036_AMP_VOL_CTRL 0x0036 +#define MAX98927_R0037_AMP_DSP_CFG 0x0037 +#define MAX98927_R0038_TONE_GEN_DC_CFG 0x0038 +#define MAX98927_R0039_DRE_CTRL 0x0039 +#define MAX98927_R003A_AMP_EN 0x003A +#define MAX98927_R003B_SPK_SRC_SEL 0x003B +#define MAX98927_R003C_SPK_GAIN 0x003C +#define MAX98927_R003D_SSM_CFG 0x003D +#define MAX98927_R003E_MEAS_EN 0x003E +#define MAX98927_R003F_MEAS_DSP_CFG 0x003F +#define MAX98927_R0040_BOOST_CTRL0 0x0040 +#define MAX98927_R0041_BOOST_CTRL3 0x0041 +#define MAX98927_R0042_BOOST_CTRL1 0x0042 +#define MAX98927_R0043_MEAS_ADC_CFG 0x0043 +#define MAX98927_R0044_MEAS_ADC_BASE_MSB 0x0044 +#define MAX98927_R0045_MEAS_ADC_BASE_LSB 0x0045 +#define MAX98927_R0046_ADC_CH0_DIVIDE 0x0046 +#define MAX98927_R0047_ADC_CH1_DIVIDE 0x0047 +#define MAX98927_R0048_ADC_CH2_DIVIDE 0x0048 +#define MAX98927_R0049_ADC_CH0_FILT_CFG 0x0049 +#define MAX98927_R004A_ADC_CH1_FILT_CFG 0x004A +#define MAX98927_R004B_ADC_CH2_FILT_CFG 0x004B +#define MAX98927_R004C_MEAS_ADC_CH0_READ 0x004C +#define MAX98927_R004D_MEAS_ADC_CH1_READ 0x004D +#define MAX98927_R004E_MEAS_ADC_CH2_READ 0x004E +#define MAX98927_R0051_BROWNOUT_STATUS 0x0051 +#define MAX98927_R0052_BROWNOUT_EN 0x0052 +#define MAX98927_R0053_BROWNOUT_INFINITE_HOLD 0x0053 +#define MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR 0x0054 +#define MAX98927_R0055_BROWNOUT_LVL_HOLD 0x0055 +#define MAX98927_R005A_BROWNOUT_LVL1_THRESH 0x005A +#define MAX98927_R005B_BROWNOUT_LVL2_THRESH 0x005B +#define MAX98927_R005C_BROWNOUT_LVL3_THRESH 0x005C +#define MAX98927_R005D_BROWNOUT_LVL4_THRESH 0x005D +#define MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS 0x005E +#define MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL 0x005F +#define MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL 0x0060 +#define MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE 0x0061 +#define MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT 0x0072 +#define MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1 0x0073 +#define MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2 0x0074 +#define MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3 0x0075 +#define MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT 0x0076 +#define MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1 0x0077 +#define MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2 0x0078 +#define MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3 0x0079 +#define MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT 0x007A +#define MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1 0x007B +#define MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2 0x007C +#define MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3 0x007D +#define MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT 0x007E +#define MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1 0x007F +#define MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2 0x0080 +#define MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3 0x0081 +#define MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM 0x0082 +#define MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY 0x0083 +#define MAX98927_R0084_ENV_TRACK_REL_RATE 0x0084 +#define MAX98927_R0085_ENV_TRACK_HOLD_RATE 0x0085 +#define MAX98927_R0086_ENV_TRACK_CTRL 0x0086 +#define MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ 0x0087 +#define MAX98927_R00FF_GLOBAL_SHDN 0x00FF +#define MAX98927_R0100_SOFT_RESET 0x0100 +#define MAX98927_R01FF_REV_ID 0x01FF + +/* MAX98927_R0018_PCM_RX_EN_A */ +#define MAX98927_PCM_RX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_RX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_RX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_RX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_RX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_RX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_RX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_RX_CH7_EN (0x1 << 7) + +/* MAX98927_R001A_PCM_TX_EN_A */ +#define MAX98927_PCM_TX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_TX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_TX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_TX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_TX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_TX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_TX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_TX_CH7_EN (0x1 << 7) + +/* MAX98927_R001E_PCM_TX_CH_SRC_A */ +#define MAX98927_PCM_TX_CH_SRC_A_V_SHIFT (0) +#define MAX98927_PCM_TX_CH_SRC_A_I_SHIFT (4) + +/* MAX98927_R001F_PCM_TX_CH_SRC_B */ +#define MAX98927_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 5) + +/* MAX98927_R0020_PCM_MODE_CFG */ +#define MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 2) +#define MAX98927_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) + +#define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98927_R0021_PCM_MASTER_MODE */ +#define MAX98927_PCM_MASTER_MODE_MASK (0x3 << 0) +#define MAX98927_PCM_MASTER_MODE_SLAVE (0x0 << 0) +#define MAX98927_PCM_MASTER_MODE_MASTER (0x3 << 0) +#define MAX98927_PCM_MASTER_MODE_HYBRID (0x1 << 0) + +#define MAX98927_PCM_MASTER_MODE_MCLK_MASK (0xF << 2) +#define MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT (2) + +/* MAX98927_R0022_PCM_CLK_SETUP */ +#define MAX98927_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98927_R0023_PCM_SR_SETUP1 */ +#define MAX98927_PCM_SR_SET1_SR_MASK (0xF << 0) + +#define MAX98927_PCM_SR_SET1_SR_8000 (0x0 << 0) +#define MAX98927_PCM_SR_SET1_SR_11025 (0x1 << 0) +#define MAX98927_PCM_SR_SET1_SR_12000 (0x2 << 0) +#define MAX98927_PCM_SR_SET1_SR_16000 (0x3 << 0) +#define MAX98927_PCM_SR_SET1_SR_22050 (0x4 << 0) +#define MAX98927_PCM_SR_SET1_SR_24000 (0x5 << 0) +#define MAX98927_PCM_SR_SET1_SR_32000 (0x6 << 0) +#define MAX98927_PCM_SR_SET1_SR_44100 (0x7 << 0) +#define MAX98927_PCM_SR_SET1_SR_48000 (0x8 << 0) + +/* MAX98927_R0024_PCM_SR_SETUP2 */ +#define MAX98927_PCM_SR_SET2_SR_MASK (0xF << 4) +#define MAX98927_PCM_SR_SET2_SR_SHIFT (4) +#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xf << 0) + +/* MAX98927_R0025_PCM_TO_SPK_MONOMIX_A */ +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT (6) + +/* MAX98927_R0036_AMP_VOL_CTRL */ +#define MAX98927_AMP_VOL_SEL (0x1 << 7) +#define MAX98927_AMP_VOL_SEL_WIDTH (1) +#define MAX98927_AMP_VOL_SEL_SHIFT (7) +#define MAX98927_AMP_VOL_MASK (0x7f << 0) +#define MAX98927_AMP_VOL_WIDTH (7) +#define MAX98927_AMP_VOL_SHIFT (0) + +/* MAX98927_R0037_AMP_DSP_CFG */ +#define MAX98927_AMP_DSP_CFG_DCBLK_EN (0x1 << 0) +#define MAX98927_AMP_DSP_CFG_DITH_EN (0x1 << 1) +#define MAX98927_AMP_DSP_CFG_RMP_BYPASS (0x1 << 4) +#define MAX98927_AMP_DSP_CFG_DAC_INV (0x1 << 5) +#define MAX98927_AMP_DSP_CFG_RMP_SHIFT (4) + +/* MAX98927_R0039_DRE_CTRL */ +#define MAX98927_DRE_CTRL_DRE_EN (0x1 << 0) +#define MAX98927_DRE_EN_SHIFT 0x1 + +/* MAX98927_R003A_AMP_EN */ +#define MAX98927_AMP_EN_MASK (0x1 << 0) + +/* MAX98927_R003B_SPK_SRC_SEL */ +#define MAX98927_SPK_SRC_MASK (0x3 << 0) + +/* MAX98927_R003C_SPK_GAIN */ +#define MAX98927_SPK_PCM_GAIN_MASK (0x7 << 0) +#define MAX98927_SPK_PDM_GAIN_MASK (0x7 << 4) +#define MAX98927_SPK_GAIN_WIDTH (3) + +/* MAX98927_R003E_MEAS_EN */ +#define MAX98927_MEAS_V_EN (0x1 << 0) +#define MAX98927_MEAS_I_EN (0x1 << 1) + +/* MAX98927_R0040_BOOST_CTRL0 */ +#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1f << 0) +#define MAX98927_BOOST_CTRL0_PVDD_MASK (0x1 << 7) +#define MAX98927_BOOST_CTRL0_PVDD_EN_SHIFT (7) + +/* MAX98927_R0052_BROWNOUT_EN */ +#define MAX98927_BROWNOUT_BDE_EN (0x1 << 0) +#define MAX98927_BROWNOUT_AMP_EN (0x1 << 1) +#define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) +#define MAX98927_BROWNOUT_DSP_SHIFT (2) + +/* MAX98927_R0100_SOFT_RESET */ +#define MAX98927_SOFT_RESET (0x1 << 0) + +/* MAX98927_R00FF_GLOBAL_SHDN */ +#define MAX98927_GLOBAL_EN_MASK (0x1 << 0) + +struct max98927_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98927_pdata *pdata; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int v_l_slot; + unsigned int i_l_slot; + bool interleave_mode; + unsigned int ch_size; + unsigned int rate; + unsigned int iface; + unsigned int master; + unsigned int digital_gain; +}; +#endif -- 2.7.4

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[alsa-devel] ALSA Overlay 24bit data 32bit slots
by Ed McNicholas 28 Mar '17

28 Mar '17

I am interfacing with a dsp that we have programmed and there are only options for slot widths of 16 or 32 bits but we are using a 24 bit data format. When I record using arecord -v -f S32_LE -r 48000 -t wav -c4 /home/nyquist/test.wav I get valid audio recording but the lower 8bits is 0 as would be expected. When I record using : arecord -v -f S24_LE -r 48000 -t wav -c4 /home/nyquist/test.wav I get garbage as it appears to be using 24bit wide tdm slots. My overlay is below and I have set what I believe are the correct slot configurations for the device. We have 4 channel input and one channel output 32bit slot width. Any help would be greatly appreciated as I am sure I am missing something. /dts-v1/; /plugin/; #include <dt-bindings/gpio/gpio.h> #include <dt-bindings/pinctrl/am33xx.h> #include <dt-bindings/interrupt-controller/irq.h> #include <dt-bindings/input/linux-event-codes.h> / { compatible = "ti,beaglebone", "ti,beaglebone-black", "ti,beaglebone-green"; /* identification */ part-number = "NQ-ASB-01"; version = "00A0", "A0"; fragment@0 { target = <&am33xx_pinmux>; __overlay__ { bone_audio_cape_audio_pins: pinmux_bone_audio_cape_audio_pins { pinctrl-single,pins = < 0x190 (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_aclkx.mcasp0_aclkx */ 0x194 (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_fsx.mcasp0_fsx, INPUT */ 0x198 (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr0.mcasp0_axr0 */ 0x19c (PIN_INPUT_PULLDOWN | MUX_MODE2) /* mcasp0_ahclkr.mcasp0_axr2 */ 0x1ac (PIN_INPUT_PULLDOWN | MUX_MODE0) /* MCASP0_AHCLKX -> MCASP0_AHCLKX (I2S_MCLK_OUT)- in */ 0x06c ( PIN_OUTPUT_PULLUP | MUX_MODE7 ) /* mcasp0_axr1 GPIO1_27 | OUTPUT | PULLUP */ >; }; /* dsp SPI Pins */ bb_spi0_pins: pinmux_bb_spi0_pins { pinctrl-single,pins = < 0x150 0x30 /* p84 spi0_sclk.spi0_sclk, INPUT_PULLUP | MODE0 */ 0x154 0x30 /* p85 spi0_d0.spi0_d0, INPUT_PULLUP | MODE0 */ 0x158 0x10 /* p86 spi0_d1.spi0_d1, OUTPUT_PULLUP | MODE0 */ 0x15c 0x10 /* p87 spi0_cs0.spi0_cs0, OUTPUT_PULLUP | MODE0 */ >; }; }; }; /* Relay Spi Interface */ fragment@1 { target = <&spi0>; __overlay__ { #address-cells = <1>; #size-cells = <0>; status = "okay"; pinctrl-names = "default"; pinctrl-0 = <&bb_spi0_pins>; ti,pio-mode; /* disable dma when used as an overlay, dma gets stuck at 160 bits... */ ti,pindir-d0-out-d1-in = <1>; dspspi: channel@0 { #address-cells = <1>; #size-cells = <0>; #sound-dai-cells = <0>; compatible = "nq,dsp"; reg = <0>; spi-max-frequency = <16000000>; spi-cpha; }; }; }; /* MCASP Interface */ fragment@2 { target = <&mcasp0>; __overlay__ { #sound-dai-cells = <0>; compatible = "ti,am33xx-mcasp-audio"; pinctrl-names = "default"; pinctrl-0 = <&bone_audio_cape_audio_pins>; status = "okay"; op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <4>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 2 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 >; tx-num-evt = <1>; rx-num-evt = <1>; }; }; /*Sound Interface Config */ fragment@3 { target-path="/"; __overlay__ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "Nyquist Mixer DSP"; simple-audio-card,format = "dsp_a"; simple-audio-card,bitclock-master = <&sound_master>; simple-audio-card,frame-master = <&sound_master>; simple-audio-card,widgets = "Line", "Line In"; simple-audio-card,cpu { sound-dai = <&mcasp0>; dai-tdm-slot-num = <4>; dai-tdm-slot-width = <32>; dai-tdm-slot-tx-mask = <1 0 0 0>; dai-tdm-slot-rx-mask = <1 1 1 1>; }; sound_master: simple-audio-card,codec { sound-dai = <&dspspi>; dai-tdm-slot-num = <4>; dai-tdm-slot-width = <32>; dai-tdm-slot-tx-mask = <1 0 0 0>; dai-tdm-slot-rx-mask = <1 1 1 1>; }; }; }; }; }; Ed McNicholas Sr Embedded Engineer [Bogen Signature Logo]<http://www.bogen.com/> Bogen Communications, Inc. 3862 Quadrangle Blvd Suite 150 Orlando, FL 32817 Direct: (201) 995-2087 Cell: (321) 217-5518

2 1

[alsa-devel] [PATCH 00/18] ALSA: firewire-motu: new driver for MOTU FireWire series
by Takashi Sakamoto 28 Mar '17

28 Mar '17

Hi, This patchset updates my previous RFCv2, and go for ALSA upstream. http://mailman.alsa-project.org/pipermail/alsa-devel/2017-January/117211.ht… This patchset newly adds a driver into ALSA firewire stack, to support for some models in MOTU FireWire series: - 828mk2 - 828mk3 (FireWire only) - 828mk3 (Hybrid) The driver supports playbacking/capturing PCM frames and MIDI messages for any sampling transfer frequencies, with current ALSA IEC 61883-1/6 packet streaming engine. Unfortunately, units on MOTU FireWire series have many quirks against IEC 61883-1/6. This brings much commits and descriptions on this patchset. Sorry for reviewers but I'm glad to get your assist for this patchset. As I note in 8th commit, at sampling transfer frequency based on 44.1kHz, the driver handles the units with choppy noises every few seconds. This is an issue which is not resolved yet. Changes from RFCv2: - drop trial support for 828. If you have interests in development, please refer to one commit of my previous RFC. It describes v1 protocol. - http://mailman.alsa-project.org/pipermail/alsa-devel/2017-January/117226.ht… - improve commit messages. Takashi Sakamoto (18): ALSA: firewire-motu: add skeleton for Mark of the unicorn (MOTU) FireWire series ALSA: firewire-motu: postpone sound card registration ALSA: firewire-motu: add a structure for model-dependent parameters. ALSA: firewire-motu: add an abstraction layer for three types of protocols ALSA: firewire-lib: record cycle count for the first packet ALSA: firewire-lib: add support for source packet header field in CIP header ALSA: firewire-lib: enable CIP_DBC_IS_END_EVENT for both directions of stream ALSA: firewire-motu: add MOTU specific protocol layer ALSA: firewire-motu: handle transactions specific for MOTU FireWire models ALSA: firewire-motu: add stream management functionality ALSA: firewire-motu: add proc node to show current statuc of clock and packet formats ALSA: firewire-motu: add PCM functionality ALSA: firewire-motu: add MIDI functionality ALSA: firewire-motu: add hwdep interface ALSA: firewire-motu: enable to read transaction cache via hwdep interface ALSA: firewire-motu: add support for MOTU 828mk2 as a model with protocol version 2 ALSA: firewire-lib: add a quirk of packet without valid EOH in CIP format ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3 include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 10 +- sound/firewire/Kconfig | 12 + sound/firewire/Makefile | 1 + sound/firewire/amdtp-stream.c | 36 ++- sound/firewire/amdtp-stream.h | 9 +- sound/firewire/motu/Makefile | 4 + sound/firewire/motu/amdtp-motu.c | 388 ++++++++++++++++++++++++++++++++ sound/firewire/motu/motu-hwdep.c | 198 ++++++++++++++++ sound/firewire/motu/motu-midi.c | 169 ++++++++++++++ sound/firewire/motu/motu-pcm.c | 398 +++++++++++++++++++++++++++++++++ sound/firewire/motu/motu-proc.c | 118 ++++++++++ sound/firewire/motu/motu-protocol-v2.c | 237 ++++++++++++++++++++ sound/firewire/motu/motu-protocol-v3.c | 312 ++++++++++++++++++++++++++ sound/firewire/motu/motu-stream.c | 381 +++++++++++++++++++++++++++++++ sound/firewire/motu/motu-transaction.c | 137 ++++++++++++ sound/firewire/motu/motu.c | 264 ++++++++++++++++++++++ sound/firewire/motu/motu.h | 161 +++++++++++++ 18 files changed, 2827 insertions(+), 11 deletions(-) create mode 100644 sound/firewire/motu/Makefile create mode 100644 sound/firewire/motu/amdtp-motu.c create mode 100644 sound/firewire/motu/motu-hwdep.c create mode 100644 sound/firewire/motu/motu-midi.c create mode 100644 sound/firewire/motu/motu-pcm.c create mode 100644 sound/firewire/motu/motu-proc.c create mode 100644 sound/firewire/motu/motu-protocol-v2.c create mode 100644 sound/firewire/motu/motu-protocol-v3.c create mode 100644 sound/firewire/motu/motu-stream.c create mode 100644 sound/firewire/motu/motu-transaction.c create mode 100644 sound/firewire/motu/motu.c create mode 100644 sound/firewire/motu/motu.h -- 2.9.3

2 20

[alsa-devel] [PATCH v2 0/2] ASoC: STI: Fix null ptr deference in IRQ handler
by Arnaud Pouliquen 27 Mar '17

27 Mar '17

V2: - Fix missing sets of reader->substream - Add spinlock to protect player/reader->substream access Arnaud Pouliquen (2): ASoC: STI: Fix reader substream pointer set ASoC: STI: Fix null ptr deference in IRQ handler sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 34 +++++++++++++++++++++++----------- sound/soc/sti/uniperif_reader.c | 27 +++++++++++++++++++++++---- 3 files changed, 47 insertions(+), 15 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH v2] ASoC: intel: Don't print FW version repeatedly
by Takashi Iwai 27 Mar '17

27 Mar '17

Intel SST driver spews an info message "FW Versoin xxxx" at each time the device gets initialized. Since it's triggered at each PM (or even runtime PM), it appears too frequently, which is rather annoying. Moreover, the firmware in the driver is cached, thus it's assured to be always the same version once after loaded; i.e. it's superfluous to show the same version number repeatedly. This patch suppresses the superfluous messages by replacing with dev_info_once(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Use dev_info_oce() sound/soc/intel/atom/sst/sst_ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 14c2d9d18180..65f3696133d3 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -236,7 +236,7 @@ static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, retval = init->result; goto ret; } - dev_info(sst_drv_ctx->dev, "FW Version %02x.%02x.%02x.%02x\n", + dev_info_once(sst_drv_ctx->dev, "FW Version %02x.%02x.%02x.%02x\n", init->fw_version.type, init->fw_version.major, init->fw_version.minor, init->fw_version.build); dev_dbg(sst_drv_ctx->dev, "Build date %s Time %s\n", -- 2.11.1

2 2

[alsa-devel] [PATCH] ASoC: intel: Don't print FW version repeatedly
by Takashi Iwai 27 Mar '17

27 Mar '17

Intel SST driver spews an info message "FW Versoin xxxx" at each time the device gets initialized. Since it's triggered at each PM (or even runtime PM), it appears so ofetn, and rather becomes annoying than useful. This patch suppresses the superfluous messages by checking the currently loaded FW version with the previously loaded one. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst/sst_ipc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 14c2d9d18180..20b01e02ed8f 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -236,7 +236,9 @@ static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, retval = init->result; goto ret; } - dev_info(sst_drv_ctx->dev, "FW Version %02x.%02x.%02x.%02x\n", + if (memcmp(&sst_drv_ctx->fw_version, &init->fw_version, + sizeof(init->fw_version))) + dev_info(sst_drv_ctx->dev, "FW Version %02x.%02x.%02x.%02x\n", init->fw_version.type, init->fw_version.major, init->fw_version.minor, init->fw_version.build); dev_dbg(sst_drv_ctx->dev, "Build date %s Time %s\n", -- 2.11.1

3 7

[alsa-devel] [PATCH] ASoC: intel: remove unused variable data and associated code
by Colin King 27 Mar '17

27 Mar '17

From: Colin Ian King <colin.king(a)canonical.com> The variable 'data' is assigned null and never re-assigned. There is also a redundant check for data being non-null which is always false, so remove this and the variable data and dma_addr as they are not used once the dead code has been removed. Detected with CoverityScan, CID#1324015 ("'Constant' variable gaurds dead code") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index a3459d1682a6..d33bdaf92c57 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2000,10 +2000,8 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw, u32 param_size, char *param) { int ret; - unsigned char *data = NULL; u32 header = 0; u32 payload_size = 0, transfer_parameter_size = 0; - dma_addr_t dma_addr = 0; struct sst_hsw_transfer_parameter *parameter; struct device *dev = hsw->dev; @@ -2047,10 +2045,6 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw, kfree(parameter); - if (data) - dma_free_coherent(hsw->dsp->dma_dev, - param_size, (void *)data, dma_addr); - return ret; } -- 2.11.0

3 2

[alsa-devel] [PATCH 0/3] ASoC: Enable a new IC master mode: bcm2835<=>IC<=>cs42xx8
by Matt Flax 27 Mar '17

27 Mar '17

This patch set lets the ASoC system specify that an IC between the SoC and codec is master. This is intended to put both the SoC and Codec into slave modes. By default un-patched SoC and Codec drivers will return -EINVAL if they aren't enabled and tested for this mode. soc-dia.h has the new SND_SOC_DAIFMT_IBM_IFM definition set as : #define SND_SOC_DAIFMT_IBM_IFM (5 << 12) /* IC clk & FRM master */ The cs42xx8 codec driver is enabled for this mode and so too is the BCM2835 SoC driver. This forms a chain : bcm2835<=>IC<=>cs42xx8 where the IC is bit and frame master. Matt Flax (3): ASoC : Add an IC bit and frame master mode (SoC and Codec slave). ASoC: cs42xx8: allow IC master mode. ASoC: bcm2835: Add mutichannel mode in DSP and IC master modes. include/sound/soc-dai.h | 9 ++++++--- sound/soc/bcm/bcm2835-i2s.c | 21 ++++++++++++++++++++- sound/soc/codecs/cs42xx8.c | 1 + 3 files changed, 27 insertions(+), 4 deletions(-) -- 2.7.4

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[alsa-devel] sound/soc/intel/atom/sst/sst_loader.c:407]: (style) Redundant condition
by David Binderman 27 Mar '17

27 Mar '17

Hello there, linux-4.11-rc4/sound/soc/intel/atom/sst/sst_loader.c:407]: (style) Redundant condition: If 'EXPR == 4', the comparison 'EXPR != 3' is always true. Source code is if (sst_drv_ctx->sst_state != SST_RESET || sst_drv_ctx->sst_state == SST_SHUTDOWN) Regards David Binderman

1 0

[alsa-devel] Internal Speaker problem
by Laszlo Papp 27 Mar '17

27 Mar '17

Dear Alsa Developers, My internal speaker in the Lenovo Thinkpad T510 laptop stopped working. Normally, it would provide either no sound or just for a couple of seconds and then it would go off. I do not have auto-mute enabled. I checked it with alsamixer that it is disabled. My ALSA information is located at http://www.alsa-project.org/db/?f=87547de5ff55e44e360aa2382d4e5d3b7bbed091 Do you know how I could fix this? Ys, L.

1 1

[alsa-devel] [PATCH] ASoC: jack - check status of GPIO-based pins on resume
by Dmitry Torokhov 24 Mar '17

24 Mar '17

From: Dmitry Torokhov <dtor(a)chromium.org> For GPIO-backed pins that are not configured as wakeup sources, we may miss change in their state that happens while system is suspended. Let's use PM notifier to refresh their state upon resume. Signed-off-by: Dmitry Torokhov <dtor(a)chromium.org> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 30 ++++++++++++++++++++++++++++++ 2 files changed, 31 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index b86168a21d56..c0b8bf463940 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -722,6 +722,7 @@ struct snd_soc_jack_gpio { /* private: */ struct snd_soc_jack *jack; struct delayed_work work; + struct notifier_block pm_notifier; struct gpio_desc *desc; void *data; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fbaa1bb41102..d90af94ec1be 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -19,6 +19,7 @@ #include <linux/workqueue.h> #include <linux/delay.h> #include <linux/export.h> +#include <linux/suspend.h> #include <trace/events/asoc.h> /** @@ -293,6 +294,27 @@ static void gpio_work(struct work_struct *work) snd_soc_jack_gpio_detect(gpio); } +static int snd_soc_jack_pm_notifier(struct notifier_block *nb, + unsigned long action, void *data) +{ + struct snd_soc_jack_gpio *gpio = + container_of(nb, struct snd_soc_jack_gpio, pm_notifier); + + switch (action) { + case PM_POST_SUSPEND: + case PM_POST_HIBERNATION: + case PM_POST_RESTORE: + /* + * Use workqueue so we do not have to care about running + * concurrently with work triggered by the interrupt handler. + */ + queue_delayed_work(system_power_efficient_wq, &gpio->work, 0); + break; + } + + return NOTIFY_DONE; +} + /** * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack * @@ -369,6 +391,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, i, ret); } + /* + * Register PM notifier so we do not miss state transitions + * happening while system is asleep. + */ + gpios[i].pm_notifier.notifier_call = snd_soc_jack_pm_notifier; + register_pm_notifier(&gpios[i].pm_notifier); + /* Expose GPIO value over sysfs for diagnostic purposes */ gpiod_export(gpios[i].desc, false); @@ -428,6 +457,7 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { gpiod_unexport(gpios[i].desc); + unregister_pm_notifier(&gpios[i].pm_notifier); free_irq(gpiod_to_irq(gpios[i].desc), &gpios[i]); cancel_delayed_work_sync(&gpios[i].work); gpiod_put(gpios[i].desc); -- 2.12.1.500.gab5fba24ee-goog -- Dmitry

2 5

[alsa-devel] [PATCH] ASoC: jack - check status of GPIO-based pins on resume
by Dmitry Torokhov 24 Mar '17

24 Mar '17

For GPIO-backed pins that are not configured as wakeup sources, we may miss change in their state that happens while system is suspended. Let's use PM notifier to refresh their state upon resume. Signed-off-by: Dmitry Torokhov <dtor(a)chromium.org> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 30 ++++++++++++++++++++++++++++++ 2 files changed, 31 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index b86168a21d56..c0b8bf463940 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -722,6 +722,7 @@ struct snd_soc_jack_gpio { /* private: */ struct snd_soc_jack *jack; struct delayed_work work; + struct notifier_block pm_notifier; struct gpio_desc *desc; void *data; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fbaa1bb41102..d90af94ec1be 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -19,6 +19,7 @@ #include <linux/workqueue.h> #include <linux/delay.h> #include <linux/export.h> +#include <linux/suspend.h> #include <trace/events/asoc.h> /** @@ -293,6 +294,27 @@ static void gpio_work(struct work_struct *work) snd_soc_jack_gpio_detect(gpio); } +static int snd_soc_jack_pm_notifier(struct notifier_block *nb, + unsigned long action, void *data) +{ + struct snd_soc_jack_gpio *gpio = + container_of(nb, struct snd_soc_jack_gpio, pm_notifier); + + switch (action) { + case PM_POST_SUSPEND: + case PM_POST_HIBERNATION: + case PM_POST_RESTORE: + /* + * Use workqueue so we do not have to care about running + * concurrently with work triggered by interrupt handler. + */ + queue_delayed_work(system_power_efficient_wq, &gpio->work, 0); + break; + } + + return NOTIFY_DONE; +} + /** * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack * @@ -369,6 +391,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, i, ret); } + /* + * Register PM notifier so we do not miss state transitions + * happening while system is asleep. + */ + gpios[i].pm_notifier.notifier_call = snd_soc_jack_pm_notifier; + register_pm_notifier(&gpios[i].pm_notifier); + /* Expose GPIO value over sysfs for diagnostic purposes */ gpiod_export(gpios[i].desc, false); @@ -428,6 +457,7 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { gpiod_unexport(gpios[i].desc); + unregister_pm_notifier(&gpios[i].pm_notifier); free_irq(gpiod_to_irq(gpios[i].desc), &gpios[i]); cancel_delayed_work_sync(&gpios[i].work); gpiod_put(gpios[i].desc); -- 2.12.1.578.ge9c3154ca4-goog -- Dmitry

1 0

[alsa-devel] [PATCH][RFC] ASoC: core: remove pointless auxiliary from snd_soc_component
by Kuninori Morimoto 24 Mar '17

24 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 1a653aa44725 ("ASoC: core: replace aux_comp_list to ...") tried to replace aux_comp_list to component_dev_list, but it failed because of binding timing. Thus, Sylwester fixuped it by commit d2e3a1358c37 ("ASoC: Fix binding and probing of auxiliary..."). One of main purpose of commit 1a653aa44725 ("ASoC: core: replace...") was remove replaceable list (= list_aux) from snd_soc_component by using new "auxiliary" flags (but it failed). Because of this background, current code has reborned card_aux_list (= same as original list_aux), and almost pointless "auxiliary" flags. Let's remove pointless "auxiliary" flags by this patch This means, it is same as revert both commit 1a653aa44725 ("ASoC: core: replace aux_comp_list to ...") and commit d2e3a1358c37 ("ASoC: Fix binding and probing of auxiliary..."). Not-yet-Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> request-Tested-by-from: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- >> Sylwester I created this patch, but I can't check. Sylwester, can you please check this patch ? include/sound/soc.h | 1 - sound/soc/soc-core.c | 15 +++++---------- 2 files changed, 5 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dd78507..316fdce 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -815,7 +815,6 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e7d876a..5933851 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1777,7 +1777,6 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1789,14 +1788,13 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp, *tmp; + struct snd_soc_component *comp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, - card_aux_list) { + list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1805,7 +1803,6 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } - list_del(&comp->card_aux_list); } } } @@ -1821,14 +1818,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; + &card->aux_comp_list, card_aux_list) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->card_aux_list); } } } -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: simple-card: fix simple_dai clk lookup
by Lucas Stach 24 Mar '17

24 Mar '17

The clock needs to be stored in the simple_dai structure, so it can be enabled later on. This has been broken during the conversion to use devm_* functions for the clk lookup. Fixes: e984fd61e860 (ASoC: simple-card: use devm_get_clk_from_child()) Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/generic/simple-card-utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4924575d2e95..343b291fc372 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -115,6 +115,7 @@ int asoc_simple_card_parse_clk(struct device *dev, clk = devm_get_clk_from_child(dev, node, NULL); if (!IS_ERR(clk)) { simple_dai->sysclk = clk_get_rate(clk); + simple_dai->clk = clk; } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; } else { -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: blackfin: constify snd_soc_ops structures
by Bhumika Goyal 24 Mar '17

24 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Cross compiled the .o files for blackfin architecture. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/blackfin/bfin-eval-adau1373.c | 2 +- sound/soc/blackfin/bfin-eval-adav80x.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 72ac789..64b88fd 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -119,7 +119,7 @@ static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static struct snd_soc_ops bfin_eval_adau1373_ops = { +static const struct snd_soc_ops bfin_eval_adau1373_ops = { .hw_params = bfin_eval_adau1373_hw_params, }; diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 1037477..99e5eca 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -64,7 +64,7 @@ static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_ops bfin_eval_adav80x_ops = { +static const struct snd_soc_ops bfin_eval_adav80x_ops = { .hw_params = bfin_eval_adav80x_hw_params, }; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rcar: enable PCM RATE untile 192000
by Kuninori Morimoto 24 Mar '17

24 Mar '17

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> R-Car sound can handle untile 192000 rate. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e0b2e9a..5f6e84f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -96,7 +96,7 @@ #include <linux/pm_runtime.h> #include "rsnd.h" -#define RSND_RATES SNDRV_PCM_RATE_8000_96000 +#define RSND_RATES SNDRV_PCM_RATE_8000_192000 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) static const struct of_device_id rsnd_of_match[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rcar: ssi: don't set SSICR.CKDV = 000 with SSIWSR.CONT
by Kuninori Morimoto 24 Mar '17

24 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car Datasheet is indicating "SSICR.CKDV = 000 is invalid when SSIWSR.WS_MODE = 1 or SSIWSR.CONT = 1". Current driver will set CONT, thus, we shouldn't use CKDV = 000. This patch fixup it. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 9594d44..d0602c1 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -245,6 +245,15 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { /* + * It will set SSIWSR.CONT here, but SSICR.CKDV = 000 + * with it is not allowed. (SSIWSR.WS_MODE with + * SSICR.CKDV = 000 is not allowed either). + * Skip it. See SSICR.CKDV + */ + if (j == 0) + continue; + + /* * this driver is assuming that * system word is 32bit x chan * see rsnd_ssi_init() -- 1.9.1

4 5

[alsa-devel] [PATCH v2 0/2] wm8960: Relax bit clock computation
by Daniel Baluta 24 Mar '17

24 Mar '17

This patch series allows playing S20_3LE .wav samples with wm8960 codec. First patch does a small refactoring of sysclk frequency search because wm8960_configure_sysclk was getting pretty convoluted. The second patch allows relaxing bitclock computation in the way that if an exact bitclk couldn't be derived from sysclk it chooses the smalles available bitclk greater than the desired bitclk. Changes since v1: * dropped the RFC tag * comments in each individual patch Daniel Baluta (2): ASoC: codec: wm8960: Refactor sysclk freq search ASoC: codec: wm8960: Relax bit clock computation sound/soc/codecs/wm8960.c | 119 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 97 insertions(+), 22 deletions(-) -- 2.7.4

4 9

[alsa-devel] [PATCH v3 0/2] wm8960: Relax bit clock computation
by Daniel Baluta 24 Mar '17

24 Mar '17

This patch series allows playing S20_3LE .wav samples with wm8960 codec. First patch does a small refactoring of sysclk frequency search because wm8960_configure_sysclk was getting pretty convoluted. The second patch allows relaxing bitclock computation in the way that if an exact bitclk couldn't be derived from sysclk it chooses the smalles available bitclk greater than the desired bitclk. Changes since v2: * do not differentiate between relaxed and exact match Changes since v1: * dropped the RFC tag * comments in each individual patch Daniel Baluta (2): ASoC: codec: wm8960: Refactor sysclk freq search ASoC: codec: wm8960: Relax bit clock computation sound/soc/codecs/wm8960.c | 90 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 71 insertions(+), 19 deletions(-) -- 2.7.4

3 4

[alsa-devel] [PATCH 1/6] regulator: helpers: Add regmap set_soft_start helper
by Charles Keepax 24 Mar '17

24 Mar '17

Add a helper function regulator_set_soft_start_regmap to allow regmap based regulators to easily enable soft start. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/regulator/helpers.c | 18 ++++++++++++++++++ include/linux/regulator/driver.h | 4 ++++ 2 files changed, 22 insertions(+) diff --git a/drivers/regulator/helpers.c b/drivers/regulator/helpers.c index 379cdac..a75e7da 100644 --- a/drivers/regulator/helpers.c +++ b/drivers/regulator/helpers.c @@ -446,6 +446,24 @@ int regulator_set_bypass_regmap(struct regulator_dev *rdev, bool enable) EXPORT_SYMBOL_GPL(regulator_set_bypass_regmap); /** + * regulator_set_soft_start_regmap - Default set_soft_start() using regmap + * + * @rdev: device to operate on. + */ +int regulator_set_soft_start_regmap(struct regulator_dev *rdev) +{ + unsigned int val; + + val = rdev->desc->soft_start_val_on; + if (!val) + val = rdev->desc->soft_start_mask; + + return regmap_update_bits(rdev->regmap, rdev->desc->soft_start_reg, + rdev->desc->soft_start_mask, val); +} +EXPORT_SYMBOL_GPL(regulator_set_soft_start_regmap); + +/** * regulator_get_bypass_regmap - Default get_bypass() using regmap * * @rdev: device to operate on. diff --git a/include/linux/regulator/driver.h b/include/linux/regulator/driver.h index dac8e7b1..7e2e70d 100644 --- a/include/linux/regulator/driver.h +++ b/include/linux/regulator/driver.h @@ -345,6 +345,9 @@ struct regulator_desc { unsigned int active_discharge_off; unsigned int active_discharge_mask; unsigned int active_discharge_reg; + unsigned int soft_start_reg; + unsigned int soft_start_mask; + unsigned int soft_start_val_on; unsigned int enable_time; @@ -476,6 +479,7 @@ int regulator_set_voltage_time_sel(struct regulator_dev *rdev, unsigned int new_selector); int regulator_set_bypass_regmap(struct regulator_dev *rdev, bool enable); int regulator_get_bypass_regmap(struct regulator_dev *rdev, bool *enable); +int regulator_set_soft_start_regmap(struct regulator_dev *rdev); int regulator_set_active_discharge_regmap(struct regulator_dev *rdev, bool enable); -- 2.1.4

4 10

[alsa-devel] [PATCH v2] ASoC: imx-wm8960: Let codec driver enable/disable its MCLK
by Daniel Baluta 24 Mar '17

24 Mar '17

From: Daniel Baluta <daniel.baluta(a)nxp.com> WM8962 needs its MCLK when powerup in wm8962_resume(). Thus it's better to control the MCLK in codec driver. Thus remove the clock enable in machine driver accordingly. While at it, get rid of imx_wm8962_remove function since it is now empty. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: * s/wm8960/imx-wm890/ in subject prefix * s/dirver/driver in commit message * took ownership over the patch from Nicolin Chen as per his agreement. sound/soc/fsl/imx-wm8962.c | 40 ++++++++-------------------------------- 1 file changed, 8 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 1b60958..3d894d9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -33,7 +33,6 @@ struct imx_wm8962_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - struct clk *codec_clk; unsigned int clk_frequency; }; @@ -163,6 +162,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_wm8962_data *data; + struct clk *codec_clk; int int_port, ext_port; int ret; @@ -231,19 +231,14 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - ret = PTR_ERR(data->codec_clk); + codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(codec_clk)) { + ret = PTR_ERR(codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } - data->clk_frequency = clk_get_rate(data->codec_clk); - ret = clk_prepare_enable(data->codec_clk); - if (ret) { - dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); - goto fail; - } + data->clk_frequency = clk_get_rate(codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -258,10 +253,10 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -277,16 +272,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } - of_node_put(ssi_np); - of_node_put(codec_np); - - return 0; - -clk_fail: - clk_disable_unprepare(data->codec_clk); fail: of_node_put(ssi_np); of_node_put(codec_np); @@ -294,17 +282,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) return ret; } -static int imx_wm8962_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - - if (!IS_ERR(data->codec_clk)) - clk_disable_unprepare(data->codec_clk); - - return 0; -} - static const struct of_device_id imx_wm8962_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8962", }, { /* sentinel */ } @@ -318,7 +295,6 @@ static struct platform_driver imx_wm8962_driver = { .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, - .remove = imx_wm8962_remove, }; module_platform_driver(imx_wm8962_driver); -- 2.7.4

2 1

[alsa-devel] [GIT PULL] Broxton audio fw update
by Vinod Koul 24 Mar '17

24 Mar '17

Hi, Please pull to receive updated audio firmware for broxton platform. The following changes since commit e39f0e3e6897ad865b3704f61218ae83f98a85da: Merge git://git.marvell.com/mwifiex-firmware (2017-03-14 08:58:33 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git apl for you to fetch changes up to a4dde74b504522d239e8521617b53b03f393ed6a: linux-firmware: intel: Update Broxton audio firmware (2017-03-22 21:53:18 +0530) ---------------------------------------------------------------- Subhransu S. Prusty (1): linux-firmware: intel: Update Broxton audio firmware WHENCE | 4 +++- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v2219.bin | Bin 0 -> 505608 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v2219.bin -- ~Vinod

2 1

[alsa-devel] sound: use-after-free in snd_seq_cell_alloc
by Dmitry Vyukov 24 Mar '17

24 Mar '17

Hello, I've got the following report while running syzkaller fuzzer on 093b995e3b55a0ae0670226ddfcb05bfbf0099ae. Unfortunately this is not reproducible so far. ================================================================== BUG: KASAN: use-after-free in __lock_acquire+0x39df/0x3a80 kernel/locking/lockdep.c:3230 at addr ffff8800495fb6a0 Read of size 8 by task syz-executor7/17253 CPU: 2 PID: 17253 Comm: syz-executor7 Not tainted 4.11.0-rc3+ #364 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: __asan_report_load8_noabort+0x29/0x30 mm/kasan/report.c:337 __lock_acquire+0x39df/0x3a80 kernel/locking/lockdep.c:3230 lock_acquire+0x1ee/0x590 kernel/locking/lockdep.c:3762 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 snd_seq_cell_alloc.isra.1+0x17d/0x6a0 sound/core/seq/seq_memory.c:238 snd_seq_event_dup+0x15d/0xa00 sound/core/seq/seq_memory.c:309 snd_seq_fifo_event_in+0xe3/0x420 sound/core/seq/seq_fifo.c:128 snd_seq_deliver_single_event.constprop.12+0x814/0x980 sound/core/seq/seq_clientmgr.c:615 snd_seq_deliver_event+0x12c/0x720 sound/core/seq/seq_clientmgr.c:818 snd_seq_kernel_client_dispatch+0x131/0x160 sound/core/seq/seq_clientmgr.c:2319 snd_seq_system_notify+0x11b/0x160 sound/core/seq/seq_system.c:112 snd_seq_client_notify_subscription+0x16f/0x210 sound/core/seq/seq_clientmgr.c:1414 subscribe_port sound/core/seq/seq_ports.c:434 [inline] check_and_subscribe_port+0x725/0x950 sound/core/seq/seq_ports.c:510 snd_seq_port_connect+0x3e5/0x700 sound/core/seq/seq_ports.c:579 snd_seq_ioctl_subscribe_port+0x24b/0x300 sound/core/seq/seq_clientmgr.c:1443 snd_seq_kernel_client_ctl+0x122/0x160 sound/core/seq/seq_clientmgr.c:2350 snd_seq_oss_midi_open+0x640/0x850 sound/core/seq/oss/seq_oss_midi.c:375 snd_seq_oss_synth_setup_midi+0x109/0x4d0 sound/core/seq/oss/seq_oss_synth.c:281 snd_seq_oss_open+0x70f/0x8b0 sound/core/seq/oss/seq_oss_init.c:274 odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 soundcore_open+0x321/0x630 sound/sound_core.c:639 chrdev_open+0x257/0x730 fs/char_dev.c:392 do_dentry_open+0x710/0xc80 fs/open.c:751 vfs_open+0x105/0x220 fs/open.c:864 do_last fs/namei.c:3349 [inline] path_openat+0x1151/0x35b0 fs/namei.c:3490 do_filp_open+0x249/0x370 fs/namei.c:3525 do_sys_open+0x502/0x6d0 fs/open.c:1051 SYSC_openat fs/open.c:1078 [inline] SyS_openat+0x30/0x40 fs/open.c:1072 entry_SYSCALL_64_fastpath+0x1f/0xc2 RIP: 0033:0x445b79 RSP: 002b:00007f02452a6858 EFLAGS: 00000292 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 0000000000708000 RCX: 0000000000445b79 RDX: 0000000000000000 RSI: 0000000020063ff0 RDI: ffffffffffffff9c RBP: 0000000000000086 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000292 R12: 00000000006e0fe0 R13: 0000000000001000 R14: 0000000000000003 R15: 0000000020000000 Object at ffff8800495fb600, in cache kmalloc-192 size: 192 Allocated: PID = 17260 kmalloc include/linux/slab.h:490 [inline] kzalloc include/linux/slab.h:663 [inline] snd_seq_pool_new+0x93/0x2f0 sound/core/seq/seq_memory.c:470 snd_seq_fifo_new+0xc3/0x360 sound/core/seq/seq_fifo.c:41 snd_seq_open+0x404/0x520 sound/core/seq/seq_clientmgr.c:333 snd_open+0x1fa/0x3f0 sound/core/sound.c:177 chrdev_open+0x257/0x730 fs/char_dev.c:392 do_dentry_open+0x710/0xc80 fs/open.c:751 vfs_open+0x105/0x220 fs/open.c:864 do_last fs/namei.c:3349 [inline] path_openat+0x1151/0x35b0 fs/namei.c:3490 do_filp_open+0x249/0x370 fs/namei.c:3525 do_sys_open+0x502/0x6d0 fs/open.c:1051 SYSC_open fs/open.c:1069 [inline] SyS_open+0x2d/0x40 fs/open.c:1064 entry_SYSCALL_64_fastpath+0x1f/0xc2 Freed: PID = 17280 kfree+0xd7/0x250 mm/slab.c:3831 snd_seq_pool_delete+0x52/0x70 sound/core/seq/seq_memory.c:498 snd_seq_fifo_resize+0x263/0x3e0 sound/core/seq/seq_fifo.c:275 snd_seq_ioctl_set_client_pool+0x481/0x600 sound/core/seq/seq_clientmgr.c:1849 snd_seq_ioctl+0x204/0x470 sound/core/seq/seq_clientmgr.c:2131 vfs_ioctl fs/ioctl.c:45 [inline] do_vfs_ioctl+0x1af/0x16d0 fs/ioctl.c:685 SYSC_ioctl fs/ioctl.c:700 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:691 entry_SYSCALL_64_fastpath+0x1f/0xc2 Memory state around the buggy address: ffff8800495fb580: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8800495fb600: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >ffff8800495fb680: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ^ ffff8800495fb700: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8800495fb780: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ==================================================================

2 1

[alsa-devel] Regressions on x86(_64) and >128 mixer controls
by Mark Hills 24 Mar '17

24 Mar '17

I have an echo3g (Layla) PCI soundcard on x86_64. alsamixer and alsactl fail with the following: alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. In question is idx=128, and this "value" array in alsa-lib: struct snd_ctl_elem_value { struct snd_ctl_elem_id id; /* W: element ID */ unsigned int indirect: 1; /* W: indirect access - obsoleted */ union { union { long value[128]; long *value_ptr; /* obsoleted */ } integer; At first it appears just to be hitting the fixed limit of 128. However, this used to work -- and at the time I was using 32-bit kernel and userland on an otherwise similar system (not they are both 64-bit) Increasing the array size cascades to a new problem which suggests these controls don't actually exist: ./alsactl/alsactl: get_control:258: Cannot read control '2,0,0,PCM Playback Volume,0': Inappropriate ioctl for device So I investigated the origin of this 256, which is here in linux/sound/pci/echoaudio/echoaudio.c: static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct echoaudio *chip; chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); /* = 16 */ uinfo->dimen.d[1] = num_busses_in(chip); /* = 16 */ uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; /* = 256 */ return 0; } These values are derived from constants in the source code for this Layla3G device; I can find nothing on this path that would cause variation between a 32-bit and 64-bit kernel -- the code suggests like this value has always been 256 in the kernel, so I can't come up with a reason that it could vary. I don't have enough knowledge of the hardware to know if 256 /is/ the correct value here. Something which may be related happened for a period when I was using 32-bit userland with 64-bit kernel. This broke a number of tools (incl. echomixer) Compare alsa-lib/include/alsa/asound.h: struct snd_ctl_elem_info { struct snd_ctl_elem_id id; /* W: element ID */ snd_ctl_elem_type_t type; /* R: value type - SNDRV_CTL_ELEM_TYPE_* */ unsigned int access; /* R: value access (bitmask) - SNDRV_CTL_ELEM_ACCESS_* */ unsigned int count; /* count of values */ __kernel_pid_t owner; /* owner's PID of this control */ union { struct { long min; /* R: minimum value */ long max; /* R: maximum value */ long step; /* R: step (0 variable) */ } integer; and notice the use of 'long' in this struct. It has a shadow/equivalent/identical in linux/uapi/sound/asound.h, copied to userland, using: if (ioctl(hw->fd, SNDRV_CTL_IOCTL_ELEM_INFO, info) < 0) return -errno; The kernel and userland can have different interpretations of 'long'; do these really not need to be the 'int' type? Though maybe I am missing something -- surely I am not the only person to have used a 32-bit userland on a 64-bit kernel? -- Mark

1 0

[alsa-devel] [PATCH] ALSA: au88x0: avoid theoretical uninitialized access
by Arnd Bergmann 24 Mar '17

24 Mar '17

The latest gcc-7.0.1 snapshot points out that we if nr_ch is zero, we never initialize some variables: sound/pci/au88x0/au88x0_core.c: In function 'vortex_adb_allocroute': sound/pci/au88x0/au88x0_core.c:2304:68: error: 'mix[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/pci/au88x0/au88x0_core.c:2305:58: error: 'src[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized] I assume this can never happen in practice, but adding a check here doesn't hurt either and avoids the warning. The code has been unchanged since the start of git history. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/au88x0/au88x0_core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index e1af24f87566..c308a4f70550 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2279,6 +2279,9 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, } else { int src[2], mix[2]; + if (nr_ch < 1) + return -EINVAL; + /* Get SRC and MIXER hardware resources. */ for (i = 0; i < nr_ch; i++) { if ((mix[i] = -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: core: remove pointless auxiliary from snd_soc_component
by Kuninori Morimoto 24 Mar '17

24 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 1a653aa44725 ("ASoC: core: replace aux_comp_list to ...") tried to replace aux_comp_list to component_dev_list, but it failed because of binding timing. Thus, Sylwester fixuped it by commit d2e3a1358c37 ("ASoC: Fix binding and probing of auxiliary..."). One of main purpose of commit 1a653aa44725 ("ASoC: core: replace...") was remove replaceable list (= list_aux) from snd_soc_component by using new "auxiliary" flags (but it failed). Because of this background, current code has reborned card_aux_list (= same as original list_aux), and almost pointless "auxiliary" flags. Let's remove pointless "auxiliary" flags by this patch This means, it is same as revert both commit 1a653aa44725 ("ASoC: core: replace aux_comp_list to ...") and commit d2e3a1358c37 ("ASoC: Fix binding and probing of auxiliary..."). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 15 +++++---------- 2 files changed, 5 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dd78507..316fdce 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -815,7 +815,6 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e7d876a..5933851 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1777,7 +1777,6 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1789,14 +1788,13 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp, *tmp; + struct snd_soc_component *comp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, - card_aux_list) { + list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1805,7 +1803,6 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } - list_del(&comp->card_aux_list); } } } @@ -1821,14 +1818,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; + &card->aux_comp_list, card_aux_list) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->card_aux_list); } } } -- 1.9.1 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH] ALSA: hda - Adding a group of pin definition to fix headset problem
by Hui Wang 23 Mar '17

23 Mar '17

A new Dell laptop needs to apply ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to fix the headset problem, and the pin definiton of this machine is not in the pin quirk table yet, now adding it to the table. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1561076..efa657f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6191,6 +6191,8 @@ enum { ALC295_STANDARD_PINS, {0x17, 0x21014040}, {0x18, 0x21a19050}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC295_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x17, 0x90170110}), -- 1.9.1

3 2

[alsa-devel] [PATCH v3] ASoC: rt286: fix headphone click/crack noise on Dell XPS 9343 I2S mode
by Kai-Heng Feng 23 Mar '17

23 Mar '17

HDA mode fixed the issue by these two commits: '9476d369d7b3 ALSA: hda - Mute headphone pin on suspend on XPS13 9333' '3e1b0c4a9d56 ALSA: hda - Fix click noise at start on Dell XPS13' Apply the same workarounds to rt286 can solve the issue. When jack is plugged, it rapidly generates I2C interrupts, which triggers rt286_irq() and rt286_jack_detect(), which produces the click noise. alc_fixup_dell_xps13() in patch_realtek.c sets up a pin that can stop the frantic interrupts, hence avoids the click noise. When rt286 is under powersaving state, play a sound with headphone or plug a headphone in will produce a loud crack sound. Set AMP_OUT_MUTE before power events can make the noise less noticeable. Unmute the AMP when the power is up. v3: Implicit conversion instead of tenary operator. v2: Use 'HP Power' instead of individual power events. Link: https://bugzilla.kernel.org/show_bug.cgi?id=112611 Link: https://bugzilla.redhat.com/show_bug.cgi?id=1313434 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/soc/codecs/rt286.c | 48 +++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 45 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..97b52697f974 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,6 +36,9 @@ #define RT286_VENDOR_ID 0x10ec0286 #define RT288_VENDOR_ID 0x10ec0288 +#define AMP_OUT_MUTE 0xb080 +#define AMP_OUT_UNMUTE 0xb000 + struct rt286_priv { struct reg_default *index_cache; int index_cache_size; @@ -47,6 +50,7 @@ struct rt286_priv { struct delayed_work jack_detect_work; int sys_clk; int clk_id; + bool is_dell_dino; }; static const struct reg_default rt286_index_def[] = { @@ -472,6 +476,32 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } +/* Power event function to workaround headphone crack noise */ +static int rt286_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!rt286->is_dell_dino) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_MUTE); + break; + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_UNMUTE); + break; + default: + return 0; + } + + return 0; +} + static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -578,7 +608,9 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, - RT286_SET_PIN_SFT, 0, NULL, 0), + RT286_SET_PIN_SFT, 0, rt286_hp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output Mixer */ SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, @@ -1175,8 +1207,10 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + rt286->is_dell_dino = dmi_check_system(dmi_dell_dino); + if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + rt286->is_dell_dino) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1192,6 +1226,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { + /* Fix headphone click noise */ + if (rt286->is_dell_dino) + regmap_write(rt286->regmap, + RT286_MIC1_DET_CTRL, 0x0020); + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0x5000); } @@ -1215,7 +1254,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (rt286->is_dell_dino) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, @@ -1224,6 +1263,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, RT286_SET_GPIO_DATA, 0x40, 0x40); regmap_update_bits(rt286->regmap, RT286_GPIO_CTRL, 0xc, 0x8); + /* Workaound headphone crack noise when probing */ + regmap_write(rt286->regmap, RT286_SET_AMP_GAIN_HPO, + AMP_OUT_MUTE); } if (rt286->i2c->irq) { -- 2.12.0

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[alsa-devel] [PATCH 0/4] ASoC: sun8i-codec: Clean-up
by Mylène Josserand 21 Mar '17

21 Mar '17

Hi all, Thanks to Chen-Yu's review [1], here are some patches to clean-up my sun8i A33 audio driver. Patches tested on asoc/for-next branch with additionnal patches to apply before: https://patchwork.kernel.org/patch/9447631/ https://patchwork.kernel.org/patch/9423999/ https://patchwork.kernel.org/patch/9521121/ https://www.spinics.net/lists/arm-kernel/msg560091.html Patch 1: Removes the HP widget which is available in the analog part. Patch 2: Update the driver to use SOC_DAPM_DOUBLE and rename the controls accordingly. Patch 3: Convert SND_SOC_DAPM_MIXER to SOC_MIXER_ARRAY. Patch 3: Convert SND_SOC_DAPM_DAC to SND_SOC_DAPM_AIF_IN definition. The last Chen Yu's comments are still in progress (ie use of set_pll and set_sysclk and reworking the audio route). After this series, amixer commands to test an audio file: amixer set 'Headphone' 75% amixer set 'Headphone' on amixer set 'DAC' on amixer set 'AIF1 Slot 0 Digital DAC' on [1] https://lkml.org/lkml/2017/2/7/103 Thank you in advance, Best regards, Mylène Josserand (4): ASoC: sun8i-codec: Remove analog "HP" widget ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE ASoC: sun8i-codec: Convert to SOC_MIXER_ARRAY ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN arch/arm/boot/dts/sun8i-a33.dtsi | 4 +-- sound/soc/sunxi/sun8i-codec.c | 65 +++++++++++++++++----------------------- 2 files changed, 30 insertions(+), 39 deletions(-) -- 2.11.0

3 19

[alsa-devel] [PATCH v2 0/3] ASoC: sun8i-codec: Clean-up
by Mylène Josserand 21 Mar '17

21 Mar '17

Hello everyone, Thanks to Chen-Yu's review [1], a first version of this patchset to clean-up my sun8i A33 audio driver has been sent and merged. One patch [2] needs a rework so this cover-letter is for its V2. Changes with v1: - Remove patches 01, 02 and 03 as they are merged in asoc/for-next. - Added a patch to fix an error on an widget name - Split the patch 04 into two patches to separate the device tree modifications with the codec driver's one. Patches tested on asoc/for-next branch, last commit: 698b3eb9af. Patch 1: Fix a dapm routing error because of an unwanted space. Patch 2: Convert SND_SOC_DAPM_DAC to SND_SOC_DAPM_AIF_IN definition. The AIF interfaces are renamed to be more precise. Patch 3: Update the audio-routing's device tree to use the AIF interfaces renamed in patch 01. After this series, amixer commands to test an audio file: amixer set 'Headphone' 75% amixer set 'Headphone' on amixer set 'DAC' on amixer set 'AIF1 Slot 0 Digital DAC' on [1] https://lkml.org/lkml/2017/2/7/103 [2] https://patchwork.kernel.org/patch/9566325/ Thank you in advance, Best regards, Mylène Josserand (3): ASoC: sun8i-codec: Fix space on audio-routing widget ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN ARM: dts: sun8i: Update audio-routing with renamed widgets arch/arm/boot/dts/sun8i-a33.dtsi | 4 ++-- sound/soc/sunxi/sun8i-codec.c | 22 ++++++++++++---------- 2 files changed, 14 insertions(+), 12 deletions(-) -- 2.11.0

2 7

[alsa-devel] [PATCH v3 0/5] ASoC: sun8i-a33 clean-up for 4.11
by Mylène Josserand 21 Mar '17

21 Mar '17

Hi everyone, As discussed in this thread [1], here are a V3 of the cleanup series for the sun8i-a33 driver (introduced in v4.11-rc1) as fixes for v4.11. Based on: Linus repository - master branch Last commit: 8841b5f0cd ("Merge tag 'nfs-for-4.11-2' of git://git.linux-nfs.org/projects/anna/linux-nfs") Changes since v1 and v2: - Patch 01: No changes from v1, it applies directly - Patch 02: No changes from v1, it applies directly once patch 01 has been applied. - Patch 03: No changes from v2, it applies directly - Patch 04: Rebased on master branch - Patch 05: No changes from v2, it applies directly Patch 01 removes some unnecessary widgets. Patch 02 updates the driver to use SOC_DAPM_DOUBLE. Patch 03 removes a space on a widget name that implies an error on audio routing as the widget is unknown. Patch 04 improves some widgets (names and types). As widget names changed in patch 04, the patch 05 is needed to use the same names on audio routing. Otherwise, the device tree for sun8i-a33 will be broken and the audio codec will not work. Only patches 01 and 02 were initially wanted as fixes for v4.11 but, with Chen-Yu, we thought that patch 03 and 04 could be also applied as fixes. As patch 04 creates a break with the device tree, I added patch 05 in this series. Let me know what you think. [1]: https://lkml.org/lkml/2017/3/14/4 Thank you in advance, Best regards, Mylène Josserand (5): ASoC: sun8i-codec: Remove analog "HP" widget ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE ASoC: sun8i-codec: Fix space on audio-routing widget ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN ARM: dts: sun8i: Update audio-routing with renamed widgets arch/arm/boot/dts/sun8i-a33.dtsi | 4 +-- sound/soc/sunxi/sun8i-codec.c | 67 ++++++++++++++++++---------------------- 2 files changed, 32 insertions(+), 39 deletions(-) -- 2.11.0

3 17

[alsa-devel] [PATCH v2] ASoC: Intel: Update bxt_da7219_max98357a to add a new
by Harsha Priya 21 Mar '17

21 Mar '17

This patch adds a platform clock widget to turn off the clock only when both headset capture and headset playback are not in use. This removes turning off the clock in hw_free so that the clock is on when either capture or playback of headset is in progress. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 97 +++++++++++++++------------ 1 file changed, 53 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 2cda06c..3a8c4d9 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -55,6 +55,54 @@ enum { BXT_DPCM_AUDIO_HDMI3_PB, }; +static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI, + strlen(BXT_DIALOG_CODEC_DAI))) + return rtd->codec_dai; + } + + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = bxt_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if(SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, + DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); + if (ret) + dev_err(card->dev, "can't set codec sysclk configuration\n"); + + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -69,6 +117,8 @@ enum { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; static const struct snd_soc_dapm_route broxton_map[] = { @@ -109,6 +159,9 @@ enum { /* DMIC */ {"dmic01_hifi", NULL, "DMIC01 Rx"}, {"DMIC01 Rx", NULL, "DMIC AIF"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, }; static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -243,49 +296,6 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) .startup = bxt_fe_startup, }; -static int broxton_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 *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(codec_dai->dev, "can't set codec sysclk configuration\n"); - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static int broxton_da7219_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static const struct snd_soc_ops broxton_da7219_ops = { - .hw_params = broxton_da7219_hw_params, - .hw_free = broxton_da7219_hw_free, -}; - static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -467,7 +477,6 @@ static int broxton_refcap_startup(struct snd_pcm_substream *substream) SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp_fixup, - .ops = &broxton_da7219_ops, .dpcm_playback = 1, .dpcm_capture = 1, }, -- 1.9.1

3 2

[alsa-devel] [PATCH 1/4] dt-bindings: Document the dmas and dma-names properties for bcm2835-analog-audio
by Michael Zoran 21 Mar '17

21 Mar '17

Add documentation for the DMA properties required when using the PWM controller of the bcm2835 for analog audio output. Signed-off-by: Michael Zoran <mzoran(a)crowfest.net> --- Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt b/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt index cf573e85b11d..08fc06fc083a 100644 --- a/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt +++ b/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt @@ -1,14 +1,21 @@ BCM2835 PWM controller (Raspberry Pi controller) Required properties: -- compatible: should be "brcm,bcm2835-pwm" +- compatible: should be "brcm,bcm2835-pwm" or "brcm,bcm2835-analog-audio" - reg: physical base address and length of the controller's registers - clocks: This clock defines the base clock frequency of the PWM hardware system, the period and the duty_cycle of the PWM signal is a multiple of the base period. + +Required properties(PWM Output): - #pwm-cells: Should be 2. See pwm.txt in this directory for a description of the cells format. +Required properties(Audio Output): +- dmas: Should contain one entry pointing to the DMA channel used to + transfer audio data +- dma-names: Should contain "tx" + Examples: pwm@2020c000 { -- 2.11.0

3 9

[alsa-devel] hw/sw params API in alsa-lib miscompiled, result: unusable alsa-utils
by Deweloper 21 Mar '17

21 Mar '17

Hi, I built alsa-lib and alsa-utils (v.1.1.3) from sources, using GCC 6. When I tried launching arecord, it failed due to lack of memory. During analysis I found out that: - the message is printed because realloc(buf, period_size) returns NULL - realloc() returns NULL because period_size is 0 - period_size is 0 because snd_pcm_hw_params_get_period_size() puts 0 to *frames (instead of 32768) and returns 32768 (instead of 0) - snd_pcm_hw_params_get_period_size() misbehaves in this way because although a NEW API header is used, the OLD API implementation is actually called So it seems that I've run into exactly the same problem which was described by John Spencer in August 2013: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065031.html > my problem is that the old version is getting called instead of the > new one Did anything change in this matter since August, 2013? --

2 2

[alsa-devel] [PATCH v5 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 21 Mar '17

21 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..c4375ec 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -50,7 +50,7 @@ typedef struct { snd_pcm_format_t sformat; int schannels; int srate; - const snd_config_t *rate_converter; + snd_config_t *rate_converter; enum snd_pcm_plug_route_policy route_policy; snd_pcm_route_ttable_entry_t *ttable; int ttable_ok; @@ -64,6 +64,10 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) { + snd_config_delete(plug->rate_converter); + plug->rate_converter = NULL; + } assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1112,6 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1125,6 +1128,15 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, pcm->ops = &snd_pcm_plug_ops; pcm->fast_ops = slave->fast_ops; pcm->fast_op_arg = slave->fast_op_arg; + if (rate_converter) { + err = snd_config_copy(&plug->rate_converter, + (snd_config_t *)rate_converter); + if (err < 0) { + snd_pcm_free(pcm); + free(plug); + return err; + } + } pcm->private_data = plug; pcm->poll_fd = slave->poll_fd; pcm->poll_events = slave->poll_events; -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ALSA: x86: Select CONFIG_HDMI_LPE_AUDIO as default
by Ian W MORRISON 21 Mar '17

21 Mar '17

The upstream kernel builds for distributions such as Ubuntu which now includes binary packages for v4.11 mainline kernel release candidates are promoted as a way of testing upstream kernels to to confirm that upstream has fixed a specific issue (see https://wiki.ubuntu.com/ Kernel/MainlineBuilds). Unfortunately the long awaited patch for providing HDMI audio support for Bay Trail and Cherry Trail devices does not include this support through a module built by default. Through including by default of the two associated CONFIG settings (SND_X86 and HDMI_LPE_AUDIO), upstream kernel builds would automatically provide the much desired HDMI audio support by default. This patch uses a Kconfig 'default' statement to include the driver as default. Changes in version 2: CONFIG_SND_X86 now a bool and changed default m to default y Signed-off-by: Ian W Morrison <linuxium(a)linuxium.com.au> --- sound/x86/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc..cac2270 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 - tristate "X86 sound devices" + bool "X86 sound devices" depends on X86 + default y ---help--- X86 sound devices that don't fall under SoC or PCI categories @@ -9,6 +10,7 @@ if SND_X86 config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" depends on DRM_I915 + default y select SND_PCM help Choose this option to support HDMI LPE Audio mode -- 1.9.1

2 10

[alsa-devel] [PATCH] ASoC: STI: Fix null ptr deference in IRQ handler
by Arnaud Pouliquen 21 Mar '17

21 Mar '17

With RTlinux a race condition has been found that leads to NULL ptr crash: - On CPU 0: uni_player_irq_handler is called to treat XRUN "(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked, dev_err(player->dev, "FIFO underflow error detected") is printed and then snd_pcm_stream_lock should be called to lock stream for stopping. - On CPU 1: application stop and close the stream. Issue is that the stop and shutdown functions are executed while "FIFO underflow error detected" is printed. So when CPU 0 call snd_pcm_stream_lock is player->substream is already null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 9 +++------ sound/soc/sti/uniperif_reader.c | 8 +++++--- 2 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 60ae31a..6fa9eee 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -65,8 +65,10 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) unsigned int status; unsigned int tmp; + snd_pcm_stream_lock(player->substream); if (player->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ + snd_pcm_stream_unlock(player->substream); return IRQ_NONE; } @@ -88,9 +90,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); } ret = IRQ_HANDLED; @@ -104,9 +104,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } @@ -138,13 +136,12 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } + snd_pcm_stream_unlock(player->substream); return ret; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 5992c6a..9273f59 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -46,9 +46,11 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) struct uniperif *reader = dev_id; unsigned int status; + snd_pcm_stream_lock(reader->substream); if (reader->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ dev_warn(reader->dev, "unexpected IRQ\n"); + snd_pcm_stream_unlock(reader->substream); return IRQ_HANDLED; } @@ -60,13 +62,13 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stream_lock(reader->substream); snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(reader->substream); - return IRQ_HANDLED; + ret = IRQ_HANDLED; } + snd_pcm_stream_unlock(reader->substream); + return ret; } -- 1.9.1

2 6

[alsa-devel] sound: another deadlock in snd_seq_pool_done
by Dmitry Vyukov 21 Mar '17

21 Mar '17

Hello, The following program creates processes deadlocked in snd_seq_pool_done: https://gist.githubusercontent.com/dvyukov/97efc9cb6d63b1b2c7b737b82cc8b0d6… After few seconds I get: # ps afxu | grep a.out root 8660 2.0 0.0 0 0 pts/0 Zl 16:27 0:00 [a.out] <defunct> # kill -9 8660 # cat /proc/8660/status Name: a.out State: Z (zombie) Tgid: 8660 Ngid: 0 Pid: 8660 PPid: 1 TracerPid: 0 Uid: 0 0 0 0 Gid: 0 0 0 0 FDSize: 0 Groups: 0 NStgid: 8660 NSpid: 8660 NSpgid: 8660 NSsid: 2971 Threads: 2 SigQ: 1/3304 SigPnd: 0000000000000000 ShdPnd: 0000000000000100 SigBlk: 0000000000000000 SigIgn: 0000000180000000 SigCgt: 0000000000000440 CapInh: 0000000000000000 CapPrm: 0000003fffffffff CapEff: 0000003fffffffff CapBnd: 0000003fffffffff CapAmb: 0000000000000000 NoNewPrivs: 0 Seccomp: 0 Cpus_allowed: f Cpus_allowed_list: 0-3 Mems_allowed: 00000000,00000001 Mems_allowed_list: 0 voluntary_ctxt_switches: 12 nonvoluntary_ctxt_switches: 0 # cat /proc/8660/task/*/stack [<ffffffff835406db>] snd_seq_pool_done+0x31b/0x620 sound/core/seq/seq_memory.c:436 [<ffffffff8353a11e>] snd_seq_ioctl_set_client_pool+0x1ae/0x600 sound/core/seq/seq_clientmgr.c:1836 [<ffffffff835382ba>] snd_seq_ioctl+0x2da/0x4d0 sound/core/seq/seq_clientmgr.c:2130 [<ffffffff81aced2f>] vfs_ioctl fs/ioctl.c:45 [inline] [<ffffffff81aced2f>] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:685 [<ffffffff81ad038f>] SYSC_ioctl fs/ioctl.c:700 [inline] [<ffffffff81ad038f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:691 [<ffffffff8457dc41>] entry_SYSCALL_64_fastpath+0x1f/0xc2

2 6

[alsa-devel] [PATCH] ALSA: x86: Make CONFIG_SND_X86 bool
by Takashi Iwai 21 Mar '17

21 Mar '17

CONFIG_SND_X86 is a menu config to filter only for x86-specific drivers in its sub-menu, and this doesn't have to be tristate but rather it should be a bool. Also, like other sub-menu configs, it's more user-friendly to be default=y; it's merely a menu config and the actual drivers are configured in the sub-menu, after all. Fixes: 287599cf2d77 ("ALSA: add Intel HDMI LPE audio driver for BYT/CHT-T") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc597c..8adf4d1bd46e 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 - tristate "X86 sound devices" + bool "X86 sound devices" depends on X86 + default y ---help--- X86 sound devices that don't fall under SoC or PCI categories -- 2.11.1

1 0

[alsa-devel] [PATCH v4 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 21 Mar '17

21 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..f1ae814 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -64,6 +64,10 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) { + snd_config_delete((snd_config_t*)plug->rate_converter); + plug->rate_converter = NULL; + } assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1112,6 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1125,6 +1128,14 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, pcm->ops = &snd_pcm_plug_ops; pcm->fast_ops = slave->fast_ops; pcm->fast_op_arg = slave->fast_op_arg; + if (rate_converter) { + if ((err = snd_config_copy((snd_config_t**)&plug->rate_converter, + (snd_config_t *)rate_converter)) < 0) { + snd_pcm_free(pcm); + free(plug); + return err; + } + } pcm->private_data = plug; pcm->poll_fd = slave->poll_fd; pcm->poll_events = slave->poll_events; -- 2.7.4

2 1

Re: [alsa-devel] [RESEND PATCH v2 1/2] ASoC: stm32: add bindings for SAI
by Olivier MOYSAN 21 Mar '17

21 Mar '17

Hello Rob, On 03/18/2017 09:43 PM, Rob Herring wrote: > On Thu, Mar 16, 2017 at 5:50 AM, Olivier MOYSAN <olivier.moysan(a)st.com> wrote: >> Hello Rob, >> >> Thanks for your feedback. >> >> On 03/15/2017 06:46 PM, Rob Herring wrote: >>> On Tue, Mar 07, 2017 at 10:44:31AM +0100, olivier moysan wrote: >>>> This patch adds documentation of device tree bindings for the >>>> STM32 SAI ASoC driver. >>>> >>>> Signed-off-by: olivier moysan <olivier.moysan(a)st.com> >>>> --- >>>> .../devicetree/bindings/sound/st,stm32-sai.txt | 79 ++++++++++++++++++++++ >>>> 1 file changed, 79 insertions(+) >>>> create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt >>>> >>>> diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt >>>> new file mode 100644 >>>> index 0000000..b3be26d >>>> --- /dev/null >>>> +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt >>>> @@ -0,0 +1,79 @@ >>>> +STMicroelectronics STM32 Serial Audio Interface (SAI). >>>> + >>>> +The SAI interface (Serial Audio Interface) offers a wide set of audio protocols >>>> +as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. >>>> +The SAI contains two independent audio sub-blocks. Each sub-block has >>>> +its own clock generator and I/O lines controller. >>>> + >>>> +Required properties: >>>> + - compatible: Should be "st,stm32f4-sai" >>>> + - reg: Base address and size of SAI common register set. >>>> + - clocks: Must contain phandle and clock specifier pairs for each entry >>>> + in clock-names. >>>> + - clock-names: Must contain "clk_x8k" and "clk_x11k" >>>> + "clk_x8k": SAI parent clock for sampling rates multiple of 8kHz. >>>> + "clk_x11k": SAI parent clock for sampling rates multiple of 11.025kHz. >>> >>> clk_ is redundant. >>> >> >> Do you mean, removing clk_ prefix ? > > Yes. > >> >>>> + - interrupts: cpu DAI interrupt line shared by SAI sub-blocks >>>> + >>>> +Note: Each SAI controller should have an alias correctly numbered >>>> +in "aliases" node, to provide explicit name to CPU DAI. >>> >>> No, it shouldn't. Why do you need that? >>> >> >> Yes, this is not mandatory. This is convenient to identify sai >> controller, as we may have up to 8 instances. >> I can make it optional, or remove it if you think it is not relevant. > > Remove it. > One more thing regarding aliases. In future patches, I will add alsa controls to configure sai. Some stm32 socs exhibit several sai (4 instances for H7) It will be necessary to identify to which sai each control apply, through the control names: SAI1A control X SAI2A control X ... sub-node compatible already allows to identify A or B sub-instance. An alternative to alias, may be to use dev_name information to identify sai instance. But this does not produce user friendly control names. Another way is to use compatible also. The drawback here is to multiply compatible names. (many soc families and many instances) What is your recommendation ? If there is another solution, I did not consider, please let me know. >> >>> >>>> + >>>> +Optional properties: >>>> + - resets: Reference to a reset controller asserting the SAI >>>> + >>>> +SAI subnodes: >>>> +Two subnodes corresponding to SAI sub-block instances A et B can be defined. >>>> +Subnode can be omitted for unsused sub-block. >>>> + >>>> +SAI subnodes required properties: >>>> + - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" >>>> + for SAI sub-block A or B respectively. >>>> + - reg: Base address and size of SAI sub-block register set. >>>> + - clocks: Must contain one phandle and clock specifier pair >>>> + for sai_ck which feeds the internal clock generator. >>>> + - clock-names: Must contain "sai_ck". >>>> + - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt >>>> + - dma-names: identifier string for each DMA request line >>>> + "tx": if sai sub-block is configured as playback DAI >>>> + "rx": if sai sub-block is configured as capture DAI >>>> + - pinctrl-names: should contain only value "default" >>>> + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt >>>> + >>>> +Example: >>>> +sai1: sai1@40015800 { >>>> + compatible = "st,stm32f4-sai"; >>>> + reg = <0x40015800 0x400>; >> >> reg = <0x40015800 0x4>; >> >>>> + clocks = <&rcc 1 CLK_SAIQ_PDIV>, <&rcc 1 CLK_I2SQ_PDIV>; >>>> + clock-names = "clk_x8k", "clk_x11k"; >>>> + interrupts = <87>; >>>> + resets = <&rcc 310>; >>>> + >>>> + sai1b: sai@1 { >>> >>> unit address needs a reg property. >>> >>> audio-controller@1 >>> >> >> Right. This needs to be updated as follows: >> sai1b: sai1b@40015824 { > > audio-controller@... > >> ... >> reg = <0x40015824 0x1C>; >> ... >> }; >> >>>> + #sound-dai-cells = <0>; >>>> + compatible = "st,stm32-sai-sub-b"; >>>> + clocks = <&rcc 1 CLK_SAI2>; >>>> + clock-names = "sai_ck"; >>>> + dmas = <&dma2 5 0 0x400 0x1>; >>>> + dma-names = "tx"; >>>> + }; >>>> +}; >>>> + >>>> +sound { >>>> + compatible = "simple-audio-card"; >>> >>> Perhaps you want to use the graph-card here. It's getting close to >>> merging. >>> >> >> mclk-fs property is required when SAI is configured as master. >> It seems that mclk-fs is not supported in current version >> of audio graph card. So I thing I should keep simple card for the time >> being. Anyway, I plan further updates for SAI, so I will >> have opportunity to change to graph card at that time. > > Well, it's not easy to switch later and maintain compatibility. Is it > complicated to add mclk-fs property? The intent is graph-card supports > everything simple-card supports. > > Rob > Right, it should not be too complicated to add mclk-fs support to graph-card. BRs olivier

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[alsa-devel] [PATCH] ALSA: x86: Remove CONFIG_SND_X86 as unnecessary
by Ian W MORRISON 21 Mar '17

21 Mar '17

Create a specific menu for the CONFIG_HDMI_LPE_AUDIO setting. Add additional narrative to further explain the function of the CONFIG_HDMI_LPE_AUDIO setting. Remove subsequently unnecessary CONFIG_SND_X86 setting. Signed-off-by: Ian W Morrison <linuxium(a)linuxium.com.au> --- sound/x86/Kconfig | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc..a89d769 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,10 +1,5 @@ -menuconfig SND_X86 - tristate "X86 sound devices" - depends on X86 - ---help--- - X86 sound devices that don't fall under SoC or PCI categories - -if SND_X86 +menu "HDMI LPE for SoC audio support" + comment "Required for HDMI audio support on Intel Baytrail and Cherrytrail" config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" @@ -13,4 +8,4 @@ config HDMI_LPE_AUDIO help Choose this option to support HDMI LPE Audio mode -endif # SND_X86 +endmenu -- 1.9.1

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Re: [alsa-devel] [PATCH v2] ALSA: x86: Select CONFIG_HDMI_LPE_AUDIO as default
by Ian W MORRISON 21 Mar '17

21 Mar '17

On 21 March 2017 at 09:02, Ian W MORRISON <ianwmorrison(a)gmail.com> wrote: > > > On 21 March 2017 at 04:57, Takashi Iwai <tiwai(a)suse.de> wrote: > >> On Mon, 20 Mar 2017 16:12:01 +0100, >> Ian W MORRISON wrote: >> > >> > On 20 March 2017 at 21:04, Takashi Iwai <tiwai(a)suse.de> wrote: >> > >> > > On Mon, 20 Mar 2017 10:41:07 +0100, >> > > Ian W MORRISON wrote: >> > > > >> > > > On 20 March 2017 at 19:41, Takashi Iwai <tiwai(a)suse.de> wrote: >> > > > >> > > > > On Mon, 20 Mar 2017 09:17:30 +0100, >> > > > > Ian W MORRISON wrote: >> > > > > > >> > > > > > Oops ... forgot to copy alsa-devel and Pierre-Louis. >> > > > > > >> > > > > > On 20 March 2017 at 18:59, Takashi Iwai <tiwai(a)suse.de> wrote: >> > > > > > >> > > > > > > On Mon, 20 Mar 2017 08:42:32 +0100, >> > > > > > > Ian W MORRISON wrote: >> > > > > > > > >> > > > > > > > The upstream kernel builds for distributions such as Ubuntu >> > > which now >> > > > > > > > includes binary packages for v4.11 mainline kernel release >> > > > > candidates are >> > > > > > > > promoted as a way of testing upstream kernels to to confirm >> that >> > > > > upstream >> > > > > > > > has fixed a specific issue (see https://wiki.ubuntu.com/ >> > > > > > > > Kernel/MainlineBuilds). >> > > > > > > > >> > > > > > > > Unfortunately the long awaited patch for providing HDMI >> audio >> > > > > support for >> > > > > > > > Bay Trail and Cherry Trail devices does not include this >> support >> > > > > through >> > > > > > > a >> > > > > > > > module built by default. >> > > > > > > > >> > > > > > > > Through including by default of the two associated CONFIG >> > > settings >> > > > > > > (SND_X86 >> > > > > > > > and HDMI_LPE_AUDIO), upstream kernel builds would >> automatically >> > > > > provide >> > > > > > > the >> > > > > > > > much desired HDMI audio support by default. >> > > > > > > > >> > > > > > > > This patch uses a Kconfig 'default' statement to include the >> > > driver >> > > > > as >> > > > > > > > default. >> > > > > > > > >> > > > > > > > Changes in version 2: CONFIG_SND_X86 now a bool and changed >> > > default >> > > > > m to >> > > > > > > > default y >> > > > > > > > >> > > > > > > > Signed-off-by: Ian W Morrison <linuxium(a)linuxium.com.au> >> > > > > > > > --- >> > > > > > > > sound/x86/Kconfig | 4 +++- >> > > > > > > > 1 file changed, 3 insertions(+), 1 deletion(-) >> > > > > > > > >> > > > > > > > diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig >> > > > > > > > index 84c8f8fc..cac2270 100644 >> > > > > > > > --- a/sound/x86/Kconfig >> > > > > > > > +++ b/sound/x86/Kconfig >> > > > > > > > @@ -1,6 +1,7 @@ >> > > > > > > > menuconfig SND_X86 >> > > > > > > > - tristate "X86 sound devices" >> > > > > > > > + bool "X86 sound devices" >> > > > > > > > depends on X86 >> > > > > > > > + default y >> > > > > > > >> > > > > > > This one is OK, but ... >> > > > > > > >> > > > > > > > ---help--- >> > > > > > > > X86 sound devices that don't fall under SoC or PCI >> > > > > categories >> > > > > > > > >> > > > > > > > @@ -9,6 +10,7 @@ if SND_X86 >> > > > > > > > config HDMI_LPE_AUDIO >> > > > > > > > tristate "HDMI audio without HDaudio on Intel Atom >> > > platforms" >> > > > > > > > depends on DRM_I915 >> > > > > > > > + default y >> > > > > > > >> > > > > > > ... this is wrong. Each driver config itself should be left >> > > > > > > unspecified. >> > > > > > > >> > > > > > > It's distributor's job to choose the right config here. >> > > > > > > >> > > > > > > Actually this goes back to one of my earlier points: A >> distributor >> > > > > doesn't >> > > > > > have to set 'HDMI' as HDMI audio is automatically provided. >> > > > > >> > > > > Provided by who...? >> > > > > >> > > > >> > > > >> > > > I suppose it is provided as a result of the architecture one runs >> the >> > > > primary Makefile on. With 'uname -m' returning 'x86_64' running >> 'make >> > > > defconfig' results in 'CONFIG_HDMI=y' being set so >> > > 'drivers/video/Makefile' >> > > > automatically makes 'drivers/video/hdmi.c' and as >> > > > 'arch/x86/configs/x86_64_defconfig' includes 'CONFIG_DRM=y' and >> > > > 'CONFIG_DRM_I915=y' and 'drivers/gpu/drm/i915/Makefile' makes >> > > > 'intel_audio.c'. >> > > >> > > The defconfig stuff supports only the limited device sets that are >> > > supposed to be very common. Is HDMI_LPE_AUDIO classified really as >> > > such a thing? I don't know... >> > > >> > > >> > In this instance I'd suggest that HDMI_LPE_AUDIO is common as without it >> > you do not get HDMI audio on all x86_64 devices. However see my comment >> > below. >> > >> > >> > > >> > > > > > This is just an >> > > > > > extension because by setting 'HDMI_LPE_AUDIO' the missing audio >> > > support >> > > > > for >> > > > > > BYT and CHT SoCs is then provided. Therefore, in this albeit >> unusual >> > > > > > instance, I reason is it appropriate to set HDMI_LPE_AUDIO so >> that >> > > audio >> > > > > is >> > > > > > automatically provided regardless of distribution. If a >> distributor >> > > > > didn't >> > > > > > want to allow audio for BYT and CHT SoC based devices running >> their >> > > > > distro >> > > > > > then they could always remove it from their distro specific >> config. >> > > > > >> > > > > It's a wrong approach. What we're discussing about is just a >> > > > > configuration for a new individual driver, and the same rule >> should be >> > > > > applied to it like others. >> > > > > >> > > > > Check other drivers. See whether default=y (or =m) is set to >> > > > > CONFIG_E1000E, as a random example. With your argument, it >> should be >> > > > > set to y or m as default, since the Ethernet functionality is >> already >> > > > > provided by the network core. >> > > > > >> > > > > In general, we don't set the default values to the driver configs >> > > > > unless there is a VERY specific reason to do so. >> > > > > >> > > > > >> > > > I'm trying to get HDMI audio by default for BYT & CHT SoC but >> currently >> > > > this requires 'CONFIG_HDMI_LPE_AUDIO' set with a value of 'm'. I >> don't >> > > want >> > > > to argue a 'special case' or promote a 'wrong approach' but just >> get HDMI >> > > > audio working so is there another way to achieve this? >> > > >> > > What's wrong with manually setting CONFIG_HDMI_LPE_AUDIO=m or =y? >> > > IOW, do all features on your CHT/BYT machines work without adjusting >> > > manually after defconfig? If you have to do it in anyway, what makes >> > > it special for HDMI_LPD_AUDIO? >> > > >> > > >> > I've just tested various scenarios using a variety of devices with >> > v4.11-rc3. Having CONFIG_HDMI_LPE_AUDIO set by default is sufficient to >> get >> > audio on BYT/CHT devices over HDMI. >> > >> > For example installing Ubuntu 16.04.2 on a BYT/CHT device, then >> upgrading >> > the kernel to v4.11-rc3 using the deb-binary from >> http://kernel.ubuntu.com/~ >> > kernel-ppa/mainline/v4.11-rc3/ (an example of a current distro which is >> > compiling without setting CONFIG_HDMI_LPE_AUDIO) and there is no audio >> over >> > HDMI. Alternatively downloading v4.11-rc3 source, making defconfig, >> > manually setting CONFIG_HDMI_LPE_AUDIO, compiling and installing the >> > kernel on the BYT/CHT device will give you audio over HDMI. >> > >> > If the BYT/CHT device has other sound cards (e.g accessed via a >> headphone >> > jack) then audio without setting CONFIG_HDMI_LPE_AUDIO may be possible >> > depending on the codec used and whether it is provided by default in the >> > kernel. So for example a Tronsmart Ara (CHT) device has audio over >> > headphones using the pre-compiled deb-binary above but no audio over >> HDMI. >> > Whereas with a (1st gen) Intel Compute Stick (BYT) device no audio is >> > possible using the pre-compiled deb-binary as the device only has an >> HDMI >> > port. To get sound you must therefore currently use a self-compiled >> kernel >> > with CONFIG_HDMI_LPE_AUDIO set. >> > >> > Whilst Canonical may set CONFIG_HDMI_LPE_AUDIO to provide audio on >> BYT/CHT >> > devices by when is unknown and there is no guarantee they actually will. >> > >> > This will be similar for all other distros with the worst case scenario >> > that no distributor adds CONFIG_HDMI_LPE_AUDIO to their configs >> resulting >> > in (arguably) little point including HDMI LPE audio support in the >> kernel. >> > >> > I've laboured the point about 'BYT/CHT devices audio over HDMI' but to >> > answer your question completely there are other BYT/CHT devices >> (typically >> > tablets) that need a whole other bunch of patches to fix this and that. >> > They may need specialist kernel configs with specific settings but that >> is >> > a separate issue to the basic functional need of having audio over HDMI. >> > >> > Does the above explanation justify my current approach or is there an >> > alternative that can be considered? >> >> Sorry, no, you've explained how much you love the HDMI audio, but it >> doesn't answer my question clearly. >> > > Whatever love there was is diminishing rapidly! > > Let me set the context: the range of BYT/CHT devices sold as HDMI sticks > or HDMI boxes which are designed to connect to an HDMI output device (TV or > monitor) and typically have Windows installed as the OS and are used > primarily as media PCs, HTPCs or as 'mini' PCs. They become essentially > useless for their designated purpose when installed with a mainstream Linux > based OS such as Ubuntu at the moment because there is no audio over HDMI. > > >> >> My question is simple: are *all* features the upstream kernel supports >> for your device now (e.g. input, network or whatever) enabled by the >> pure arch/x86/configs/x86_64_defconfig *without* manually setting the >> configs to y or m? I bet you likely have to set some configs there >> manually to m or y. >> >> > No. CONFIG_HDMI_LPE_AUDIO is the only config that needs to be manually set > to get *all* features from the upstream kernel. > > >> Then, if the manual selection for such missing configs is acceptable, >> what makes the HDMI audio so special and it must be set to y/m as >> default, while other devices aren't set like that...? That's the >> question. >> >> >> > The requested patch sets CONFIG_HDMI_LPE_AUDIO so mainstream Linux distro > kernels would automatically provide the new kernel feature of HDMI LPE > audio support. > Further analysis of Canonical's v4.11-rcN build compared with other available distro builds of v4.11-rcN leads me to conclude that given CONFIG_HDMI_LPE_AUDIO is not set by default then Canonical's config is at fault. Consequently I have raised a bug report ( https://bugs.launchpad.net/ubuntu/+bug/1674523) in order to (hopefully) address the issue. I will additionally submit a further patch that (arguably) improves the current Kconfig file for the CONFIG_HDMI_LPE_AUDIO setting by creating a specific menu for the CONFIG_HDMI_LPE_AUDIO setting, adding additional narrative to further explain the function of the CONFIG_HDMI_LPE_AUDIO setting and removing the subsequently unnecessary CONFIG_SND_X86 setting. As a result, this patch ([PATCH v2] ALSA: x86: Select CONFIG_HDMI_LPE_AUDIO as default) is withdrawn. > > >> thanks, >> >> Takashi >> > >

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[alsa-devel] [PATCH v5 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 21 Mar '17

21 Mar '17

Hi Rob, Mark These are v5 of OF-graph base audio card patch-set. It fixed small things which was pointed by Rob. 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH v1 0/2] conf/ucm: Remove intel UCM conf files from alsa-lib repo
by fuweix.tang＠intel.com 21 Mar '17

21 Mar '17

From: Fuwei Tang <fuweix.tang(a)intel.com> Remove the intel UCM configs from alsa-lib repo, and move them to a dedicated UCM and topology file repo. Another series will add them to a dedicated UCM conf repo. License can be changed to BSD for intel files, and left as "unknown" for other conf files when they are moved from alsa-lib to the new repo. Fuwei Tang (2): conf/ucm: Remove conf files of broxton-rt298 platform conf/ucm: Remove conf files of skylake-rt286 platform configure.ac | 2 - src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/broxton-rt298/Hdmi1 | 22 ----- src/conf/ucm/broxton-rt298/Hdmi2 | 14 --- src/conf/ucm/broxton-rt298/HiFi | 129 -------------------------- src/conf/ucm/broxton-rt298/Makefile.am | 4 - src/conf/ucm/broxton-rt298/broxton-rt298.conf | 58 ------------ src/conf/ucm/skylake-rt286/Hdmi1 | 20 ---- src/conf/ucm/skylake-rt286/Hdmi2 | 20 ---- src/conf/ucm/skylake-rt286/HiFi | 129 -------------------------- src/conf/ucm/skylake-rt286/Makefile.am | 4 - src/conf/ucm/skylake-rt286/skylake-rt286.conf | 60 ------------ 12 files changed, 1 insertion(+), 463 deletions(-) delete mode 100644 src/conf/ucm/broxton-rt298/Hdmi1 delete mode 100644 src/conf/ucm/broxton-rt298/Hdmi2 delete mode 100644 src/conf/ucm/broxton-rt298/HiFi delete mode 100644 src/conf/ucm/broxton-rt298/Makefile.am delete mode 100644 src/conf/ucm/broxton-rt298/broxton-rt298.conf delete mode 100644 src/conf/ucm/skylake-rt286/Hdmi1 delete mode 100644 src/conf/ucm/skylake-rt286/Hdmi2 delete mode 100644 src/conf/ucm/skylake-rt286/HiFi delete mode 100644 src/conf/ucm/skylake-rt286/Makefile.am delete mode 100644 src/conf/ucm/skylake-rt286/skylake-rt286.conf -- 2.7.4

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[alsa-devel] [PATCH v1 0/3] ucm-conf: Add intel UCM conf files to alsa-ucm-conf repo
by fuweix.tang＠intel.com 21 Mar '17

21 Mar '17

From: Fuwei Tang <fuweix.tang(a)intel.com> Add the intel UCM configs to a dedicated UCM conf repo.License can be changed to BSD for intel files,and left as "unknown" for other conf files when they are moved from alsa-lib to the new repo.The alsa-lib LGPL license does not cover runtime configuration files. The configs are moved from alsa-lib repo.The original authorship and commit message of all config files will be reserved. I write the setup code from scratch instead of copying it from Takashi in this version. Fuwei Tang (1): ucm-conf: Set up for installation Vinod Koul (2): ucm-conf: broxton: add broxton-rt298 conf files ucm-conf: skylake: add skylake-rt286 conf files .gitignore | 13 ++++ ChangeLog | 1 + INSTALL | 1 + Makefile.am | 1 + broxton-rt298/Hdmi1 | 22 +++++++ broxton-rt298/Hdmi2 | 14 +++++ broxton-rt298/HiFi | 129 +++++++++++++++++++++++++++++++++++++++ broxton-rt298/Makefile.am | 4 ++ broxton-rt298/broxton-rt298.conf | 58 ++++++++++++++++++ configure.ac | 24 ++++++++ gitcompile | 5 ++ skylake-rt286/Hdmi1 | 20 ++++++ skylake-rt286/Hdmi2 | 20 ++++++ skylake-rt286/HiFi | 129 +++++++++++++++++++++++++++++++++++++++ skylake-rt286/Makefile.am | 4 ++ skylake-rt286/skylake-rt286.conf | 60 ++++++++++++++++++ 16 files changed, 505 insertions(+) create mode 100644 .gitignore create mode 100644 ChangeLog create mode 120000 INSTALL create mode 100644 Makefile.am create mode 100644 broxton-rt298/Hdmi1 create mode 100644 broxton-rt298/Hdmi2 create mode 100644 broxton-rt298/HiFi create mode 100644 broxton-rt298/Makefile.am create mode 100644 broxton-rt298/broxton-rt298.conf create mode 100644 configure.ac create mode 100755 gitcompile create mode 100644 skylake-rt286/Hdmi1 create mode 100644 skylake-rt286/Hdmi2 create mode 100644 skylake-rt286/HiFi create mode 100644 skylake-rt286/Makefile.am create mode 100644 skylake-rt286/skylake-rt286.conf -- 2.7.4

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[alsa-devel] snd_hda_intel: probe of 0000:00:01.1 failed with error -2
by Jorge Fernandez Monteagudo 20 Mar '17

20 Mar '17

Hi all! I've a system with a custom distribution and a 3.16.7 linux kernel. Sometimes when the sound system is initialized I get this error and the sound is not working. I see in dmesg [ 36.104413] snd_hda_intel 0000:00:01.1: irq 80 for MSI/MSI-X [ 36.111812] sound hdaudioC1D0: autoconfig: line_outs=1 (0xa/0x0/0x0/0x0/0x0) type:line [ 36.111815] sound hdaudioC1D0: speaker_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 36.111817] sound hdaudioC1D0: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) [ 36.111819] sound hdaudioC1D0: mono: mono_out=0x0 [ 36.111820] sound hdaudioC1D0: dig-out=0x10/0x15 [ 36.111821] sound hdaudioC1D0: inputs: [ 36.111823] sound hdaudioC1D0: Mic=0xd [ 36.111824] sound hdaudioC1D0: Line=0xc [ 36.111826] sound hdaudioC1D0: dig-in=0x12 [ 36.115940] sound hdaudioC0D0: ignore pin 0x7, too many assigned pins [ 36.115942] sound hdaudioC0D0: autoconfig: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 36.115944] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 36.115945] sound hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 36.115946] sound hdaudioC0D0: mono: mono_out=0x0 [ 36.115947] sound hdaudioC0D0: dig-out=0x3/0x5 [ 36.115948] sound hdaudioC0D0: inputs: [ 37.119900] snd_hda_intel: probe of 0000:00:01.1 failed with error -2 [ 37.127608] sound hdaudioC0D0: autoconfig: line_outs=1 (0xa/0x0/0x0/0x0/0x0) type:line [ 37.127613] sound hdaudioC0D0: speaker_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 37.127615] sound hdaudioC0D0: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) [ 37.127617] sound hdaudioC0D0: mono: mono_out=0x0 [ 37.127618] sound hdaudioC0D0: dig-out=0x10/0x15 [ 37.127620] sound hdaudioC0D0: inputs: [ 37.127622] sound hdaudioC0D0: Mic=0xd [ 37.127624] sound hdaudioC0D0: Line=0xc [ 37.127625] sound hdaudioC0D0: dig-in=0x12 and # cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xff760000 irq 39 What's the meaning of this error? Regards, Jorge

1 0

[alsa-devel] [RFC PATCH 0/2] ASoC: wm8994: master clock issues
by olivier moysan 20 Mar '17

20 Mar '17

Hello, I face two issues using wm8994 codec driver. Below is a description of the use cases rising the problem. I added the patches which allow to fix these issues. These minimalist patches can be seen as a starting point for a discussion on the best way to handle the use cases exposed. - ASoC: wm8994: default master clock selection Use case: CPU DAI and codec are managed through simple card. Wolson codec wm8994 is set as slave of CPU DAI and CPU DAI feeds codec with master clock. Master clock is derived from mclk-fs property and provided to CPU DAI and codec through snd_soc_dai_set_sysclk API. Analysis: The simple card assumes id 0 for master clock. However wm8994 codec exposes master clock ids from 1 to 4. (MCLK1, MCLK2 ..) The proposal in this patch is to define a default selection regarding master clock when id 0 is provided. - ASoC: wm8994: delay aifxclk activation Use case: Wolson codec wm8994 is set as slave of CPU DAI and CPU DAI feeds codec with master clock. CPU DAI generates master clock when enabled in trigger callback. So master clock is available only at runtime. Master clock is not available when configuring codec through alsa controls. Analysis: When we configure aifx clock while master clock is not active, related registers are not updated in wm8894 codec. A way to go around this problem, is to delay aifxclk activation until SND_SOC_DAPM_POST_PMU DAPM stage. The purpose of patch here is to illustrate this workaround. What I cannot clearly figure out, is how this may impact other uses of wm8894 codec. I try to know if it's worth investigating further in this direction or if it will not be applicable. If not applicable, what maybe the other options ? regards olivier olivier moysan (2): ASoC: wm8994: default master clock selection ASoC: wm8994: delay aifxclk activation sound/soc/codecs/wm8994.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH 0/2] conf/ucm: Remove intel UCM conf files from alsa-lib repo
by fuweix.tang＠intel.com 20 Mar '17

20 Mar '17

From: Fuwei Tang <fuweix.tang(a)intel.com> Remove the intel UCM configs from alsa-lib repo. Another series will add them to a dedicated UCM conf repo.License can be changed to BSD for intel files,and left as "unknown" for other conf files when they are moved from alsa-lib to the new repo.The alsa-lib LGPL license does not cover runtime configuration files. Fuwei Tang (2): conf/ucm: Remove conf files of broxton-rt298 platform conf/ucm: Remove conf files of skylake-rt286 platfrom configure.ac | 2 - src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/broxton-rt298/Hdmi1 | 22 ----- src/conf/ucm/broxton-rt298/Hdmi2 | 14 --- src/conf/ucm/broxton-rt298/HiFi | 129 -------------------------- src/conf/ucm/broxton-rt298/Makefile.am | 4 - src/conf/ucm/broxton-rt298/broxton-rt298.conf | 58 ------------ src/conf/ucm/skylake-rt286/Hdmi1 | 20 ---- src/conf/ucm/skylake-rt286/Hdmi2 | 20 ---- src/conf/ucm/skylake-rt286/HiFi | 129 -------------------------- src/conf/ucm/skylake-rt286/Makefile.am | 4 - src/conf/ucm/skylake-rt286/skylake-rt286.conf | 60 ------------ 12 files changed, 1 insertion(+), 463 deletions(-) delete mode 100644 src/conf/ucm/broxton-rt298/Hdmi1 delete mode 100644 src/conf/ucm/broxton-rt298/Hdmi2 delete mode 100644 src/conf/ucm/broxton-rt298/HiFi delete mode 100644 src/conf/ucm/broxton-rt298/Makefile.am delete mode 100644 src/conf/ucm/broxton-rt298/broxton-rt298.conf delete mode 100644 src/conf/ucm/skylake-rt286/Hdmi1 delete mode 100644 src/conf/ucm/skylake-rt286/Hdmi2 delete mode 100644 src/conf/ucm/skylake-rt286/HiFi delete mode 100644 src/conf/ucm/skylake-rt286/Makefile.am delete mode 100644 src/conf/ucm/skylake-rt286/skylake-rt286.conf -- 2.7.4

3 4

[alsa-devel] [BUG] src/conf/alsa.conf: Hard-coded path `/etc/asound.conf`, needs `@func sysconfdir`.
by Alain Kalker 20 Mar '17

20 Mar '17

The file `src/conf/alsa.conf` contains the hard-coded path `/etc/asound.conf`, which in some use cases may not be correct. One particular use case: Termux[1], a "Linux subsystem for Android", plans to use ALSA plugins to communicate with the OpenSL ES implementation in Android. In doing so, it needs to not conflict with any vendor provided ALSA implentation, which could conceivably use an `/etc/asound.conf` of its own. At any rate, the system `/etc` directory is normally not writable by non-root users of Android. I would suggest adding a `snd_func_sysconfdir` in `src/confmisc.c (analogous to `snd_func_datadir`) so packagers and users can place a global `asound.conf` file in a directory to which they have write access and which doesn't conflict with the systems ALSA. The desired path can already be specified at build time by using `./configure --sysconfdir=<dir>`, which conveniently defaults to PREFIX/etc in any case. It is available as configure variable: `sysconfdir='${prefix}/etc'`, and only needs to be substituted to be available to `src/confmisc.c`. Kind regards, Alain Kalker [1]: https://termux.com/

2 1

Re: [alsa-devel] [PATCH] ALSA: hda/ca0132: Remove double parentheses
by Takashi Iwai 20 Mar '17

20 Mar '17

On Wed, 15 Mar 2017 23:41:23 +0100, Matthias Kaehlcke wrote: > > The extra pairs of parantheses are not needed and causes clang to > generate warnings like this: > > sound/pci/hda/patch_ca0132.c:1171:14: error: equality comparison with extraneous parentheses [-Werror,-Wparentheses-equality] > if ((buffer == NULL)) > ~~~~~~~^~~~~~~ > sound/pci/hda/patch_ca0132.c:1171:14: note: remove extraneous parentheses around the comparison to silence this warning > if ((buffer == NULL)) > ~ ^ ~ > sound/pci/hda/patch_ca0132.c:1171:14: note: use '=' to turn this equality comparison into an assignment > if ((buffer == NULL)) > > Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH 1/1] ALSA: es1688: Use strcpy() instead of sprintf()
by Nicolas Iooss 20 Mar '17

20 Mar '17

There is no point in using sprintf() without a format string when strcpy() can perform the same operation. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- sound/isa/es1688/es1688_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index e2cf508841b1..81cf26fa28d6 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -742,7 +742,7 @@ int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, int device) pcm->private_data = chip; pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - sprintf(pcm->name, snd_es1688_chip_id(chip)); + strcpy(pcm->name, snd_es1688_chip_id(chip)); chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, -- 2.12.0

2 1

[alsa-devel] [PATCH] ALSA: hda - add support for docking station for HP 840 G3
by Jaroslav Kysela 20 Mar '17

20 Mar '17

This tested patch adds missing initialization for Line-In/Out PINs for the docking station for HP 840 G3. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c15c51b..69266b8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -261,6 +261,7 @@ enum { CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, + CXT_FIXUP_HP_DOCK, CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, }; @@ -778,6 +779,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_eapd, }, + [CXT_FIXUP_HP_DOCK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21011020 }, /* line-out */ + { 0x18, 0x2181103f }, /* line-in */ + { } + } + }, [CXT_FIXUP_HP_SPECTRE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -839,6 +848,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), + SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), @@ -871,6 +881,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, + { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, {} }; -- 2.9.3

2 1

[alsa-devel] [PATCH] ALSA: hda - add support for docking station for HP 820 G2
by Jaroslav Kysela 20 Mar '17

20 Mar '17

This tested patch adds missing initialization for Line-In/Out PINs for the docking station for HP 820 G2. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73a0046..147bddb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4847,6 +4847,7 @@ enum { ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, ALC280_FIXUP_HP_DOCK_PINS, + ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -5388,6 +5389,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC280_FIXUP_HP_GPIO4 }, + [ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x21011020 }, /* line-out */ + { 0x18, 0x2181103f }, /* line-in */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HP_GPIO_MIC1_LED + }, [ALC280_FIXUP_HP_9480M] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_9480m, @@ -5646,7 +5657,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -5814,6 +5825,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, + {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, -- 2.9.3

2 1

[alsa-devel] [PATCH] rawmidi: virtual: fix reading into a small buffer
by Clemens Ladisch 20 Mar '17

20 Mar '17

In the special case for handling partial messages, the pointer calculations were wrong, which would result in data corruption. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- alsa-lib/src/rawmidi/rawmidi_virt.c +++ alsa-lib/src/rawmidi/rawmidi_virt.c @@ -263,8 +263,8 @@ static ssize_t snd_rawmidi_virtual_read(snd_rawmidi_t *rmidi, void *buffer, size } size1 = virt->in_buf_size - virt->in_buf_ofs; if ((size_t)size1 > size) { - virt->in_buf_ofs += size1 - size; - memcpy(buffer, virt->in_buf_ptr, size); + memcpy(buffer, virt->in_buf_ptr + virt->in_buf_ofs, size); + virt->in_buf_ofs += size; result += size; break; }

3 2

[alsa-devel] [PATCH] ALSA: ctxfi: Fix the incorrect check of dma_set_mask() call
by Takashi Iwai 20 Mar '17

20 Mar '17

In the commit [15c75b09f8d1: ALSA: ctxfi: Fallback DMA mask to 32bit], I forgot to put "!" at dam_set_mask() call check in cthw20k1.c (while cthw20k2.c is OK). This patch fixes that obvious bug. (As a side note: although the original commit was completely wrong, it's still working for most of machines, as it sets to 32bit DMA mask in the end. So the bug severity is low.) Fixes: 15c75b09f8d1 ("ALSA: ctxfi: Fallback DMA mask to 32bit") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ctxfi/cthw20k1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index ab4cdab5cfa5..79edd88d5cd0 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1905,7 +1905,7 @@ static int hw_card_start(struct hw *hw) return err; /* Set DMA transfer mask */ - if (dma_set_mask(&pci->dev, DMA_BIT_MASK(dma_bits))) { + if (!dma_set_mask(&pci->dev, DMA_BIT_MASK(dma_bits))) { dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(dma_bits)); } else { dma_set_mask(&pci->dev, DMA_BIT_MASK(32)); -- 2.11.1

1 0

[alsa-devel] [PATCH] ALSA: x86: Select CONFIG_HDMI_LPE_AUDIO as default
by Ian W MORRISON 20 Mar '17

20 Mar '17

The upstream kernel builds for distributions such as Ubuntu which now includes binary packages for v4.11 mainline kernel release candidates are promoted as a way of testing upstream kernels to to confirm that upstream has fixed a specific issue (see https://wiki.ubuntu.com/Kernel/MainlineBuilds). Unfortunately the long awaited patch for providing HDMI audio support for Bay Trail and Cherry Trail devices does not include this support through a module built by default. By defining the default of the two associated CONFIG settings (SND_X86 and HDMI_LPE_AUDIO) as a module, upstream kernel builds would automatically provide the much desired HDMI audio support by default. This patch uses a Kconfig 'default' statement to include the driver as default. Signed-off-by: Ian W Morrison <linuxium(a)linuxium.com.au> --- sound/x86/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc..439850c 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 tristate "X86 sound devices" depends on X86 + default m ---help--- X86 sound devices that don't fall under SoC or PCI categories @@ -9,6 +10,7 @@ if SND_X86 config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" depends on DRM_I915 + default m select SND_PCM help Choose this option to support HDMI LPE Audio mode -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: omap: constify snd_soc_ops structures
by Bhumika Goyal 20 Mar '17

20 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Cross compiled the .o files for arm architecture. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 2 +- sound/soc/omap/omap-twl4030.c | 2 +- sound/soc/omap/omap3pandora.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 25a33e9..d565102 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -49,7 +49,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops am3517evm_ops = { +static const struct snd_soc_ops am3517evm_ops = { .hw_params = am3517evm_hw_params, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fdecb70..71e5f31 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -124,7 +124,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; } -static struct snd_soc_ops n810_ops = { +static const struct snd_soc_ops n810_ops = { .startup = n810_startup, .hw_params = n810_hw_params, .shutdown = n810_shutdown, diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 89fe95e..614b18d 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -70,7 +70,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops omap_abe_ops = { +static const struct snd_soc_ops omap_abe_ops = { .hw_params = omap_abe_hw_params, }; diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 7431314..a24b0de 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -73,7 +73,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, return snd_soc_runtime_set_dai_fmt(rtd, fmt); } -static struct snd_soc_ops omap_twl4030_ops = { +static const struct snd_soc_ops omap_twl4030_ops = { .hw_params = omap_twl4030_hw_params, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 732e749..4e3de71 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -184,7 +184,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_ops omap3pandora_ops = { +static const struct snd_soc_ops omap3pandora_ops = { .hw_params = omap3pandora_hw_params, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index aa4053b..e409677 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -68,7 +68,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, return err; } -static struct snd_soc_ops osk_ops = { +static const struct snd_soc_ops osk_ops = { .startup = osk_startup, .hw_params = osk_hw_params, .shutdown = osk_shutdown, diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index a768457..66ac358 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -123,7 +123,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops rx51_ops = { +static const struct snd_soc_ops rx51_ops = { .startup = rx51_startup, .hw_params = rx51_hw_params, }; -- 2.7.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: rt5665: fix wrong shift rt5665_if2_1_adc_in_enum
by Bard Liao 20 Mar '17

20 Mar '17

The shift is RT5665_IF2_1_ADC_IN_SFT not RT5665_IF3_ADC_IN_SFT. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 5545d08..21b07ca 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2256,7 +2256,7 @@ static const char * const rt5665_if2_1_adc_in_src[] = { static const SOC_ENUM_SINGLE_DECL( rt5665_if2_1_adc_in_enum, RT5665_DIG_INF2_DATA, - RT5665_IF3_ADC_IN_SFT, rt5665_if2_1_adc_in_src); + RT5665_IF2_1_ADC_IN_SFT, rt5665_if2_1_adc_in_src); static const struct snd_kcontrol_new rt5665_if2_1_adc_in_mux = SOC_DAPM_ENUM("IF2_1 ADC IN Source", rt5665_if2_1_adc_in_enum); -- 2.7.4

1 1

[alsa-devel] USB Focusrite Scarlett 6i6 2nd Gen - Mixer not working (new VID, quirks and probably matrix needed)
by Matthias Andree 19 Mar '17

19 Mar '17

Greetings, I recently acquired a Focusrite Scarlett 6i6 *2nd Gen*. It appears to work for sound output (haven't tested much else yet), but mixing/routing won't work, and to add insult to injury, Pulseaudio misrepresents it as a multi-channel sound card, defaults to 5.1 and deprives it of the bass. Selecting 5.0 improves the situation as a workaround for stereo output. Hardware differences between 1st and 2nd Gen aside (where pad and inst/line switches moved into software), the interface does not create useful mixer entries. Digging around I figured we didn't have new stuff in the current kernel Git tree, so I chose to continue with what I had and found: in Ubuntu 16.04's 4.8.0 HWE kernel "Advanced Linux Sound Architecture Driver Version k4.8.0-42-generic.", I figured that the USB IDs (VID 0x1235 as usual, PID 0x8203 for the 6i6 2nd Gen) aren't recognized. I added them to the quirks section, and to the mixer_scarlett.c, to no avail. I get this on load, and the second message identically for each attempt to access the card through alsamixer, which alsamixer terminates with SIGPIPE. [10834.242351] usbcore: registered new interface driver snd-usb-audio [10834.329623] usb 2-6: cannot get ctl value: req = 0x81, wValue = 0x901, wIndex = 0x100, type = 4 With the patched module, sound stops after a few notes but I assume that's pulseaudio f*cking up, while it works with the unpatched module. So it appears we need to dig up and implement a bit of its innards and teach mixer_scarlett about it. How do we best go about getting this 2nd Gen Scarlett 6i6 fully supported? Do I need to sniff the USB communications on Windows while poking at the interface with its mixer control tool, or do we have other options? How did the 1st gen driver come to pass, and who did it and might be available for assistance? Is there any user-space poking we can do to dig up the needed information? Thanks. Cheers, Matthias

1 0

[alsa-devel] [RESEND PATCH v2 0/2] Add STM32 SAI support
by olivier moysan 18 Mar '17

18 Mar '17

This patch-set handles the Serial Audio Interface (SAI) IP on STM32 platforms. The SAI IP is composed of two Sub-block A and B. (see the figure below) Each sub-block makes use of individual and shared resources. Shared resources: - Reset line. Reset common and sub-block registers. - Bus interface clocks. This clock is not exposed in device as it it managed by clock driver, according to kernel clocks gating. - Common register. SAI IP exhibits a common configuration register to manage synchronization modes. NB: These modes are not yet implemented in this version of the SAI driver. - Interrupt. Sub-blocks have their own interrupt status registers but they share the same interrupt line. Individual resources: - Register set - DMA request line. - Communication interface. Each sub-block has its own GPIOs and associated bus lines. - Kernel clock. Each sub-block has its own dedicated clock for its communication interface. To reflect this architecture Sub-block A and B are handled by 2 child devices. Sub-block A and B devices can be configured independently either as transmitter or receiver. A PCM device is associated to each sub-block. A sub-block has to select the appropriated parent clock at runtime, depending on the audio stream sampling rate to be played or captured. Two parent clocks must be provided to support sampling rates multiples of 8 kHz or 11.025kHz. Interface clock +--------------------------------+ +---------------> | SAI IP | Reset | +-----------------+ | +---------------> | |common registers | | | +-----------------+ | | | | +----------------------+ | | | Sub-block A | | Kernel clock A | | | | Bus A +--------------------> | +----------------+ +----------------> | | |A registers | | | | | +----------------+ | | | +----------------------+ | | | | +----------------------+ | | | Sub-block B | | Kernel clock B | | | | Bus B +--------------------> | +----------------+ +-----------------> | | |B registers | | | | | +----------------+ | | | +----------------------+ | | | +--------------------------------+ Changes in v2: - Fix warnings olivier moysan (2): ASoC: stm32: add bindings for SAI ASoC: stm32: add SAI driver .../devicetree/bindings/sound/st,stm32-sai.txt | 79 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 6 + sound/soc/stm/stm32_sai.c | 123 +++ sound/soc/stm/stm32_sai.h | 202 +++++ sound/soc/stm/stm32_sai_sub.c | 899 +++++++++++++++++++++ 8 files changed, 1319 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_sai.c create mode 100644 sound/soc/stm/stm32_sai.h create mode 100644 sound/soc/stm/stm32_sai_sub.c -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: mediatek: constify snd_soc_ops structures
by Bhumika Goyal 18 Mar '17

18 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 1e7e8d4..aa5b31b 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -129,7 +129,7 @@ static int mt2701_cs42448_fe_ops_startup(struct snd_pcm_substream *substream) return 0; } -static struct snd_soc_ops mt2701_cs42448_48k_fe_ops = { +static const struct snd_soc_ops mt2701_cs42448_48k_fe_ops = { .startup = mt2701_cs42448_fe_ops_startup, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 46c8e6a..e0c2b23 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -67,7 +67,7 @@ static int mt8173_max98090_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops mt8173_max98090_ops = { +static const struct snd_soc_ops mt8173_max98090_ops = { .hw_params = mt8173_max98090_hw_params, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 467f704..5e383eb 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -75,7 +75,7 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt8173_rt5650_rt5514_ops = { +static const struct snd_soc_ops mt8173_rt5650_rt5514_ops = { .hw_params = mt8173_rt5650_rt5514_hw_params, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 1b8b2a7..fed1f15 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -79,7 +79,7 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt8173_rt5650_rt5676_ops = { +static const struct snd_soc_ops mt8173_rt5650_rt5676_ops = { .hw_params = mt8173_rt5650_rt5676_hw_params, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ba65f41..a784708 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -105,7 +105,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt8173_rt5650_ops = { +static const struct snd_soc_ops mt8173_rt5650_ops = { .hw_params = mt8173_rt5650_hw_params, }; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: samsung: constify snd_soc_ops structures
by Bhumika Goyal 18 Mar '17

18 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. Cross compiled the files arndale_rt5631.o, h1940_uda1380.o, jive_wm8750.o, littlemill.o, neo1973_wm8753.o and rx1950_uda1380.o for arm architecture. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/samsung/arndale_rt5631.c | 2 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 2 +- sound/soc/samsung/tobermory.c | 2 +- 15 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index ee1fda9..6f40587 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -53,7 +53,7 @@ static int arndale_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops arndale_ops = { +static const struct snd_soc_ops arndale_ops = { .hw_params = arndale_hw_params, }; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 5f5825f..3c8d58a 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -120,7 +120,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops h1940_ops = { +static const struct snd_soc_ops h1940_ops = { .startup = h1940_startup, .hw_params = h1940_hw_params, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 7fcb51f..529b10d 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -79,7 +79,7 @@ static int jive_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops jive_ops = { +static const struct snd_soc_ops jive_ops = { .hw_params = jive_hw_params, }; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 7cb204e..d057eae 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -135,7 +135,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops littlemill_ops = { +static const struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 65602b9..8e097f2 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -114,7 +114,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) /* * Neo1973 WM8753 HiFi DAI opserations. */ -static struct snd_soc_ops neo1973_hifi_ops = { +static const struct snd_soc_ops neo1973_hifi_ops = { .hw_params = neo1973_hifi_hw_params, .hw_free = neo1973_hifi_hw_free, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index fa096ab..324e16a 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -73,7 +73,7 @@ static struct snd_soc_jack_gpio hp_jack_gpios[] = { }, }; -static struct snd_soc_ops rx1950_ops = { +static const struct snd_soc_ops rx1950_ops = { .startup = rx1950_startup, .hw_params = rx1950_hw_params, }; diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index dcc008d..6de63f3 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -211,7 +211,7 @@ static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) return 0; } -static struct snd_soc_ops simtec_snd_ops = { +static const struct snd_soc_ops simtec_snd_ops = { .hw_params = simtec_hw_params, }; diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 81a7894..a6a4bc7 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -199,7 +199,7 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops s3c24xx_uda134x_ops = { +static const struct snd_soc_ops s3c24xx_uda134x_ops = { .startup = s3c24xx_uda134x_startup, .shutdown = s3c24xx_uda134x_shutdown, .hw_params = s3c24xx_uda134x_hw_params, diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 425ee2b..72408ae 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -77,7 +77,7 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, /* * SmartQ WM8987 HiFi DAI operations. */ -static struct snd_soc_ops smartq_hifi_ops = { +static const struct snd_soc_ops smartq_hifi_ops = { .hw_params = smartq_hifi_hw_params, }; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index a2f2363..7fc7cc6 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -144,7 +144,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops smdk_spdif_ops = { +static const struct snd_soc_ops smdk_spdif_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 6e4dfa7..5c3980b 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -93,7 +93,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8580 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index ff57b19..307122f 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -80,7 +80,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8994 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e62149..7bbb822 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -91,7 +91,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops smdk_wm8994_pcm_ops = { +static const struct snd_soc_ops smdk_wm8994_pcm_ops = { .hw_params = smdk_wm8994_pcm_hw_params, }; diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 24cc9d6..25e5be9 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -131,7 +131,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, return tm2_start_sysclk(rtd->card); } -static struct snd_soc_ops tm2_aif1_ops = { +static const struct snd_soc_ops tm2_aif1_ops = { .hw_params = tm2_aif1_hw_params, }; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 3310eda..097663d 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -110,7 +110,7 @@ static int tobermory_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tobermory_ops = { +static const struct snd_soc_ops tobermory_ops = { .hw_params = tobermory_hw_params, }; -- 2.7.4

2 1

[alsa-devel] dmix/dshare: discrepancy between snd_pcm_{status, state, delay}
by Cheng Sun 18 Mar '17

18 Mar '17

Hi all, I've been tracking down the cause of an indefinite hang in PortAudio whenever an xrun occurs, and bisected the issue to the following. After commits faf53c19 (dshare) and 38a2d2ed (dmix), the semantics of snd_pcm_state and snd_pcm_delay are no longer the same as a query to snd_pcm_status, and a subsequent snd_pcm_status_get_*. Note that these two commits remove the lines that set the state field in the snd_pcm_status_t: - status->state = snd_pcm_state(dshare->spcm); This causes the behaviour where, when a dmix's slave suffers an xrun, and the dmix's status is queried, snd_pcm_status_get_state returns RUNNING, and yet snd_pcm_state returns XRUN. The reason why I am not attaching a patch right now is because I am confused as to the commit message in faf53c19, which explicitly states: Note: snd_pcm_dshare_delay() is not updated. Which leads me to think that this change in semantics was perhaps deliberate (if not documented anywhere). However I'm not sure what the rationale is, so I'm not clear on the right way to go about fixing it. If anyone could shed any further light into this, I'd be very grateful. Cheers, Cheng

1 0

[alsa-devel] [PATCH v3 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 18 Mar '17

18 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..c643278 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -64,6 +64,10 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) { + snd_config_delete((snd_config_t*)plug->rate_converter); + plug->rate_converter = NULL; + } assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1112,14 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; + if (rate_converter) { + if ((err = snd_config_copy((snd_config_t**)&plug->rate_converter, + (snd_config_t *)rate_converter)) < 0) { + free(plug); + return err; + } + } + plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1119,6 +1130,10 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, err = snd_pcm_new(&pcm, SND_PCM_TYPE_PLUG, name, slave->stream, slave->mode); if (err < 0) { + if (plug->rate_converter) { + snd_config_delete((snd_config_t*)plug->rate_converter); + plug->rate_converter = NULL; + } free(plug); return err; } -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: cs35l35: Stash dev pointer directly rather than CODEC pointer
by Charles Keepax 17 Mar '17

17 Mar '17

The driver stashes a CODEC pointer in the cs35l35_private structure, which is used to obtain a struct device pointer for error messages in the interrupt handler. However, doing so is not very safe as the interrupt is registered, as it should be in bus probe, but the CODEC pointer can't be safely stored until the ASoC level probe. This leaves a window between the two probes where if any interrupts are received a NULL pointer will be deferenced in the IRQ handler. Fix this issue by saving a pointer to the device directly and passing that to the error messages in the interrupt handler rather than using the CODEC pointer to access the device pointer. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 29 ++++++++++++++--------------- sound/soc/codecs/cs35l35.h | 2 +- 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 05117fc..1d9f332 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -741,8 +741,6 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) struct monitor_cfg *monitor_config = &cs35l35->pdata.mon_cfg; int ret; - cs35l35->codec = codec; - /* Set Platform Data */ if (cs35l35->pdata.bst_vctl) regmap_update_bits(cs35l35->regmap, CS35L35_BST_CVTR_V_CTL, @@ -1004,7 +1002,6 @@ static struct regmap_config cs35l35_regmap = { static irqreturn_t cs35l35_irq(int irq, void *data) { struct cs35l35_private *cs35l35 = data; - struct snd_soc_codec *codec = cs35l35->codec; unsigned int sticky1, sticky2, sticky3, sticky4; unsigned int mask1, mask2, mask3, mask4, current1; @@ -1032,7 +1029,7 @@ static irqreturn_t cs35l35_irq(int irq, void *data) /* handle the interrupts */ if (sticky1 & CS35L35_CAL_ERR) { - dev_crit(codec->dev, "Calibration Error\n"); + dev_crit(cs35l35->dev, "Calibration Error\n"); /* error is no longer asserted; safe to reset */ if (!(current1 & CS35L35_CAL_ERR)) { @@ -1051,10 +1048,10 @@ static irqreturn_t cs35l35_irq(int irq, void *data) } if (sticky1 & CS35L35_AMP_SHORT) { - dev_crit(codec->dev, "AMP Short Error\n"); + dev_crit(cs35l35->dev, "AMP Short Error\n"); /* error is no longer asserted; safe to reset */ if (!(current1 & CS35L35_AMP_SHORT)) { - dev_dbg(codec->dev, "Amp short error release\n"); + dev_dbg(cs35l35->dev, "Amp short error release\n"); regmap_update_bits(cs35l35->regmap, CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, 0); @@ -1069,11 +1066,11 @@ static irqreturn_t cs35l35_irq(int irq, void *data) } if (sticky1 & CS35L35_OTW) { - dev_warn(codec->dev, "Over temperature warning\n"); + dev_warn(cs35l35->dev, "Over temperature warning\n"); /* error is no longer asserted; safe to reset */ if (!(current1 & CS35L35_OTW)) { - dev_dbg(codec->dev, "Over temperature warn release\n"); + dev_dbg(cs35l35->dev, "Over temperature warn release\n"); regmap_update_bits(cs35l35->regmap, CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, 0); @@ -1088,10 +1085,10 @@ static irqreturn_t cs35l35_irq(int irq, void *data) } if (sticky1 & CS35L35_OTE) { - dev_crit(codec->dev, "Over temperature error\n"); + dev_crit(cs35l35->dev, "Over temperature error\n"); /* error is no longer asserted; safe to reset */ if (!(current1 & CS35L35_OTE)) { - dev_dbg(codec->dev, "Over temperature error release\n"); + dev_dbg(cs35l35->dev, "Over temperature error release\n"); regmap_update_bits(cs35l35->regmap, CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, 0); @@ -1106,7 +1103,7 @@ static irqreturn_t cs35l35_irq(int irq, void *data) } if (sticky3 & CS35L35_BST_HIGH) { - dev_crit(codec->dev, "VBST error: powering off!\n"); + dev_crit(cs35l35->dev, "VBST error: powering off!\n"); regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, CS35L35_PDN_AMP, CS35L35_PDN_AMP); regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, @@ -1114,7 +1111,7 @@ static irqreturn_t cs35l35_irq(int irq, void *data) } if (sticky3 & CS35L35_LBST_SHORT) { - dev_crit(codec->dev, "LBST error: powering off!\n"); + dev_crit(cs35l35->dev, "LBST error: powering off!\n"); regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, CS35L35_PDN_AMP, CS35L35_PDN_AMP); regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, @@ -1122,13 +1119,13 @@ static irqreturn_t cs35l35_irq(int irq, void *data) } if (sticky2 & CS35L35_VPBR_ERR) - dev_dbg(codec->dev, "Error: Reactive Brownout\n"); + dev_dbg(cs35l35->dev, "Error: Reactive Brownout\n"); if (sticky4 & CS35L35_VMON_OVFL) - dev_dbg(codec->dev, "Error: VMON overflow\n"); + dev_dbg(cs35l35->dev, "Error: VMON overflow\n"); if (sticky4 & CS35L35_IMON_OVFL) - dev_dbg(codec->dev, "Error: IMON overflow\n"); + dev_dbg(cs35l35->dev, "Error: IMON overflow\n"); return IRQ_HANDLED; } @@ -1365,6 +1362,8 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, if (!cs35l35) return -ENOMEM; + cs35l35->dev = dev; + i2c_set_clientdata(i2c_client, cs35l35); cs35l35->regmap = devm_regmap_init_i2c(i2c_client, &cs35l35_regmap); if (IS_ERR(cs35l35->regmap)) { diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index c203081..156d2f0 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -265,7 +265,7 @@ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) struct cs35l35_private { - struct snd_soc_codec *codec; + struct device *dev; struct cs35l35_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[2]; -- 2.1.4

3 2

[alsa-devel] [RFC PATCH v2 0/2] master clock issues
by olivier moysan 17 Mar '17

17 Mar '17

Hello, I face two issues using wm8994 codec driver. Below is a description of the use cases rising the problem. I added the patches which allow to fix these issues. These minimalist patches can be seen as a starting point for a discussion on the best way to handle the use cases exposed. - ASoC: wm8994: default master clock selection Use case: CPU DAI and codec are managed through simple card. Wolson codec wm8994 is set as slave of CPU DAI and CPU DAI feeds codec with master clock. Master clock is derived from mclk-fs property and provided to CPU DAI and codec through snd_soc_dai_set_sysclk API. Analysis: The simple card assumes id 0 for master clock. However wm8994 codec exposes master clock ids from 1 to 4. (MCLK1, MCLK2 ..) The proposal in this patch is to define a default selection regarding master clock when id 0 is provided. - ASoC: wm8994: delay aifxclk activation Use case: Wolson codec wm8994 is set as slave of CPU DAI and CPU DAI feeds codec with master clock. CPU DAI generates master clock when enabled in trigger callback. So master clock is available only at runtime. Master clock is not available when configuring codec through alsa controls. Analysis: When we configure aifx clock while master clock is not active, related registers are not updated in wm8894 codec. A way to go around this problem, is to delay aifxclk activation until SND_SOC_DAPM_POST_PMU DAPM stage. The purpose of patch here is to illustrate this workaround. What I cannot clearly figure out, is how this may impact other uses of wm8894 codec. I try to know if it's worth investigating further in this direction or if it will not be applicable. If not applicable, what maybe the other options ? Changes in v2: Add change log on aifxclk delay patch. regards olivier olivier moysan (2): ASoC: wm8994: default master clock selection ASoC: wm8994: delay aifxclk activation sound/soc/codecs/wm8994.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: rt286: fix headphone click/crack noise on Dell XPS 9343 I2S mode
by Kai-Heng Feng 17 Mar '17

17 Mar '17

HDA mode fixed the issue by these two commits: '9476d369d7b3 ALSA: hda - Mute headphone pin on suspend on XPS13 9333' '3e1b0c4a9d56 ALSA: hda - Fix click noise at start on Dell XPS13' Apply the same workarounds to rt286 can solve the issue. When jack is plugged, it rapidly generates I2C interrupts, which triggers rt286_irq() and rt286_jack_detect(), which produces the click noise. alc_fixup_dell_xps13() in patch_realtek.c sets up a pin that can stop the frantic interrupts, hence avoids the click noise. When rt286 is under powersaving state, play a sound with headphone or plug a headphone in will produce a loud crack sound. Set AMP_OUT_MUTE before power events can make the noise less noticeable. Link: https://bugzilla.kernel.org/show_bug.cgi?id=112611 Link: https://bugzilla.redhat.com/show_bug.cgi?id=1313434 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/soc/codecs/rt286.c | 48 +++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 45 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..ec4caef045e9 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,6 +36,9 @@ #define RT286_VENDOR_ID 0x10ec0286 #define RT288_VENDOR_ID 0x10ec0288 +#define AMP_OUT_MUTE 0xb080 +#define AMP_OUT_UNMUTE 0xb000 + struct rt286_priv { struct reg_default *index_cache; int index_cache_size; @@ -47,6 +50,7 @@ struct rt286_priv { struct delayed_work jack_detect_work; int sys_clk; int clk_id; + bool is_dell_dino; }; static const struct reg_default rt286_index_def[] = { @@ -472,6 +476,32 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } +/* Power event function to workaround headphone crack noise */ +static int rt286_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!rt286->is_dell_dino) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_MUTE); + break; + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_UNMUTE); + break; + default: + return 0; + } + + return 0; +} + static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -578,7 +608,9 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, - RT286_SET_PIN_SFT, 0, NULL, 0), + RT286_SET_PIN_SFT, 0, rt286_hp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output Mixer */ SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, @@ -1175,8 +1207,10 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + rt286->is_dell_dino = dmi_check_system(dmi_dell_dino) ? true : false; + if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + rt286->is_dell_dino) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1192,6 +1226,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { + /* Fix headphone click noise */ + if (rt286->is_dell_dino) + regmap_write(rt286->regmap, + RT286_MIC1_DET_CTRL, 0x0020); + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0x5000); } @@ -1215,7 +1254,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (rt286->is_dell_dino) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, @@ -1224,6 +1263,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, RT286_SET_GPIO_DATA, 0x40, 0x40); regmap_update_bits(rt286->regmap, RT286_GPIO_CTRL, 0xc, 0x8); + /* Workaound headphone crack noise when probing */ + regmap_write(rt286->regmap, RT286_SET_AMP_GAIN_HPO, + AMP_OUT_MUTE); } if (rt286->i2c->irq) { -- 2.12.0

2 3

[alsa-devel] [PATCH v2 1/3] dt-bindings: Document the dmas and dma-names properties for VC4 HDMI
by Eric Anholt 16 Mar '17

16 Mar '17

From: Boris Brezillon <boris.brezillon(a)free-electrons.com> These are optional, but necessary for HDMI audio support. Signed-off-by: Boris Brezillon <boris.brezillon(a)free-electrons.com> Signed-off-by: Eric Anholt <eric(a)anholt.net> Acked-by: Rob Herring <robh(a)kernel.org> --- v2: no changes Documentation/devicetree/bindings/display/brcm,bcm-vc4.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/display/brcm,bcm-vc4.txt b/Documentation/devicetree/bindings/display/brcm,bcm-vc4.txt index 34c7fddcea39..ca02d3e4db91 100644 --- a/Documentation/devicetree/bindings/display/brcm,bcm-vc4.txt +++ b/Documentation/devicetree/bindings/display/brcm,bcm-vc4.txt @@ -34,6 +34,9 @@ Optional properties for HDMI: - hpd-gpios: The GPIO pin for HDMI hotplug detect (if it doesn't appear as an interrupt/status bit in the HDMI controller itself). See bindings/pinctrl/brcm,bcm2835-gpio.txt +- dmas: Should contain one entry pointing to the DMA channel used to + transfer audio data +- dma-names: Should contain "audio-rx" Required properties for DPI: - compatible: Should be "brcm,bcm2835-dpi" -- 2.11.0

2 4

Re: [alsa-devel] [RFC PATCH 00/13] Introduce first class virtual address spaces
by Thomas Gleixner 16 Mar '17

16 Mar '17

On Wed, 15 Mar 2017, Till Smejkal wrote: > On Wed, 15 Mar 2017, Andy Lutomirski wrote: > > > VAS segments on the other side would provide a functionality to > > > achieve the same without the need of any mounted filesystem. However, > > > I agree, that this is just a small advantage compared to what can > > > already be achieved with the existing functionality provided by the > > > Linux kernel. > > > > I see this "small advantage" as "resource leak and security problem". > > I don't agree here. VAS segments are basically in-memory files that are > handled by the kernel directly without using a file system. Hence, if an Why do we need yet another mechanism to represent something which looks like a file instead of simply using existing mechanisms and extend them? Thanks, tglx

2 3

[alsa-devel] [PATCH] ASoC: rcar: dma: remove unnecessary "volatile"
by Kuninori Morimoto 16 Mar '17

16 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 2a3af642eb20("ASoC: rcar: clear DE bit only in PDMACHCR...") added rsnd_dmapp_bset(), but it used copy-paste. Thus, it had unnecessary "volatile", and had below warning on x86. This patch fix it. sound/soc/sh/rcar/dma.c: In function 'rsnd_dmapp_bset': >> sound/soc/sh/rcar/dma.c:463:21: warning: passing argument 1 of \ 'ioread32' discards 'volatile' qualifier from pointer target \ type [-Wdiscarded-qualifiers] u32 val = ioread32(addr); ^~~~ In file included from arch/x86/include/asm/io.h:203:0, from arch/x86/include/asm/realmode.h:5, from arch/x86/include/asm/acpi.h:33, from arch/x86/include/asm/fixmap.h:19, from arch/x86/include/asm/apic.h:10, from arch/x86/include/asm/smp.h:12, from include/linux/smp.h:59, from include/linux/topology.h:33, from include/linux/gfp.h:8, from include/linux/idr.h:16, from include/linux/kernfs.h:14, from include/linux/sysfs.h:15, from include/linux/kobject.h:21, from include/linux/of.h:21, from include/linux/of_dma.h:16, from sound/soc/sh/rcar/dma.c:12: include/asm-generic/iomap.h:31:21: note: expected 'void *' \ but argument is of type 'volatile void *' extern unsigned int ioread32(void __iomem *); ^~~~~~~~ Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index c2e199b..241cb3b 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -459,7 +459,7 @@ static void rsnd_dmapp_bset(struct rsnd_dma *dma, u32 data, u32 mask, u32 reg) struct rsnd_mod *mod = rsnd_mod_get(dma); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); - volatile void __iomem *addr = rsnd_dmapp_addr(dmac, dma, reg); + void __iomem *addr = rsnd_dmapp_addr(dmac, dma, reg); u32 val = ioread32(addr); val &= ~mask; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5665: fix define of RT5665_HP_DRIVER_5X
by Bard Liao 16 Mar '17

16 Mar '17

It is (0x3 << 2), not (0x2 << 2). Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 12f7080..a30f5e6 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1106,7 +1106,7 @@ #define RT5665_HP_DRIVER_MASK (0x3 << 2) #define RT5665_HP_DRIVER_1X (0x0 << 2) #define RT5665_HP_DRIVER_3X (0x1 << 2) -#define RT5665_HP_DRIVER_5X (0x2 << 2) +#define RT5665_HP_DRIVER_5X (0x3 << 2) #define RT5665_LDO1_DVO_MASK (0x3) #define RT5665_LDO1_DVO_09 (0x0) #define RT5665_LDO1_DVO_10 (0x1) -- 2.7.4

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[alsa-devel] [PATCH 0/4] ASoC: simple-card: tidyup unreadable code
by Kuninori Morimoto 16 Mar '17

16 Mar '17

Hi Mark These are not a big deal, but tidyup/cleanup unreadable code on simple-card/simple-scu-card driver. Kuninori Morimoto (4): 1) ASoC: simple-card: use defined dev on probe() 2) ASoC: simple-scu-card: use defined dev on probe() 3) ASoC: simple-card: add new simple_priv_to_card() macro 4) ASoC: simple-scu-card: add new simple_priv_to_card() macro sound/soc/generic/simple-card.c | 43 +++++++++++++++++++++---------------- sound/soc/generic/simple-scu-card.c | 37 +++++++++++++++++-------------- 2 files changed, 45 insertions(+), 35 deletions(-) -- 1.9.1

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[alsa-devel] [RFC PATCH 0/2] wm8960: Relax bit clock computation
by Daniel Baluta 16 Mar '17

16 Mar '17

This patch series allows running S20_3LE encoded samples on wm8960 codec. First patch does a small refactoring of sysclk frequency search because wm8960_configure_sysclk was getting pretty convoluted. The second patch allows relaxing bitclock computation in the way that if an exact bitclk couldn't be derived from sysclk it chooses the smalles available bitclk greater than the desired bitclk. For a better reviewing experience you can checked the squashed version of this patches here: http://pastebin.com/gBJ3DBSu Note that this is a follow up of: "ASoC: wm8960: Use physical width for bclk" discussion archived here: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-March/118735.html Daniel Baluta (2): ASoC: codec: wm8960: Refactor sysclk freq search ASoC: codec: wm8960: Relax bit clock computation sound/soc/codecs/wm8960.c | 113 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 91 insertions(+), 22 deletions(-) -- 2.7.4

3 6

[alsa-devel] Applied "ASoC: rockchip: add bindings for rk3368 i2s" to the asoc tree
by Mark Brown 15 Mar '17

15 Mar '17

The patch ASoC: rockchip: add bindings for rk3368 i2s has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f49efce4ed6fdec17a32406eccb56fc167f43b6c Mon Sep 17 00:00:00 2001 From: Jianqun Xu <jay.xu(a)rock-chips.com> Date: Wed, 15 Mar 2017 14:14:00 +0800 Subject: [PATCH] ASoC: rockchip: add bindings for rk3368 i2s Add devicetree bindings for i2s controller found on rk3368 processors from rockchip. Signed-off-by: Jianqun Xu <jay.xu(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index a6600f6dea64..206aba1b34bb 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -9,6 +9,7 @@ Required properties: - "rockchip,rk3066-i2s": for rk3066 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 + - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 - reg: physical base address of the controller and length of memory mapped region. -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: tas2552: Improve DT binding document example
by Fabio Estevam 15 Mar '17

15 Mar '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> Improve DT binding document example by providing the mandatory regulator properties. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/tas2552.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt index c49992c..075df91 100644 --- a/Documentation/devicetree/bindings/sound/tas2552.txt +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -25,6 +25,9 @@ Example: tas2552: tas2552@41 { compatible = "ti,tas2552"; reg = <0x41>; + vbat-supply = <&reg_vbat>; + iovdd-supply = <&reg_iovdd>; + avdd-supply = <&reg_avdd>; enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; }; -- 2.7.4

3 2

[alsa-devel] Applied "ASoC: pxa: constify snd_soc_ops structures" to the asoc tree
by Mark Brown 15 Mar '17

15 Mar '17

The patch ASoC: pxa: constify snd_soc_ops structures has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 943b73112dfdd6747871a3347f70de6876f4dd4b Mon Sep 17 00:00:00 2001 From: Bhumika Goyal <bhumirks(a)gmail.com> Date: Tue, 14 Mar 2017 01:16:40 +0530 Subject: [PATCH] ASoC: pxa: constify snd_soc_ops structures Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. The .o files did not compile for all the changed .c files. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/brownstone.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/hx4700.c | 2 +- sound/soc/pxa/imote2.c | 2 +- sound/soc/pxa/magician.c | 4 ++-- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/raumfeld.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- 11 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index b6cb9950f05d..9a3f5b799720 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -74,7 +74,7 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, } /* machine stream operations */ -static struct snd_soc_ops brownstone_ops = { +static const struct snd_soc_ops brownstone_ops = { .hw_params = brownstone_wm8994_hw_params, }; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 311774e9ca46..054e0d65db9d 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -154,7 +154,7 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops corgi_ops = { +static const struct snd_soc_ops corgi_ops = { .startup = corgi_startup, .hw_params = corgi_hw_params, .shutdown = corgi_shutdown, diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 85483049b916..a9ac881c2e14 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -79,7 +79,7 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops hx4700_ops = { +static const struct snd_soc_ops hx4700_ops = { .hw_params = hx4700_hw_params, }; diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 9d0e40771ef5..78475376f971 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -42,7 +42,7 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops imote2_asoc_ops = { +static const struct snd_soc_ops imote2_asoc_ops = { .hw_params = imote2_asoc_hw_params, }; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2d4d4455fe87..2fc012b06c43 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -255,12 +255,12 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops magician_capture_ops = { +static const struct snd_soc_ops magician_capture_ops = { .startup = magician_startup, .hw_params = magician_capture_hw_params, }; -static struct snd_soc_ops magician_playback_ops = { +static const struct snd_soc_ops magician_playback_ops = { .startup = magician_startup, .hw_params = magician_playback_hw_params, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index a879aba0691f..b6693f32fc02 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -129,7 +129,7 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops poodle_ops = { +static const struct snd_soc_ops poodle_ops = { .startup = poodle_startup, .hw_params = poodle_hw_params, .shutdown = poodle_shutdown, diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 47c91730e93c..111a907c4eb9 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -132,7 +132,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops raumfeld_cs4270_ops = { +static const struct snd_soc_ops raumfeld_cs4270_ops = { .startup = raumfeld_cs4270_startup, .shutdown = raumfeld_cs4270_shutdown, .hw_params = raumfeld_cs4270_hw_params, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d38a2b519c52..1671da648e95 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -156,7 +156,7 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops spitz_ops = { +static const struct snd_soc_ops spitz_ops = { .startup = spitz_startup, .hw_params = spitz_hw_params, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 1812748b75bf..ae9c12e1ea2a 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -85,7 +85,7 @@ static int tosa_startup(struct snd_pcm_substream *substream) return 0; } -static struct snd_soc_ops tosa_ops = { +static const struct snd_soc_ops tosa_ops = { .startup = tosa_startup, }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 6d88d9acc666..5b0eccd2b4dd 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -152,7 +152,7 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static struct snd_soc_ops z2_ops = { +static const struct snd_soc_ops z2_ops = { .hw_params = z2_hw_params, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 6fbcdf02c88d..ba468e560dd2 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -132,7 +132,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops zylonite_voice_ops = { +static const struct snd_soc_ops zylonite_voice_ops = { .hw_params = zylonite_voice_hw_params, }; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: fsl: constify snd_soc_ops structures
by Bhumika Goyal 15 Mar '17

15 Mar '17

Declare snd_soc_ops structures as const as they are only stored in the ops field of a snd_soc_dai_link structure. This field is of type const, so snd_soc_ops structures having this property can be made const too. The following .o files did not compile: sound/soc/fsl/{p1022_rdk.c/p1022_ds.c/mpc8610_hpcd.c} Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- 8 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 883087f..84ef638 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -64,7 +64,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops eukrea_tlv320_snd_ops = { +static const struct snd_soc_ops eukrea_tlv320_snd_ops = { .hw_params = eukrea_tlv320_hw_params, }; diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index bb04590..9d19b80 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -48,7 +48,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, return snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16); } -static struct snd_soc_ops imx_mc13783_hifi_ops = { +static const struct snd_soc_ops imx_mc13783_hifi_ops = { .hw_params = imx_mc13783_hifi_hw_params, }; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index ddf49f3..a639b52 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -174,7 +174,7 @@ static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) /** * mpc8610_hpcd_ops: ASoC machine driver operations */ -static struct snd_soc_ops mpc8610_hpcd_ops = { +static const struct snd_soc_ops mpc8610_hpcd_ops = { .startup = mpc8610_hpcd_startup, }; diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index 198eeb3..d7ec3d2 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -73,7 +73,7 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { +static const struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index a1f780e..41c623c 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -184,7 +184,7 @@ static int p1022_ds_machine_remove(struct snd_soc_card *card) /** * p1022_ds_ops: ASoC machine driver operations */ -static struct snd_soc_ops p1022_ds_ops = { +static const struct snd_soc_ops p1022_ds_ops = { .startup = p1022_ds_startup, }; diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index d4d88a8..4afbdd6 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -188,7 +188,7 @@ static int p1022_rdk_machine_remove(struct snd_soc_card *card) /** * p1022_rdk_ops: ASoC machine driver operations */ -static struct snd_soc_ops p1022_rdk_ops = { +static const struct snd_soc_ops p1022_rdk_ops = { .startup = p1022_rdk_startup, }; diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index ae403c2..66fb6c4 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -23,7 +23,7 @@ static struct snd_soc_card imx_phycore; -static struct snd_soc_ops imx_phycore_hifi_ops = { +static const struct snd_soc_ops imx_phycore_hifi_ops = { }; static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index b454972..cdaf163 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -139,7 +139,7 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops wm1133_ev1_ops = { +static const struct snd_soc_ops wm1133_ev1_ops = { .hw_params = wm1133_ev1_hw_params, }; -- 2.7.4

2 1

[alsa-devel] [PATCH 00/30] ASoC: hdac_hdmi: Add support to enable MST audio
by jeeja.kp＠intel.com 15 Mar '17

15 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> Display port 1.2 introduces new capability Multi-stream transport (MST) which will allow multiple stream to route to single display port where multiple monitors are connected. Multiple monitors will be connected by a MST Hub or a monitor capable of daisy-chaining. With MST support, a pin can support multiple ports and on each port a monitor can be connected. This patch series enables DP MST Audio by extending the current design from pin to port. o With MST, pin mux for CVT selection will be used to select CVT for pin-port instead of pin. So create the pin mux accordingly. o Pin is marked as MST capable when connect event reports pipe as valid. if pipe is -1, then pin is not MST capable and default to port 0. if pipe is valid, then pin is MST capable and port = pipe. o For MST capable pin, select the device entry before configuring the pin widget verbs controlling the sink device operations. o Add support to handle multiple Port mapped to same converter by creating as port list in pcm and report jack event based on the no of ports connected/disconnected . o i915 acomp APIs are now used to read the ELD info and support is extended for MST port as well. o Configure port/pin/audio infoframe in widget event handlers for dynamic routing when monitor is connected/disconnected instead of DAI ops. o Use set_tdm slot to set the stream tag instead of dai params and store it in pcm context and use this in configuring the converter in widget event handler. o Use pin widget to represent the output to enable/disable path when a device is connected/disconnected. Added API to create the pin widget and this will be called by the machine driver after jack creation. o Add channel map support for bxt machine. o Move from snd_jack to ASOC jack framework, create the jack in machine driver. o Create port pin widget to enable and disable the path dynamically when a jack is connected/disconnected. Note: This patch series has dependency on the series "ASoC: Intel: Skylake: Driver updates" Jeeja KP (29): ASoC: hdac_hdmi: Register widget event handlers ASoC: Intel: Skylake: Use set_tdm_slot to set the dma channel ASoC: hdac_hdmi: Move channel info from pin to PCM structure ASoC: Intel: bxt: add channel map support in rt298 machine ASoC: Intel: bxt: add channel map support in bxt_da7219_max98357a machine ASoC: hdac_hdmi: Begin to add support for DP Multi-stream audio ASoC: Intel: Skylake: Add route change to rt286 machine ASoC: Intel: Skylake: Add route change to nau88l25_max98357a machine ASoC: Intel: Skylake: Add route change to nau88l25_ssm4567 machine ASoC: Intel: bxt: Add route change to rt298 machine ASoC: Intel: bxt: Add route change to da7219_max98357a machine ASoC: hdac_hdmi: Add support to handle MST capable pin ASoC: hdac_hdmi: Add MST verb support ASoC: hdac_hdmi: Handle MST pin jack detection at boot/resume ASoc: hdac_hdmi: Configure pin verbs for MST ASoC: hdac_hdmi: Add support for multiple ports to a PCM ASoC: hdac_hdmi: Use ASoC jack instead of snd_jack ASoC: Intel: Skylake: Create ASoC jack for hdmi in rt286 machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in skl_nau88l25_max98357a machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in skl_nau88l25_ssm4567 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_rt298 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_da7219_max98357 machine ASoC: hdac_hdmi: Add machine pin widget for each port ASoC: Intel: Skylake: Add jack port initialize in rt286 machine ASoC: Intel: Skylake: Add jack port initialize in skl_nau88l25_max98357a machine ASoC: Intel: Skylake: Add jack port initialize in skl_nau88l25_ssm4567 machine ASoC: Intel: Skylake: Add jack port initialize in bxt_rt298 machine ASoC: Intel: Skylake: Add jack port initialize in bxt_da7219_max98357a machine ASoC: Intel: Skylake: Fix to delete DSP pipe after stopping pipe Sandeep Tayal (1): ASoC: hdac_hdmi: use audio component framework to read ELD sound/soc/codecs/hdac_hdmi.c | 1331 ++++++++++++++--------- sound/soc/codecs/hdac_hdmi.h | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 75 +- sound/soc/intel/boards/bxt_rt298.c | 78 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 33 +- sound/soc/intel/boards/skl_rt286.c | 30 +- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 10 +- 9 files changed, 1045 insertions(+), 553 deletions(-) -- 2.5.0

3 54

[alsa-devel] [PATCH 00/10] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 15 Mar '17

15 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides updates and fixes on the following: Fix parameter overwrite for KPB DSP Module. To handle module download with size > dma buffer size. Fix to handle dsp pipes correctly. Disable notification after dsp boot. Remove get dsp_ops in cleanup routine. Disable interrupt when DSP is in D3. Update DSP core state in D0. Reload the firmware in case of D3 failure. Dharageswari R (1): ASoC: Intel: Skylake: Fix parameter overwrite for KPB Module G Kranthi (2): ASoC: Intel: Skylake: Disable notifications at boot after DSP FW init ASoC: Intel: Skylake: Remove get dsp_ops in cleanup routine Guneshwor Singh (1): ASoC: Intel: Skylake: Fix not to stop src pipe in pre pmd event handler Jeeja KP (6): ASoC: Intel: Skylake: Fix to delete DSP pipe after stopping pipe ASoC: Intel: bxtn: Disable interrupt when DSP is in D3 ASoC: Intel: bxtn: Update DSP core state in D0 ASoC: Intel: bxtn: Reload the firmware in case of D3 failure ASoC: Intel: Skylake: Remove BE prepare ops ASoC: Intel: Skylake: Fix module load when module size > DMA buffer size sound/soc/intel/skylake/bxt-sst.c | 14 ++++++- sound/soc/intel/skylake/skl-messages.c | 12 ++---- sound/soc/intel/skylake/skl-pcm.c | 19 +--------- sound/soc/intel/skylake/skl-sst-cldma.c | 26 ++++++++----- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 6 ++- sound/soc/intel/skylake/skl-sst-dsp.h | 3 ++ sound/soc/intel/skylake/skl-sst-ipc.c | 67 ++++++++++++++++++++++----------- sound/soc/intel/skylake/skl-sst-ipc.h | 7 ++++ sound/soc/intel/skylake/skl-sst.c | 49 ++++++++++++++++++------ sound/soc/intel/skylake/skl-topology.c | 56 ++++++++++++++------------- 11 files changed, 162 insertions(+), 99 deletions(-) -- 2.5.0

4 20

[alsa-devel] [PATCH V2 00/10] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 15 Mar '17

15 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides updates and fixes on the following: Fix parameter overwrite for KPB DSP Module. To handle module download with size > dma buffer size. Fix to handle dsp pipes correctly. Disable notification after dsp boot. Remove get dsp_ops in cleanup routine. Disable interrupt when DSP is in D3. Update DSP core state in D0. Reload the firmware in case of D3 failure. Changes in v2: - fixed the typo's in commit message. - handle error in case when ipc fails in case of load module. Dharageswari R (1): ASoC: Intel: Skylake: Fix parameter overwrite for KPB Module G Kranthi (2): ASoC: Intel: Skylake: Disable notifications at boot after DSP FW init ASoC: Intel: Skylake: Remove get dsp_ops in cleanup routine Guneshwor Singh (1): ASoC: Intel: Skylake: Fix not to stop src pipe in pre pmd event handler Jeeja KP (6): ASoC: Intel: Skylake: Fix to delete DSP pipe after stopping pipe ASoC: Intel: bxtn: Disable interrupt when DSP is in D3 ASoC: Intel: bxtn: Update DSP core state in D0 ASoC: Intel: bxtn: Reload the firmware in case of D3 failure ASoC: Intel: Skylake: Remove BE prepare ops ASoC: Intel: Skylake: Fix module load when module size > DMA buffer size sound/soc/intel/skylake/bxt-sst.c | 14 ++++++- sound/soc/intel/skylake/skl-messages.c | 12 ++---- sound/soc/intel/skylake/skl-pcm.c | 19 +--------- sound/soc/intel/skylake/skl-sst-cldma.c | 26 ++++++++----- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 6 ++- sound/soc/intel/skylake/skl-sst-dsp.h | 3 ++ sound/soc/intel/skylake/skl-sst-ipc.c | 66 ++++++++++++++++++++++----------- sound/soc/intel/skylake/skl-sst-ipc.h | 7 ++++ sound/soc/intel/skylake/skl-sst.c | 53 ++++++++++++++++++++------ sound/soc/intel/skylake/skl-topology.c | 56 ++++++++++++++-------------- 11 files changed, 165 insertions(+), 99 deletions(-) -- 2.5.0

3 13

[alsa-devel] [PATCH] ASoC: rcar: clear DE bit only in PDMACHCR when it stops
by Kuninori Morimoto 15 Mar '17

15 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car datasheet indicates "Clear DE in PDMACHCR" for transfer stop, but current code clears all bits in PDMACHCR. Because of this, DE bit might never been cleared, and it causes CMD overflow. This patch fixes this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 1f405c8..c2e199b 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -454,6 +454,20 @@ static u32 rsnd_dmapp_read(struct rsnd_dma *dma, u32 reg) return ioread32(rsnd_dmapp_addr(dmac, dma, reg)); } +static void rsnd_dmapp_bset(struct rsnd_dma *dma, u32 data, u32 mask, u32 reg) +{ + struct rsnd_mod *mod = rsnd_mod_get(dma); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + volatile void __iomem *addr = rsnd_dmapp_addr(dmac, dma, reg); + u32 val = ioread32(addr); + + val &= ~mask; + val |= (data & mask); + + iowrite32(val, addr); +} + static int rsnd_dmapp_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -461,10 +475,10 @@ static int rsnd_dmapp_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); int i; - rsnd_dmapp_write(dma, 0, PDMACHCR); + rsnd_dmapp_bset(dma, 0, PDMACHCR_DE, PDMACHCR); for (i = 0; i < 1024; i++) { - if (0 == rsnd_dmapp_read(dma, PDMACHCR)) + if (0 == (rsnd_dmapp_read(dma, PDMACHCR) & PDMACHCR_DE)) return 0; udelay(1); } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] ASoC: Add AU1761 audio codec as selectable option
by me＠rockwellcollins.com 15 Mar '17

15 Mar '17

From: Bryce Ferguson <bryce.ferguson(a)rockwellcollins.com> This commit adds the ADI AU1761 audio codec as a selectable option in the kernel config. Currently the driver can only be selected for ADI blackfin devices or if SND_SOC_ALL_CODECS is enabled. Signed-off-by: Bryce Ferguson <bryce.ferguson(a)rockwellcollins.com> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e49e9da..43539c3b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -303,12 +303,14 @@ config SND_SOC_ADAU1761 select SND_SOC_ADAU17X1 config SND_SOC_ADAU1761_I2C - tristate + tristate "Analog Devices AU1761 CODEC - I2C" + depends on I2C select SND_SOC_ADAU1761 select REGMAP_I2C config SND_SOC_ADAU1761_SPI - tristate + tristate "Analog Devices AU1761 CODEC - SPI" + depends on SPI select SND_SOC_ADAU1761 select REGMAP_SPI -- 2.7.4

4 3

[alsa-devel] [PATCH v2 00/12] Add support for the audio codec on Allwinner V3s
by Icenowy Zheng 15 Mar '17

15 Mar '17

Allwinner V3s features a audio codec with dedicated digital and analog parts, like the ones on A23/H3, but much simpler (lack of MIC2, LINE IN and MBIAS). Add support for it. In order to make the codec usable, DMA support is also added in this series; the support of Lichee Pi Zero's dock board is also added here, as it's the only board hackable that come with ports connected to the codec. Patch 1~3 split out parts that is not available on V3s in the analog codec. Patch 4/5 adds support for V3s in analog/digital codec. Patch 6 add the gate bit as a common quirk of sun6i-dma driver, as V3s also needs it. Patch 7 really adds support for V3s in DMA engine. Patch 8 restores the inclusion of CCU headers in the DTSI file of V3s, as it's removed when merging. Patch 9/10/11 adds three parts of V3s: DMA engine, codec support and pinmux of mmc1 (used on Lichee Pi Zero dock). Patch 12 adds support for Lichee Pi Zero dock, with support of mmc1 and codec. Icenowy Zheng (12): ASoC: sun8i-codec-analog: split out mic2 ASoC: sun8i-codec-analog: split out line in ASoC: sun8i-codec-analog: split out mbias ASoC: sun8i-codec-analog: add support for V3s SoC ASoC: sun4i-codec: Add support for V3s codec dmaengine: sun6i: make gate bit in sun8i's DMA engines a common quirk dmaengine: sun6i: support V3s SoC variant ARM: dts: sun8i: restore the inclusion of ccu headers in V3s DTSI ARM: dts: sun8i: add DMA engine in V3s DTSI ARM: dts: sun8i: add audio codec support into V3s DTSI ARM: dts: sun8i: add pinmux for V3s mmc1 ARM: dts: sun8i: add device tree for Lichee Pi Zero with Dock .../devicetree/bindings/dma/sun6i-dma.txt | 1 + .../devicetree/bindings/sound/sun4i-codec.txt | 11 +- .../bindings/sound/sun8i-codec-analog.txt | 1 + arch/arm/boot/dts/Makefile | 3 +- arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts | 67 +++++++ arch/arm/boot/dts/sun8i-v3s.dtsi | 103 ++++++---- drivers/dma/sun6i-dma.c | 22 ++- sound/soc/sunxi/sun4i-codec.c | 59 ++++++ sound/soc/sunxi/sun8i-codec-analog.c | 211 +++++++++++++++++---- 9 files changed, 405 insertions(+), 73 deletions(-) create mode 100644 arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts -- 2.11.1

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[alsa-devel] [PATCH 0/3] ASoC: add dio2125 amplifier support
by Jerome Brunet 15 Mar '17

15 Mar '17

This patchset adds the support for the dio2125 from Dioo microcircuits. The dio2125 is audio driver with adjustable gain. The EN pad allows to enable or disable the outputs of the driver. The link to the datasheet is not provided as part of the commit. It is not provided the manufacturer and it is doubtful it will still be available a few years from now. Still provided here in case anyone is interested [0]. This patchset has been tested on Amlogic's meson-gxbb-p200 board. [0]: http://www.winine.com/Line%E9%9F%B3/DIO2125XM10%20V1.0.pdf Jerome Brunet (3): dt-bindings: add vendor prefix for Dioo ASoC: dio2125: add dio2125 amp driver ASoC: dio2125: add dt-bindings .../devicetree/bindings/sound/dioo,dio2125.txt | 12 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dio2125.c | 120 +++++++++++++++++++++ 5 files changed, 140 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/dioo,dio2125.txt create mode 100644 sound/soc/codecs/dio2125.c -- 2.9.3

4 12

[alsa-devel] [RFC PATCH] ASoC: wm8960: Use physical width for bclk
by Daniel Baluta 15 Mar '17

15 Mar '17

bclk is derived from sysclk with the help of bclk_divs. Anyhow, for S20_3LE format there is no bclk_divs that could be used to derive an exact bclk. We can fix this by using storage size instead the exact number of bits of the sample when computing bclk. With this approach we can play S20_3LE encoded files at the cost of some unused BCLK cycles for S20_3LE and S24_LE. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/codecs/wm8960.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e1429e3..18ca5fc 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -722,7 +722,8 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; - wm8960->bclk = snd_soc_params_to_bclk(params); + wm8960->bclk = params_physical_width(params) * + params_channels(params) * params_rate(params); if (params_channels(params) == 1) wm8960->bclk *= 2; -- 2.7.4

4 4

[alsa-devel] [PATCH v2 14/14] ASoC: sunxi: simplify optional reset handling
by Philipp Zabel 15 Mar '17

15 Mar '17

As of commit bb475230b8e5 ("reset: make optional functions really optional"), the reset framework API calls use NULL pointers to describe optional, non-present reset controls. This allows to return errors from devm_reset_control_get_optional and to call reset_control_deassert unconditionally. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Acked-by: Mark Brown <broonie(a)kernel.org> Acked-by: Chen-Yu Tsai <wens(a)csie.org> --- sound/soc/sunxi/sun4i-spdif.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index eaefd07a5ed08..cdf8a6d86010c 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -521,13 +521,12 @@ static int sun4i_spdif_probe(struct platform_device *pdev) if (quirks->has_reset) { host->rst = devm_reset_control_get_optional(&pdev->dev, NULL); - if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) { - ret = -EPROBE_DEFER; + if (IS_ERR(host->rst)) { + ret = PTR_ERR(host->rst); dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); return ret; } - if (!IS_ERR(host->rst)) - reset_control_deassert(host->rst); + reset_control_deassert(host->rst); } ret = devm_snd_soc_register_component(&pdev->dev, -- 2.11.0

1 0

[alsa-devel] [PATCH v2] ASoC: hdmi-codec: fix non static symbol warnings
by Takashi Sakamoto 14 Mar '17

14 Mar '17

Fixes the following sparse warnings: $ make C=2 sound/soc/codecs/hdmi-codec.o sound/soc/codecs/hdmi-codec.c:34:1: warning: symbol 'hdmi_device_list' was not declared. Should it be static? sound/soc/codecs/hdmi-codec.c:80:33: warning: symbol 'hdmi_codec_stereo_chmaps' was not declared. Should it be static? sound/soc/codecs/hdmi-codec.c:87:33: warning: symbol 'hdmi_codec_8ch_chmaps' was not declared. Should it be static? sound/soc/codecs/hdmi-codec.c:354:6: warning: symbol 'hdmi_codec_eld_chmap' was not declared. Should it be static? Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> --- sound/soc/codecs/hdmi-codec.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1f..6ba6aa5 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -31,7 +31,7 @@ struct hdmi_device { int cnt; }; #define pos_to_hdmi_device(pos) container_of((pos), struct hdmi_device, list) -LIST_HEAD(hdmi_device_list); +static LIST_HEAD(hdmi_device_list); static DEFINE_MUTEX(hdmi_mutex); #define DAI_NAME_SIZE 16 @@ -78,14 +78,14 @@ struct hdmi_codec_cea_spk_alloc { }; /* Channel maps stereo HDMI */ -const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = { +static const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = { { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { } }; /* Channel maps for multi-channel playbacks, up to 8 n_ch */ -const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { +static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { { .channels = 2, /* CA_ID 0x00 */ .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, /* CA_ID 0x01 */ @@ -352,7 +352,7 @@ static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) return spk_mask; } -void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp) +static void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp) { u8 spk_alloc; unsigned long spk_mask; -- 2.9.3

3 3

Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Mark Brown 14 Mar '17

14 Mar '17

On Mon, Mar 13, 2017 at 02:52:56PM -0700, Brian Norris wrote: > Sure, I can understand frustration with essentially contentless "bug > reports." And I got frustrated by a seemingly pat response that was > (IMO) a bit mistargeted. It *is* a pat response, it's one of the things I find I have to send often enough that I've got a canned response for it - I find that if I don't do things like that then after a while what gets sent becomes so brief that unless you're really familiar with what's being said it can be hard to understand the feedback. It's tricky though because as you say it's fairly obvious it's a form letter. > BTW, I don't see the patch applied anywhere (and I have no reply - > either manual or automated/patchwork - other than the above quote to say > it was). 20170309102819.bx5paxtffvmvmuov(a)sirena.org.uk, fix/core Didn't seem to have made -next today, guess it'll hit tomorrow.

1 0

Re: [alsa-devel] [PATCH 4/4] dts: bcm283x.dtsi: Add analog audio/headphone jack for bcm2835
by Michael Zoran 14 Mar '17

14 Mar '17

On Tue, 2017-03-14 at 11:07 +0100, Stefan Wahren wrote: > Am 14.03.2017 um 09:47 schrieb Michael Zoran: > > On Tue, 2017-03-14 at 09:05 +0100, Stefan Wahren wrote: > > > > > AFAIK the Raspberry Pi 3 uses pin 40 and 41. > > > > > > > I did all my testing on the RPI 3, so I'm almost certain it's > > correct. > > It matches with the schematic and it matches with what the > > downstream > > tree uses for the firmware based driver(dummy claiming of pins). > > The reduced schematic that i have name the pins as AUD_PWM0 and > AUD_PWM1, which says nothing about the actual pins. Do you have a > better > document? > > In the firmware dt-blob.dts [1] the Raspberry Pi 3 uses pin 40 and > 41. > > [1] - https://github.com/raspberrypi/firmware/blob/master/extra/dt-bl > ob.dts Here are the references I used: https://www.raspberrypi.org/documentation/hardware/raspberrypi/schemati cs/Raspberry-Pi-Rev-2.0-Model-AB-Schematics.pdf https://github.com/raspberrypi/linux/blob/rpi-4.9.y/arch/arm/boot/dts/o verlays/pwm-2chan-overlay.dts But looking at the downstream specific RPI 3 device tree, maybe it should be 40,41 instead of 40,45. The source code and tree I submitted does work. But I agree this is something that needs to be fixed.

1 0

Re: [alsa-devel] [PATCH 3/4] sound: arm: Add build system for bcm2835-analog-audio driver.
by Michael Zoran 14 Mar '17

14 Mar '17

On Tue, 2017-03-14 at 10:13 +0100, Stefan Wahren wrote: > Am 14.03.2017 um 09:58 schrieb Michael Zoran: > > On Tue, 2017-03-14 at 15:15 +0800, kbuild test robot wrote: > > > Hi Michael, > > > > > > [auto build test ERROR on sound/for-next] > > > [also build test ERROR on v4.11-rc2 next-20170310] > > > [if your patch is applied to the wrong git tree, please drop us a > > > note to help improve the system] > > > > > > url: https://github.com/0day-ci/linux/commits/Michael-Zoran/dt > > > -bin > > > dings-Document-the-dmas-and-dma-names-properties-for-bcm2835- > > > analog- > > > audio/20170314-091035 > > > base: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sou > > > nd.g > > > it for-next > > > config: arm64-allmodconfig (attached as .config) > > > compiler: aarch64-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 > > > reproduce: > > > wget https://raw.githubusercontent.com/01org/lkp-tests/ma > > > ster > > > /sbin/make.cross -O ~/bin/make.cross > > > chmod +x ~/bin/make.cross > > > # save the attached .config to linux build tree > > > make.cross ARCH=arm64 > > > > > > All errors (new ones prefixed by >>): > > > > > > /tmp/ccNU8kQ5.s: Assembler messages: > > > > > /tmp/ccNU8kQ5.s:1411: Error: unknown mnemonic `ldmia' -- > > > > > `ldmia > > > > > x26,{r0,r1,r2,r3}' > > > > > /tmp/ccNU8kQ5.s:1412: Error: unknown mnemonic `stmia' -- > > > > > `stmia > > > > > x19!,{r0,r1,r2,r3}' > > > > > /tmp/ccNU8kQ5.s:1618: Error: unknown mnemonic `ldmia' -- > > > > > `ldmia > > > > > x19!,{r0,r1,r2,r3}' > > > > > /tmp/ccNU8kQ5.s:1619: Error: unknown mnemonic `stmia' -- > > > > > `stmia > > > > > x26,{r0,r1,r2,r3}' > > > > > > --- > > > 0-DAY kernel test infrastructure Open Source > > > Technology Center > > > https://lists.01.org/pipermail/kbuild-all Intel > > > Corporation > > > > I'm a bit confused here. Why I'm I seeing assembly instruction > > build > > errors when the mail points to a DT binding document change? > > > > > > The email subject doesn't refer to a DT binding. > > I assume it's your change to enable ARM64 for all SND_ARM drivers, > which > aren't aware of that. OK, that makes sense now. Should I just submit another change adding a depends on ARM to the 1 or 2 other drivers in the directory?

1 0

Re: [alsa-devel] [PATCH 3/4] sound: arm: Add build system for bcm2835-analog-audio driver.
by Michael Zoran 14 Mar '17

14 Mar '17

On Tue, 2017-03-14 at 15:15 +0800, kbuild test robot wrote: > Hi Michael, > > [auto build test ERROR on sound/for-next] > [also build test ERROR on v4.11-rc2 next-20170310] > [if your patch is applied to the wrong git tree, please drop us a > note to help improve the system] > > url: https://github.com/0day-ci/linux/commits/Michael-Zoran/dt-bin > dings-Document-the-dmas-and-dma-names-properties-for-bcm2835-analog- > audio/20170314-091035 > base: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.g > it for-next > config: arm64-allmodconfig (attached as .config) > compiler: aarch64-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 > reproduce: > wget https://raw.githubusercontent.com/01org/lkp-tests/master > /sbin/make.cross -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=arm64 > > All errors (new ones prefixed by >>): > > /tmp/ccNU8kQ5.s: Assembler messages: > > > /tmp/ccNU8kQ5.s:1411: Error: unknown mnemonic `ldmia' -- `ldmia > > > x26,{r0,r1,r2,r3}' > > > /tmp/ccNU8kQ5.s:1412: Error: unknown mnemonic `stmia' -- `stmia > > > x19!,{r0,r1,r2,r3}' > > > /tmp/ccNU8kQ5.s:1618: Error: unknown mnemonic `ldmia' -- `ldmia > > > x19!,{r0,r1,r2,r3}' > > > /tmp/ccNU8kQ5.s:1619: Error: unknown mnemonic `stmia' -- `stmia > > > x26,{r0,r1,r2,r3}' > > --- > 0-DAY kernel test infrastructure Open Source > Technology Center > https://lists.01.org/pipermail/kbuild-all Intel > Corporation I'm a bit confused here. Why I'm I seeing assembly instruction build errors when the mail points to a DT binding document change?

1 0

Re: [alsa-devel] [PATCH 4/4] dts: bcm283x.dtsi: Add analog audio/headphone jack for bcm2835
by Michael Zoran 14 Mar '17

14 Mar '17

On Tue, 2017-03-14 at 09:05 +0100, Stefan Wahren wrote: > AFAIK the Raspberry Pi 3 uses pin 40 and 41. > I did all my testing on the RPI 3, so I'm almost certain it's correct. It matches with the schematic and it matches with what the downstream tree uses for the firmware based driver(dummy claiming of pins).

1 0

Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Kuninori Morimoto 14 Mar '17

14 Mar '17

Hi Brian > > The crasher was snd_dmaengine_pcm_register's platform ? > > No, actually it wasn't. It was spi2.0, which was a dummy, from > snd_soc_dummy_probe(). But somehow snd_soc_platform_drv_pcm_new() got > called for it... Hmm.. I couldn't reproduce your issue on my system with dummy platform. And your log is indicating something is happening which shouldn't happen. Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 0/5] MinnowBoard MAX support for DA7212
by Pierre-Louis Bossart 14 Mar '17

14 Mar '17

This patch set enables the Dialog ARD-Audio-Da7212 board connected to MinnowBoardMAX. The files for SSDT update are provided on github [1] as well as UCM files [2]. Headset capture and playback were tested and levels adjusted with the loopback dongle. The on-board mic can only be tested with a hardware mod. As an added bonus the machine driver to expose all I2S signals on the LSE connector was rebased to the latest kernel, it is useful to figure out what signals are generated by the Intel firmware/hardware. [1] https://github.com/plbossart/MinnowBoardMaxFirmware/tree/master/X64 [2] https://github.com/plbossart/UCM/tree/master/bytcht-da7213 Pierre-Louis Bossart (5): ASoC: da7213: add ACPI support ASoC: Intel: add machine driver for BYT/CHT + DA7213 ASoC: Intel: Atom: enable BYT/CHT+DA7213 machine driver ASoC: Intel: boards: add card for MinnowBoardMax/Up I2S access ASoC: Intel: Enable bytcht_nocodec machine driver sound/soc/codecs/da7213.c | 13 ++ sound/soc/intel/Kconfig | 24 +++ sound/soc/intel/atom/sst/sst_acpi.c | 25 ++- sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bytcht_da7213.c | 283 ++++++++++++++++++++++++++++++++ sound/soc/intel/boards/bytcht_nocodec.c | 208 +++++++++++++++++++++++ 6 files changed, 556 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/boards/bytcht_da7213.c create mode 100644 sound/soc/intel/boards/bytcht_nocodec.c -- 2.7.4

4 14

[alsa-devel] [PATCH] ASoC: rsnd: fix sound route path when using SRC6/SRC9
by Kuninori Morimoto 13 Mar '17

13 Mar '17

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> This patch fixes the problem that the missing value of the route path setting table and incorrect values are set in the CMD_ROUTE_SELECT register. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> [Kuninori: shared data on MIX and non-MIX case] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/cmd.c | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index abb5eaa..7d92a24 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -31,23 +31,24 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); struct device *dev = rsnd_priv_to_dev(priv); u32 data; + u32 path[] = { + [1] = 1 << 0, + [5] = 1 << 8, + [6] = 1 << 12, + [9] = 1 << 15, + }; if (!mix && !dvc) return 0; + if (ARRAY_SIZE(path) < rsnd_mod_id(mod) + 1) + return -ENXIO; + if (mix) { struct rsnd_dai *rdai; struct rsnd_mod *src; struct rsnd_dai_stream *tio; int i; - u32 path[] = { - [0] = 0, - [1] = 1 << 0, - [2] = 0, - [3] = 0, - [4] = 0, - [5] = 1 << 8 - }; /* * it is assuming that integrater is well understanding about @@ -70,16 +71,19 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, } else { struct rsnd_mod *src = rsnd_io_to_mod_src(io); - u32 path[] = { - [0] = 0x30000, - [1] = 0x30001, - [2] = 0x40000, - [3] = 0x10000, - [4] = 0x20000, - [5] = 0x40100 + u8 cmd_case[] = { + [0] = 0x3, + [1] = 0x3, + [2] = 0x4, + [3] = 0x1, + [4] = 0x2, + [5] = 0x4, + [6] = 0x1, + [9] = 0x2, }; - data = path[rsnd_mod_id(src)]; + data = path[rsnd_mod_id(src)] | + cmd_case[rsnd_mod_id(src)] << 16; } dev_dbg(dev, "ctu/mix path = 0x%08x", data); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: code cleanup for pin fixup limitation
by Takashi Sakamoto 13 Mar '17

13 Mar '17

As a commit 4cd9899f0d16 ("ASoC: Intel: Skylake: Add multiple pin formats") describes, 'fixups is applied to pin 0 only'. On the other hand, the commit left some codes as what they were. This might confuses readers. This commit fixes the issue. This doesn't change driver behaviour at all. Fixes: 4cd9899f0d16 ("ASoC: Intel: Skylake: Add multiple pin formats") Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> --- sound/soc/intel/skylake/skl-topology.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index f38f8e6..8651359 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -310,13 +310,11 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, multiplier = 5; mcfg->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) * - (mcfg->in_fmt->channels) * - (mcfg->in_fmt->bit_depth >> 3) * + in_fmt->channels * (in_fmt->bit_depth >> 3) * multiplier; - mcfg->obs = DIV_ROUND_UP(mcfg->out_fmt->s_freq, 1000) * - (mcfg->out_fmt->channels) * - (mcfg->out_fmt->bit_depth >> 3) * + mcfg->obs = DIV_ROUND_UP(out_fmt->s_freq, 1000) * + out_fmt->channels * (out_fmt->bit_depth >> 3) * multiplier; } -- 2.9.3

3 2

[alsa-devel] [PATCH 1/5] ASoC: cs35l35: Add missing return in probe
by Charles Keepax 13 Mar '17

13 Mar '17

A return statement is missing just before the error paths at the end of probe. This causes us to fall straight into the error path and disable the supplies and re-enable reset, as these are only controlled during probe this causes the part to no longer function. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 260ed42..48b45dc 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1509,6 +1509,8 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, goto err; } + return 0; + err: regulator_bulk_disable(cs35l35->num_supplies, cs35l35->supplies); -- 2.1.4

3 10

[alsa-devel] Applied "ASoC: cs35l35: Fix display revision id" to the asoc tree
by Mark Brown 13 Mar '17

13 Mar '17

The patch ASoC: cs35l35: Fix display revision id has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 82875163a8ef1e25477402c5ebb8f5beaea5e93e Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)ingics.com> Date: Sat, 11 Mar 2017 20:57:07 +0800 Subject: [PATCH] ASoC: cs35l35: Fix display revision id Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs35l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 7c4d74ec32cb..a9e45dea309e 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1463,7 +1463,7 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, } dev_info(dev, "Cirrus Logic CS35L35 (%x), Revision: %02X\n", - devid, ret & 0xFF); + devid, reg & 0xFF); /* Set the INT Masks for critical errors */ regmap_write(cs35l35->regmap, CS35L35_INT_MASK_1, -- 2.11.0

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[alsa-devel] Applied "ASoC: cs35l35: trivial fix to indentation" to the asoc tree
by Mark Brown 13 Mar '17

13 Mar '17

The patch ASoC: cs35l35: trivial fix to indentation has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 03ff570c6b45b27e233f5cb1fd04404374fd883f Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Sat, 11 Mar 2017 19:19:22 +0000 Subject: [PATCH] ASoC: cs35l35: trivial fix to indentation Remove extraneous tab to correct the nesting level indentation Detected by CoverityScan, CID#1416584 ("Nesting level does not match indentation") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs35l35.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index a9e45dea309e..a5da1d511a8a 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1375,7 +1375,8 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) cs35l35->supplies[i].supply = cs35l35_supplies[i]; - cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); + + cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); ret = devm_regulator_bulk_get(dev, cs35l35->num_supplies, cs35l35->supplies); -- 2.11.0

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[alsa-devel] [PATCH] ASoC: fix semicolon.cocci warnings
by kbuild test robot 13 Mar '17

13 Mar '17

sound/soc/codecs/cs35l35.c:706:2-3: Unneeded semicolon sound/soc/codecs/cs35l35.c:543:4-5: Unneeded semicolon sound/soc/codecs/cs35l35.c:553:4-5: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci CC: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- cs35l35.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -540,7 +540,7 @@ static int cs35l35_hw_params(struct snd_ default: dev_err(codec->dev, "ratio not supported\n"); return -EINVAL; - }; + } } else { /* Only certain ratios supported in I2S MASTER Mode */ switch (sp_sclks) { @@ -550,7 +550,7 @@ static int cs35l35_hw_params(struct snd_ default: dev_err(codec->dev, "ratio not supported\n"); return -EINVAL; - }; + } } ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL3, @@ -703,7 +703,7 @@ static int cs35l35_codec_set_sysclk(stru default: dev_err(codec->dev, "Invalid CLK Source\n"); return -EINVAL; - }; + } switch (freq) { case 5644800:

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