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[alsa-devel] [PATCH 0/4] ALSA-Riptide: Fine-tuning for seven function implementations
by SF Markus Elfring 17 Nov '17

17 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 17 Nov 2017 10:32:23 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (4): Adjust 13 function calls together with a variable assignment Delete an unnecessary variable initialisation in snd_riptide_proc_read() Reduce the scope for two variables in snd_riptide_proc_read() Use common error handling code in snd_riptide_create() sound/pci/riptide/riptide.c | 78 ++++++++++++++++++++++++++------------------- 1 file changed, 46 insertions(+), 32 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH] ASoC: Intel: Add defaults for SND_SST_ options and machine drivers
by Pierre-Louis Bossart 17 Nov '17

17 Nov '17

Add 'default m' when sensible, move common dependencies in if/endif block and clarify which options default to n - distro configurations should not enable legacy Baytrail stuff and NOCODEC (test only) Basic tests run with defconfig, oldconfig, olddefconfig, there should be no functionality change. Fixes: f6a118a800e3 ("ASoC: Intel: clarify Kconfig dependencies") Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 5 ++++ sound/soc/intel/boards/Kconfig | 60 ++++++++++++++++++++++++++---------------- 2 files changed, 43 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7b49d04e3c60..b40719396e9d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_INTEL_SST_TOPLEVEL tristate "Intel ASoC SST drivers" + default m depends on X86 || COMPILE_TEST select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON @@ -42,6 +43,7 @@ config SND_SOC_INTEL_SST_TOPLEVEL config SND_SOC_INTEL_HASWELL tristate "Intel ASoC SST driver for Haswell/Broadwell" + default m depends on SND_SOC_INTEL_SST_TOPLEVEL && SND_DMA_SGBUF depends on DMADEVICES select SND_SOC_INTEL_SST @@ -49,6 +51,7 @@ config SND_SOC_INTEL_HASWELL config SND_SOC_INTEL_BAYTRAIL tristate "Intel ASoC SST driver for Baytrail (legacy)" + default n depends on SND_SOC_INTEL_SST_TOPLEVEL depends on DMADEVICES select SND_SOC_INTEL_SST @@ -56,11 +59,13 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SST_ATOM_HIFI2_PLATFORM tristate "Intel ASoC SST driver for HiFi2 platforms (*field, *trail)" + default m depends on SND_SOC_INTEL_SST_TOPLEVEL && X86 select SND_SOC_COMPRESS config SND_SOC_INTEL_SKYLAKE tristate "Intel ASoC SST driver for SKL/BXT/KBL/GLK/CNL" + default m depends on SND_SOC_INTEL_SST_TOPLEVEL && PCI && ACPI select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..6069328c5c28 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -17,10 +17,12 @@ config SND_MFLD_MACHINE Say Y if you have such a device. If unsure select "N". +if SND_SOC_INTEL_HASWELL + config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" + default m depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell @@ -30,8 +32,8 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BDW_RT5677_MACH tristate "ASoC Audio driver for Intel Broadwell with RT5677 codec" + default m depends on X86_INTEL_LPSS && GPIOLIB && I2C - depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT5677 help This adds support for Intel Broadwell platform based boards with @@ -39,41 +41,50 @@ config SND_SOC_INTEL_BDW_RT5677_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + default m depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell Ultrabook platforms. Say Y if you have such a device. If unsure select "N". +endif + +if SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" + default n depends on X86_INTEL_LPSS && I2C depends on SND_SST_IPC_ACPI = n - depends on SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help This adds audio driver for Intel Baytrail platform based boards - with the MAX98090 audio codec. + with the MAX98090 audio codec. This driver is deprecated, use + SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH instead for better + functionality. config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" + default n depends on X86_INTEL_LPSS && I2C depends on SND_SST_IPC_ACPI = n - depends on SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. This driver is deprecated, use SND_SOC_INTEL_BYTCR_RT5640_MACH instead for better functionality. +endif + +if SND_SST_ATOM_HIFI2_PLATFORM + config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec" + default m depends on X86 && I2C && ACPI select SND_SOC_RT5640 - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR @@ -83,9 +94,9 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" + default m depends on X86 && I2C && ACPI select SND_SOC_RT5651 - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR @@ -95,9 +106,9 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + default m depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -107,9 +118,9 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" + default m depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -118,10 +129,10 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec" + default m depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_MAX98090 select SND_SOC_TS3A227E - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell @@ -130,9 +141,9 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with DA7212/7213 codec" + default m depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_DA7213 - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail & CherryTrail @@ -141,9 +152,9 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH config SND_SOC_INTEL_BYT_CHT_ES8316_MACH tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with ES8316 codec" + default m depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_ES8316 - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail & @@ -152,19 +163,23 @@ config SND_SOC_INTEL_BYT_CHT_ES8316_MACH config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" + default n depends on X86_INTEL_LPSS && I2C && ACPI - depends on SND_SST_ATOM_HIFI2_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for the MinnowBoard Max or Up boards and provides access to I2S signals on the Low-Speed - connector + connector. It is not intended to be enabled by distros by default. If unsure select "N". +endif + +if SND_SOC_INTEL_SKYLAKE + config SND_SOC_INTEL_SKL_RT286_MACH tristate "ASoC Audio driver for SKL with RT286 I2S mode" + default m depends on X86 && ACPI && I2C - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_RT286 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -176,8 +191,8 @@ config SND_SOC_INTEL_SKL_RT286_MACH config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH tristate "ASoC Audio driver for SKL with NAU88L25 and SSM4567 in I2S Mode" + default m depends on X86_INTEL_LPSS && I2C - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_NAU8825 select SND_SOC_SSM4567 select SND_SOC_DMIC @@ -190,8 +205,8 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH tristate "ASoC Audio driver for SKL with NAU88L25 and MAX98357A in I2S Mode" + default m depends on X86_INTEL_LPSS && I2C - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_NAU8825 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -204,8 +219,8 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "ASoC Audio driver for Broxton with DA7219 and MAX98357A in I2S Mode" + default m depends on X86 && ACPI && I2C - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -219,8 +234,8 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH config SND_SOC_INTEL_BXT_RT298_MACH tristate "ASoC Audio driver for Broxton with RT298 I2S mode" + default m depends on X86 && ACPI && I2C - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_RT298 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -233,9 +248,9 @@ config SND_SOC_INTEL_BXT_RT298_MACH config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH tristate "ASoC Audio driver for KBL with RT5663 and MAX98927 in I2S Mode" + default m depends on X86_INTEL_LPSS && I2C select SND_SOC_INTEL_SST - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 select SND_SOC_MAX98927 select SND_SOC_DMIC @@ -248,9 +263,9 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH tristate "ASoC Audio driver for KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + default m depends on X86_INTEL_LPSS && I2C && SPI select SND_SOC_INTEL_SST - depends on SND_SOC_INTEL_SKYLAKE select SND_SOC_RT5663 select SND_SOC_RT5514 select SND_SOC_RT5514_SPI @@ -261,5 +276,6 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH create an alsa sound card for RT5663 + RT5514 + MAX98927. Say Y if you have such a device. If unsure select "N". +endif endif -- 2.14.1

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[alsa-devel] [PATCH 0/2] ASoC: fsl_asrc: neatening
by Joe Perches 17 Nov '17

17 Nov '17

Joe Perches (2): ASoC: fsl_asrc: Fix line continuation format ASoC: fsl_asrc: constify some arrays sound/soc/fsl/fsl_asrc.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) -- 2.15.0

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Re: [alsa-devel] [PATCH RESEND 0/2] ASoC: Intel: kbl: update dmic fixup params
by Vinod Koul 17 Nov '17

17 Nov '17

On Wed, Nov 15, 2017 at 03:37:41PM +0530, Naveen Manohar wrote: > Hi Mark, > > These are resend of an already acked patch series. > I posted it a week ago but there has been update on the base branch. > I want to resend these. Kindly review and merge. > > Update dmic fixup params to always expose QUAD channel capture device, > enabling User-space to control channel count. > Assign min and max channel only based on input hw_params. > And removes choosing the channel count based on pdata. Acked-By: Vinod Koul <vinod.koul(a)intel.com> > Naveen Manohar (2): > ASoC: Intel: kbl: Remove option of choosing CH count based on pdata > ASoC: Intel: kbl: update dmic fixup params > > sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++++++------------------- > 1 file changed, 14 insertions(+), 30 deletions(-) > > -- > 1.9.1 > -- ~Vinod

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[alsa-devel] [PATCH] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 17 Nov '17

17 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later from the first command. Otherwise, Playback volume will be 0dB. This patch adds dummy write when Playback Start timing. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 59 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 59 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..836c111 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct delayed_work dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,64 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work.work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned long delay; + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delayed_work call from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. Here Let's call it after 6 LR + */ + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + break; + default: + return 0; + } + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + delay = 6000000 / priv->rate; + priv->component = &codec->component; + schedule_delayed_work(&priv->dummy_write_work, + usecs_to_jiffies(delay)); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +649,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_DELAYED_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

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[alsa-devel] [PATCH 0/2] ALSA: pcxhr: Fine-tuning for three function implementations
by SF Markus Elfring 16 Nov '17

16 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 16 Nov 2017 22:20:02 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust six function calls together with a variable assignment Use common error handling code in pcxhr_probe() sound/pci/pcxhr/pcxhr.c | 82 ++++++++++++++++++++++++++----------------------- 1 file changed, 44 insertions(+), 38 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 1/2] ASoC: da7218: fix fix child-node lookup
by Johan Hovold 16 Nov '17

16 Nov '17

Fix child-node lookup during probe, which ended up searching the whole device tree depth-first starting at the parent rather than just matching on its children. To make things worse, the parent codec node was also prematurely freed. Fixes: 4d50934abd22 ("ASoC: da7218: Add da7218 codec driver") Cc: stable <stable(a)vger.kernel.org> # 4.5 Cc: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/soc/codecs/da7218.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index b2d42ec1dcd9..56564ce90cb6 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2520,7 +2520,7 @@ static struct da7218_pdata *da7218_of_to_pdata(struct snd_soc_codec *codec) } if (da7218->dev_id == DA7218_DEV_ID) { - hpldet_np = of_find_node_by_name(np, "da7218_hpldet"); + hpldet_np = of_get_child_by_name(np, "da7218_hpldet"); if (!hpldet_np) return pdata; -- 2.15.0

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[alsa-devel] [PATCH] ASoC: rt5514: The DSP clock can be calibrated by the other clock source
by Oder Chiou 16 Nov '17

16 Nov '17

Add the option for the DSP clock that can be calibrated by the other clock source. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5514.h | 2 + sound/soc/codecs/rt5514-spi.c | 1 + sound/soc/codecs/rt5514.c | 85 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 5 ++- 4 files changed, 92 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5514.h b/include/sound/rt5514.h index ef18494..64d027d 100644 --- a/include/sound/rt5514.h +++ b/include/sound/rt5514.h @@ -14,6 +14,8 @@ struct rt5514_platform_data { unsigned int dmic_init_delay; + const char *dsp_calib_clk_name; + unsigned int dsp_calib_clk_rate; }; #endif diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 12f2ecf..b90d6d5 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -370,6 +370,7 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) return true; } +EXPORT_SYMBOL_GPL(rt5514_spi_burst_read); /** * rt5514_spi_burst_write - Write data to SPI by rt5514 address. diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2a5b5d7..61ccbc6 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -295,6 +295,33 @@ static int rt5514_dsp_voice_wake_up_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt5514_calibration(struct rt5514_priv *rt5514, bool on) +{ + if (on) { + regmap_write(rt5514->regmap, RT5514_ANA_CTRL_PLL3, 0x0000000a); + regmap_update_bits(rt5514->regmap, RT5514_PLL_SOURCE_CTRL, 0xf, + 0xa); + regmap_update_bits(rt5514->regmap, RT5514_PWR_ANA1, 0x301, + 0x301); + regmap_write(rt5514->regmap, RT5514_PLL3_CALIB_CTRL4, + 0x80000000 | rt5514->pll3_cal_value); + regmap_write(rt5514->regmap, RT5514_PLL3_CALIB_CTRL1, + 0x8bb80800); + regmap_update_bits(rt5514->regmap, RT5514_PLL3_CALIB_CTRL5, + 0xc0000000, 0x80000000); + regmap_update_bits(rt5514->regmap, RT5514_PLL3_CALIB_CTRL5, + 0xc0000000, 0xc0000000); + } else { + regmap_update_bits(rt5514->regmap, RT5514_PLL3_CALIB_CTRL5, + 0xc0000000, 0x40000000); + regmap_update_bits(rt5514->regmap, RT5514_PWR_ANA1, 0x301, 0); + regmap_update_bits(rt5514->regmap, RT5514_PLL_SOURCE_CTRL, 0xf, + 0x4); + } + + return 0; +} + static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -302,6 +329,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); struct snd_soc_codec *codec = rt5514->codec; const struct firmware *fw = NULL; + u8 buf[8]; if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) return 0; @@ -310,6 +338,35 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, rt5514->dsp_enabled = ucontrol->value.integer.value[0]; if (rt5514->dsp_enabled) { + if (rt5514->pdata.dsp_calib_clk_name && + !IS_ERR(rt5514->dsp_calib_clk)) { + if (clk_set_rate(rt5514->dsp_calib_clk, + rt5514->pdata.dsp_calib_clk_rate)) + dev_err(codec->dev, + "Can't set rate for mclk"); + + if (clk_prepare_enable(rt5514->dsp_calib_clk)) + dev_err(codec->dev, + "Can't enable dsp_calib_clk"); + + rt5514_calibration(rt5514, true); + + msleep(20); +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_read(RT5514_PLL3_CALIB_CTRL6 | + RT5514_DSP_MAPPING, + (u8 *)&buf, sizeof(buf)); +#else + dev_err(codec->dev, "There is no SPI driver for" + " loading the firmware\n"); +#endif + rt5514->pll3_cal_value = buf[0] | buf[1] << 8 | + buf[2] << 16 | buf[3] << 24; + + rt5514_calibration(rt5514, false); + clk_disable_unprepare(rt5514->dsp_calib_clk); + } + rt5514_enable_dsp_prepare(rt5514); request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); @@ -341,6 +398,20 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); + + if (rt5514->pdata.dsp_calib_clk_name && + !IS_ERR(rt5514->dsp_calib_clk)) { + msleep(20); + + regmap_write(rt5514->i2c_regmap, 0x1800211c, + rt5514->pll3_cal_value); + regmap_write(rt5514->i2c_regmap, 0x18002124, + 0x00220012); + regmap_write(rt5514->i2c_regmap, 0x18002124, + 0x80220042); + regmap_write(rt5514->i2c_regmap, 0x18002124, + 0xe0220042); + } } else { regmap_multi_reg_write(rt5514->i2c_regmap, rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); @@ -1024,12 +1095,22 @@ static int rt5514_set_bias_level(struct snd_soc_codec *codec, static int rt5514_probe(struct snd_soc_codec *codec) { struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + struct platform_device *pdev = container_of(codec->dev, + struct platform_device, dev); rt5514->mclk = devm_clk_get(codec->dev, "mclk"); if (PTR_ERR(rt5514->mclk) == -EPROBE_DEFER) return -EPROBE_DEFER; + if (rt5514->pdata.dsp_calib_clk_name) { + rt5514->dsp_calib_clk = devm_clk_get(&pdev->dev, + rt5514->pdata.dsp_calib_clk_name); + if (PTR_ERR(rt5514->dsp_calib_clk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + rt5514->codec = codec; + rt5514->pll3_cal_value = 0x0078b000; return 0; } @@ -1147,6 +1228,10 @@ static int rt5514_parse_dp(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", &rt5514->pdata.dmic_init_delay); + device_property_read_string(dev, "realtek,dsp-calib-clk-name", + &rt5514->pdata.dsp_calib_clk_name); + device_property_read_u32(dev, "realtek,dsp-calib-clk-rate", + &rt5514->pdata.dsp_calib_clk_rate); return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 2dc40e6..f0f3400 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -34,7 +34,9 @@ #define RT5514_CLK_CTRL1 0x2104 #define RT5514_CLK_CTRL2 0x2108 #define RT5514_PLL3_CALIB_CTRL1 0x2110 +#define RT5514_PLL3_CALIB_CTRL4 0x2120 #define RT5514_PLL3_CALIB_CTRL5 0x2124 +#define RT5514_PLL3_CALIB_CTRL6 0x2128 #define RT5514_DELAY_BUF_CTRL1 0x2140 #define RT5514_DELAY_BUF_CTRL3 0x2148 #define RT5514_ASRC_IN_CTRL1 0x2180 @@ -272,7 +274,7 @@ struct rt5514_priv { struct rt5514_platform_data pdata; struct snd_soc_codec *codec; struct regmap *i2c_regmap, *regmap; - struct clk *mclk; + struct clk *mclk, *dsp_calib_clk; int sysclk; int sysclk_src; int lrck; @@ -281,6 +283,7 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; + unsigned int pll3_cal_value; }; #endif /* __RT5514_H__ */ -- 2.7.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: nau8540: reset state machine for channel phase sync
by John Hsu 16 Nov '17

16 Nov '17

The four channel ADCs in NAU85L40 have difference control registers, it is hard to synchronous these four channels without correct sequence. The phase difference will not be a constant and not to conjecture easily. It may be 2.55 degree, or more ,or less. Intended to prevent phase difference of channels, the solution as follows: (1)Channel_Sync need to be enabled. (2)Do soft reset without affecting register when recording done. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8540.c | 23 ++++++++++++++++++++--- sound/soc/codecs/nau8540.h | 1 + 2 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 4d18158..2efa431 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -233,6 +233,19 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new digital_ch1_mux = SOC_DAPM_ENUM("Digital CH1 Select", digital_ch1_enum); +static int aiftx_power_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + regmap_write(nau8540->regmap, NAU8540_REG_RST, 0x0001); + regmap_write(nau8540->regmap, NAU8540_REG_RST, 0x0000); + } + return 0; +} + static const struct snd_soc_dapm_widget nau8540_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("MICBIAS2", NAU8540_REG_MIC_BIAS, 11, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", NAU8540_REG_MIC_BIAS, 10, 0, NULL, 0), @@ -270,7 +283,8 @@ static const struct snd_soc_dapm_widget nau8540_dapm_widgets[] = { SND_SOC_DAPM_MUX("Digital CH1 Mux", SND_SOC_NOPM, 0, 0, &digital_ch1_mux), - SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0, + aiftx_power_control, SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { @@ -717,9 +731,12 @@ static void nau8540_init_regs(struct nau8540 *nau8540) regmap_update_bits(regmap, NAU8540_REG_CLOCK_CTRL, NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN, NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN); - /* ADC OSR selection, CLK_ADC = Fs * OSR */ + /* ADC OSR selection, CLK_ADC = Fs * OSR; + * Channel time alignment enable. + */ regmap_update_bits(regmap, NAU8540_REG_ADC_SAMPLE_RATE, - NAU8540_ADC_OSR_MASK, NAU8540_ADC_OSR_64); + NAU8540_CH_SYNC | NAU8540_ADC_OSR_MASK, + NAU8540_CH_SYNC | NAU8540_ADC_OSR_64); } static int __maybe_unused nau8540_suspend(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/nau8540.h b/sound/soc/codecs/nau8540.h index ab2a132..5e0dd18 100644 --- a/sound/soc/codecs/nau8540.h +++ b/sound/soc/codecs/nau8540.h @@ -170,6 +170,7 @@ #define NAU8540_TDM_TX_MASK 0xf /* ADC_SAMPLE_RATE (0x3A) */ +#define NAU8540_CH_SYNC (0x1 << 14) #define NAU8540_ADC_OSR_MASK 0x3 #define NAU8540_ADC_OSR_256 0x3 #define NAU8540_ADC_OSR_128 0x2 -- 2.6.4

2 5

[alsa-devel] [PATCH 1/2] ASoC: nau8824: move key irq after jd done
by John Hsu 16 Nov '17

16 Nov '17

It is possible to get the fake key press interruption when the codec do jack detection. We think it's proper to move the key interruption configuration after jack detection done. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 0240759..b7b63ac 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -843,6 +843,11 @@ static void nau8824_jdet_work(struct work_struct *work) event_mask |= SND_JACK_HEADSET; snd_soc_jack_report(nau8824->jack, event, event_mask); + /* Enable short key press and release interruption. */ + regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING, + NAU8824_IRQ_KEY_RELEASE_DIS | + NAU8824_IRQ_KEY_SHORT_PRESS_DIS, 0); + nau8824_sema_release(nau8824); } @@ -850,13 +855,12 @@ static void nau8824_setup_auto_irq(struct nau8824 *nau8824) { struct regmap *regmap = nau8824->regmap; - /* Enable jack ejection, short key press and release interruption. */ + /* Enable jack ejection interruption. */ regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING_1, NAU8824_IRQ_INSERT_EN | NAU8824_IRQ_EJECT_EN, NAU8824_IRQ_EJECT_EN); regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING, - NAU8824_IRQ_EJECT_DIS | NAU8824_IRQ_KEY_RELEASE_DIS | - NAU8824_IRQ_KEY_SHORT_PRESS_DIS, 0); + NAU8824_IRQ_EJECT_DIS, 0); /* Enable internal VCO needed for interruptions */ nau8824_config_sysclk(nau8824, NAU8824_CLK_INTERNAL, 0); regmap_update_bits(regmap, NAU8824_REG_ENA_CTRL, -- 2.6.4

2 3

[alsa-devel] [PATCH] ASoC: intel: initialize return value properly
by Arnd Bergmann 16 Nov '17

16 Nov '17

When CONFIG_SND_SOC_HDAC_HDMI is disabled, we can run into an uninitialized variable: sound/soc/intel/skylake/skl.c: In function 'skl_resume': sound/soc/intel/skylake/skl.c:326:6: error: 'ret' may be used uninitialized in this function [-Werror=maybe-uninitialized] I have run into this on today's linux-next kernel, but it appears that this is an older problem that was just hard to trigger with randconfig builds as CONFIG_SND_SOC_HDAC_HDMI would in effect be impossible to disable when having SND_SOC_INTEL_SKYLAKE enabled. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/skylake/skl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 31d8634e8aa1..acb0ab470ca6 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -355,6 +355,7 @@ static int skl_resume(struct device *dev) if (ebus->cmd_dma_state) snd_hdac_bus_init_cmd_io(&ebus->bus); + ret = 0; } else { ret = _skl_resume(ebus); -- 2.9.0

3 2

[alsa-devel] [PATCH 0/3] Fixes for AMD Stoney ACP audio
by Alex Deucher 16 Nov '17

16 Nov '17

This patch set is just a couple fixes for the Audio CoProcessor (ACP) on AMD Stoney platforms and a small general fix for the rt5645 codec driver. The entire patch set can also be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=sound-for-next-stoney Thanks! Alex Akshu Agrawal (2): ASoC: AMD: Make the driver name consistent across files ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2 Vijendar Mukunda (1): ASoC: amd: Report accurate hw_ptr during dma sound/soc/amd/Makefile | 4 +-- sound/soc/amd/acp-pcm-dma.c | 77 ++++++++++++++++++++++++++++----------------- sound/soc/amd/acp.h | 10 ++++++ sound/soc/codecs/rt5645.c | 12 +++++++ 4 files changed, 73 insertions(+), 30 deletions(-) -- 2.5.5

4 13

[alsa-devel] [PATCH] ASoC: Intel: kbl: Modify Headset map for Multi-Playback
by Naveen Manohar 16 Nov '17

16 Nov '17

Current topology for Headset resulted in unwanted pop noise, while switching from spk->hs at the start of Headset Playback. Modified topology to fix the same. This modification also requires headset route to be updated. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bc..6dcad0a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164..271ae3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, -- 1.9.1

4 3

[alsa-devel] [PATCH] ASoC: nau8540: improve FLL performance
by John Hsu 16 Nov '17

16 Nov '17

Add these parameters to improve the FLL performance. The comments show as follows: (1)ICTRL_LATCH: FLL DSP speed capability control When FLL running at high frequency with long decimal number, DSP needs to operate at high speed. FLL DSP can optimize between performance and power consumption by ICTRL_LATCH.(111 has highest power consumption.) The default setting can be used to reduce power. (2)CUTOFF500: loop filter cutoff frequency at 500Khz It will give the best FLL performance but highest power consumption to enable the cutoff frequency. FLL Loop Filter enable to reduce FLL output noise, especially,(DCO frequency)/(FLL input reference frequency) is not a integer. (3)GAIN_ERR: FLL gain error correction threshold setting The threshold is comparison between DCO and target frequency. The value 1111 has the most sensitive threshold, that is, 1111 can have the most accurate DCO to target frequency. However, the gain error setting conditionally and inversely depends on FLL input reference clock rate. Higher FLL reference input frequency can only set lower gain error, such as 0000 for input reference from MCLK=12.288Mhz. On the other side, if FLL reference input is from Frame Sync, 48KHz, higher error gain can apply such as 1111. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8540.c | 21 ++++++++++++++------- sound/soc/codecs/nau8540.h | 5 +++++ 2 files changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 9565f9a..b08fb7e 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -615,7 +615,8 @@ static void nau8540_fll_apply(struct regmap *regmap, NAU8540_CLK_SRC_MASK | NAU8540_CLK_MCLK_SRC_MASK, NAU8540_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(regmap, NAU8540_REG_FLL1, - NAU8540_FLL_RATIO_MASK, fll_param->ratio); + NAU8540_FLL_RATIO_MASK | NAU8540_ICTRL_LATCH_MASK, + fll_param->ratio | (0x6 << NAU8540_ICTRL_LATCH_SFT)); /* FLL 16-bit fractional input */ regmap_write(regmap, NAU8540_REG_FLL2, fll_param->fll_frac); /* FLL 10-bit integer input */ @@ -636,13 +637,14 @@ static void nau8540_fll_apply(struct regmap *regmap, NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | NAU8540_FLL_FTR_SW_FILTER); regmap_update_bits(regmap, NAU8540_REG_FLL6, - NAU8540_SDM_EN, NAU8540_SDM_EN); + NAU8540_SDM_EN | NAU8540_CUTOFF500, + NAU8540_SDM_EN | NAU8540_CUTOFF500); } else { regmap_update_bits(regmap, NAU8540_REG_FLL5, NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | NAU8540_FLL_FTR_SW_MASK, NAU8540_FLL_FTR_SW_ACCU); - regmap_update_bits(regmap, - NAU8540_REG_FLL6, NAU8540_SDM_EN, 0); + regmap_update_bits(regmap, NAU8540_REG_FLL6, + NAU8540_SDM_EN | NAU8540_CUTOFF500, 0); } } @@ -657,17 +659,22 @@ static int nau8540_set_pll(struct snd_soc_codec *codec, int pll_id, int source, switch (pll_id) { case NAU8540_CLK_FLL_MCLK: regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_MCLK); + NAU8540_FLL_CLK_SRC_MASK | NAU8540_GAIN_ERR_MASK, + NAU8540_FLL_CLK_SRC_MCLK | 0); break; case NAU8540_CLK_FLL_BLK: regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_BLK); + NAU8540_FLL_CLK_SRC_MASK | NAU8540_GAIN_ERR_MASK, + NAU8540_FLL_CLK_SRC_BLK | + (0xf << NAU8540_GAIN_ERR_SFT)); break; case NAU8540_CLK_FLL_FS: regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_FS); + NAU8540_FLL_CLK_SRC_MASK | NAU8540_GAIN_ERR_MASK, + NAU8540_FLL_CLK_SRC_FS | + (0xf << NAU8540_GAIN_ERR_SFT)); break; default: diff --git a/sound/soc/codecs/nau8540.h b/sound/soc/codecs/nau8540.h index dceb04b..732b490 100644 --- a/sound/soc/codecs/nau8540.h +++ b/sound/soc/codecs/nau8540.h @@ -100,9 +100,13 @@ #define NAU8540_CLK_MCLK_SRC_MASK 0xf /* FLL1 (0x04) */ +#define NAU8540_ICTRL_LATCH_SFT 10 +#define NAU8540_ICTRL_LATCH_MASK (0x7 << NAU8540_ICTRL_LATCH_SFT) #define NAU8540_FLL_RATIO_MASK 0x7f /* FLL3 (0x06) */ +#define NAU8540_GAIN_ERR_SFT 12 +#define NAU8540_GAIN_ERR_MASK (0xf << NAU8540_GAIN_ERR_SFT) #define NAU8540_FLL_CLK_SRC_SFT 10 #define NAU8540_FLL_CLK_SRC_MASK (0x3 << NAU8540_FLL_CLK_SRC_SFT) #define NAU8540_FLL_CLK_SRC_MCLK (0 << NAU8540_FLL_CLK_SRC_SFT) @@ -127,6 +131,7 @@ /* FLL6 (0x9) */ #define NAU8540_DCO_EN (0x1 << 15) #define NAU8540_SDM_EN (0x1 << 14) +#define NAU8540_CUTOFF500 (0x1 << 13) /* PCM_CTRL0 (0x10) */ #define NAU8540_I2S_BP_SFT 7 -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: nau8824: change FVCO maximum threshold
by John Hsu 16 Nov '17

16 Nov '17

Change the maximum of FDCO which remains between 90MHz-100MHz. FDCO must be within the 90MHz-100MHz or the FFL cannot be guaranteed across the full range of operation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 8a9a993..088e0ce 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -43,7 +43,7 @@ static bool nau8824_is_jack_inserted(struct nau8824 *nau8824); /* the parameter threshold of FLL */ #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 124000000 +#define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 /* scaling for mclk from sysclk_src output */ -- 2.6.4

2 1

[alsa-devel] [PATCH 0/2] ALSA-miXart: Fine-tuning for seven function implementations
by SF Markus Elfring 16 Nov '17

16 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 16 Nov 2017 16:27:32 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust 23 function calls together with a variable assignment Use common error handling code in snd_mixart_probe() sound/pci/mixart/mixart.c | 88 ++++++++++++++++++++++------------------- sound/pci/mixart/mixart_hwdep.c | 17 +++++--- sound/pci/mixart/mixart_mixer.c | 44 +++++++++++++++------ 3 files changed, 93 insertions(+), 56 deletions(-) -- 2.15.0

1 2

[alsa-devel] [PATCH 0/2] ASoC: Intel: kbl: update dmic fixup params
by Naveen Manohar 16 Nov '17

16 Nov '17

Update dmic fixup params to always expose QUAD channel capture device, enabling User-space to control channel count. Assign min and max channel only based on input hw_params. And removes choosing the channel count based on pdata. Naveen Manohar (2): ASoC: Intel: kbl: Remove option of choosing CH count based on pdata ASoC: Intel: kbl: update dmic fixup params sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++++++------------------- 1 file changed, 14 insertions(+), 30 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: rcar: revert IOMMU support so far
by Kuninori Morimoto 16 Nov '17

16 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4821d914fe74 ("ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU") had supported IOMMU, but it breaks normal sound "recorde" and both PulseAudio's "playback/recorde". The sound will be noisy. That commit was using dma_sync_single_for_xxx(), and driver should make sure memory is protected during CPU or Device are using it. But if driver returns current "residue" data size correctly on pointer function, player/recorder will access to protected memory. IOMMU feature should be supported, but I don't know how to handle it without memory cache problem at this point. Thus, this patch simply revert it to avoid current noisy sound. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Tested-by: Ryo Kodama <ryo.kodama.vz(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 4 +-- sound/soc/sh/rcar/dma.c | 86 ++++-------------------------------------------- 2 files changed, 8 insertions(+), 82 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8cd900d..ebdfa34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1329,8 +1329,8 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fd557ab..4d750bdf 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -26,10 +26,7 @@ struct rsnd_dmaen { struct dma_chan *chan; dma_cookie_t cookie; - dma_addr_t dma_buf; unsigned int dma_len; - unsigned int dma_period; - unsigned int dma_cnt; }; struct rsnd_dmapp { @@ -71,38 +68,10 @@ struct rsnd_dma_ctrl { /* * Audio DMAC */ -#define rsnd_dmaen_sync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 1) -#define rsnd_dmaen_unsync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 0) -static void __rsnd_dmaen_sync(struct rsnd_dmaen *dmaen, struct rsnd_dai_stream *io, - int i, int sync) -{ - struct device *dev = dmaen->chan->device->dev; - enum dma_data_direction dir; - int is_play = rsnd_io_is_play(io); - dma_addr_t buf; - int len, max; - size_t period; - - len = dmaen->dma_len; - period = dmaen->dma_period; - max = len / period; - i = i % max; - buf = dmaen->dma_buf + (period * i); - - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - - if (sync) - dma_sync_single_for_device(dev, buf, period, dir); - else - dma_sync_single_for_cpu(dev, buf, period, dir); -} - static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); bool elapsed = false; unsigned long flags; @@ -115,22 +84,9 @@ static void __rsnd_dmaen_complete(struct rsnd_mod *mod, */ spin_lock_irqsave(&priv->lock, flags); - if (rsnd_io_is_working(io)) { - rsnd_dmaen_unsync(dmaen, io, dmaen->dma_cnt); - - /* - * Next period is already started. - * Let's sync Next Next period - * see - * rsnd_dmaen_start() - */ - rsnd_dmaen_sync(dmaen, io, dmaen->dma_cnt + 2); - + if (rsnd_io_is_working(io)) elapsed = true; - dmaen->dma_cnt++; - } - spin_unlock_irqrestore(&priv->lock, flags); if (elapsed) @@ -165,14 +121,8 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - if (dmaen->chan) { - int is_play = rsnd_io_is_play(io); - + if (dmaen->chan) dmaengine_terminate_all(dmaen->chan); - dma_unmap_single(dmaen->chan->device->dev, - dmaen->dma_buf, dmaen->dma_len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - } return 0; } @@ -237,11 +187,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; - dma_addr_t buf; - size_t len; - size_t period; int is_play = rsnd_io_is_play(io); - int i; int ret; cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; @@ -258,19 +204,10 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - len = snd_pcm_lib_buffer_bytes(substream); - period = snd_pcm_lib_period_bytes(substream); - buf = dma_map_single(dmaen->chan->device->dev, - substream->runtime->dma_area, - len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - if (dma_mapping_error(dmaen->chan->device->dev, buf)) { - dev_err(dev, "dma map failed\n"); - return -EIO; - } - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - buf, len, period, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); @@ -282,18 +219,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, desc->callback = rsnd_dmaen_complete; desc->callback_param = rsnd_mod_get(dma); - dmaen->dma_buf = buf; - dmaen->dma_len = len; - dmaen->dma_period = period; - dmaen->dma_cnt = 0; - - /* - * synchronize this and next period - * see - * __rsnd_dmaen_complete() - */ - for (i = 0; i < 2; i++) - rsnd_dmaen_sync(dmaen, io, i); + dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); dmaen->cookie = dmaengine_submit(desc); if (dmaen->cookie < 0) { -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix out-of-bound error
by Jaejoong Kim 16 Nov '17

16 Nov '17

The snd_usb_copy_string_desc() retrieves the usb string corresponding to the index number thought the usb_string(). And for NULL-terminated, insert '0' by using the return value of usb_string() as the index of buffer to hold the string. The problem is that the usb_string() also returns the length of the string read(>= 0), but it can also return a negative value, the error or status value of usb_control_msg(). If iClockSource is '0' as shown below, usb_string() will return -EINVAL. This will result in '0' being inserted into buf[-22], and the following KASAN out-of-bound error message will be output. AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 1 bmAttributes 0x07 Internal programmable Clock (synced to SOF) bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 0 To fix out-of-bound error, insert 0 only if the return value of usb_string() is greater than 0. ================================================================== BUG: KASAN: stack-out-of-bounds in parse_audio_unit+0x1327/0x1960 [snd_usb_audio] Write of size 1 at addr ffff88007e66735a by task systemd-udevd/18376 CPU: 0 PID: 18376 Comm: systemd-udevd Not tainted 4.13.0+ #3 Hardware name: LG Electronics 15N540-RFLGL/White Tip Mountain, BIOS 15N5 Call Trace: dump_stack+0x63/0x8d print_address_description+0x70/0x290 ? parse_audio_unit+0x1327/0x1960 [snd_usb_audio] kasan_report+0x265/0x350 __asan_store1+0x4a/0x50 parse_audio_unit+0x1327/0x1960 [snd_usb_audio] ? save_stack+0xb5/0xd0 ? save_stack_trace+0x1b/0x20 ? save_stack+0x46/0xd0 ? kasan_kmalloc+0xad/0xe0 ? kmem_cache_alloc_trace+0xff/0x230 ? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio] ? usb_audio_probe+0x4de/0xf40 [snd_usb_audio] ? usb_probe_interface+0x1f5/0x440 ? driver_probe_device+0x3ed/0x660 ? build_feature_ctl+0xb10/0xb10 [snd_usb_audio] ? save_stack_trace+0x1b/0x20 ? init_object+0x69/0xa0 ? snd_usb_find_csint_desc+0xa8/0xf0 [snd_usb_audio] snd_usb_mixer_controls+0x1dc/0x370 [snd_usb_audio] ? build_audio_procunit+0x890/0x890 [snd_usb_audio] ? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio] ? kmem_cache_alloc_trace+0xff/0x230 ? usb_ifnum_to_if+0xbd/0xf0 snd_usb_create_mixer+0x25b/0x4b0 [snd_usb_audio] ? snd_usb_create_stream+0x255/0x2c0 [snd_usb_audio] usb_audio_probe+0x4de/0xf40 [snd_usb_audio] ? snd_usb_autosuspend.part.7+0x30/0x30 [snd_usb_audio] ? __pm_runtime_idle+0x90/0x90 ? kernfs_activate+0xa6/0xc0 ? usb_match_one_id_intf+0xdc/0x130 ? __pm_runtime_set_status+0x2d4/0x450 usb_probe_interface+0x1f5/0x440 Cc: <stable(a)vger.kernel.org> Signed-off-by: Jaejoong Kim <climbbb.kim(a)gmail.com> --- The AudioControl Interface Descriptor in commit message is from lsusb output with real usb audio DAC. The usb audio product causing the OOB are as follows: http://www.lg.com/uk/lg-friends/lg-AFD-1200 It only prints OOB error and usb audio works well. :) sound/usb/mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e630813..5a83c2c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -204,7 +204,8 @@ static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen) { int len = usb_string(state->chip->dev, index, buf, maxlen - 1); - buf[len] = 0; + if (len > 0) + buf[len] = 0; return len; } -- 2.7.4

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[alsa-devel] [PATCH 0/2] ALSA: intel8x0m: Fine-tuning for five function implementations
by SF Markus Elfring 15 Nov '17

15 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 15 Nov 2017 22:00:11 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust 12 function calls together with a variable assignment Use common error handling code in two functions sound/pci/intel8x0m.c | 100 ++++++++++++++++++++++++++++---------------------- 1 file changed, 57 insertions(+), 43 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 1/2] ASoC: codecs: Add PCM186x binding documentation
by Andrew F. Davis 15 Nov '17

15 Nov '17

Add the dt-binding documentation for the TI PCM186x 2ch and 4ch Audio ADCs With Universal Front End. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- .../devicetree/bindings/sound/pcm186x.txt | 42 ++++++++++++++++++++++ 1 file changed, 42 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm186x.txt diff --git a/Documentation/devicetree/bindings/sound/pcm186x.txt b/Documentation/devicetree/bindings/sound/pcm186x.txt new file mode 100644 index 000000000000..0bc7d265c7bd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm186x.txt @@ -0,0 +1,42 @@ +Texas Instruments PCM186x Universal Audio ADC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "ti,pcm1862", + "ti,pcm1863", + "ti,pcm1864", + "ti,pcm1865" + + - reg : The I2C address of the device for I2C, the chip select + number for SPI. + + - avdd-supply: Analog core power supply (3.3v) + - dvdd-supply: Digital core power supply + - iovdd-supply: Digital IO power supply + See regulator/regulator.txt for more information + +CODEC input pins: + * VINL1 + * VINR1 + * VINL2 + * VINR2 + * VINL3 + * VINR3 + * VINL4 + * VINR4 + +The pins can be used in referring sound node's audio-routing property. + +Example: + + pcm186x: pcm186x@4a { + compatible = "ti,pcm1865"; + reg = <0x4a>; + + avdd-supply = <&reg_3v3_analog>; + dvdd-supply = <&reg_3v3>; + iovdd-supply = <&reg_1v8>; + }; -- 2.15.0

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[alsa-devel] [PATCH 0/2] ALSA: intel8x0: Fine-tuning for seven function implementations
by SF Markus Elfring 15 Nov '17

15 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 15 Nov 2017 20:40:30 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust 15 function calls together with a variable assignment Use common error handling code in two functions sound/pci/intel8x0.c | 111 ++++++++++++++++++++++++++++++--------------------- 1 file changed, 65 insertions(+), 46 deletions(-) -- 2.15.0

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Re: [alsa-devel] [GIT PULL] sound updates for 4.15-rc1
by Linus Torvalds 15 Nov '17

15 Nov '17

On Wed, Nov 15, 2017 at 8:19 AM, Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> wrote: > > The bucketization you are talking about is already implemented in practice, > when you select the SOC options you have a set of machine drivers that show > up in the menu options. So what the network drivers do - because they have an absolutely sh*tload of drivers - is to have vendor-specific helper "gatekeeping" options that do not affect the code in *any* way, but only affect the config option dependencies for that vendor. And those gatekeeping options are then "default y", so that when you add a new gatekeeper, you do not limit people by default - but you also don't add any actual code to the kernel! So for example, you'll see things like this: config NET_VENDOR_ADAPTEC bool "Adaptec devices" default y depends on PCI if NET_VENDOR_ADAPTEC ... all the ADAPTEC network driver configs go here, ... even if right now there's only one endif # NET_VENDOR_ADAPTEC and note how CONFIG_NET_VENDOR_ADAPTEC does not show up in the source AT ALL, except as a "ok, the make recurses into the adapted subdirectory". So NET_VENDOR_ADAPTEC is purely used to make the configuration choice easy. Instead of seeing a million different network driver questions, you see a hundred different network driver _vendor_ choices, and then only if you enable a vendor will you see the individual drivers. So by default, all vendors are enabled, so "make oldconfig" etc works fine, but if you are a human that goes through things by hand, you can just say 'n' to the vendor, and not have to say 'n' to the fifteen different (or sometimes just one - the ADAPTEC example really was a bad choice) individual drivers. That actually works quite well. And exactly because there is no actual code that gets enabled by these vendor choices, they can be enabled by default, so that when you add a new vendor gatekeeper and move an old config option to depend on that vendor, it doesn't affect "make oldconfig" negatively. (Networking didn't always do this, but they've been pretty consistent about it the last many years - exactly because networking used to be a nightmare just because there are _so_ many different random network cards). Linus

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[alsa-devel] [GIT PULL] ASoC fix for v4.14
by Mark Brown 15 Nov '17

15 Nov '17

The following changes since commit df6a3e245541ac61cc99f2887437e0a43dd08f2e: Merge remote-tracking branches 'asoc/topic/tfa9879', 'asoc/topic/ts3a277e', 'asoc/topic/wm8741', 'asoc/topic/wm97xx' and 'asoc/topic/zte' into asoc-next (2017-11-10 21:31:33 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.14-intel-kconfig for you to fetch changes up to 20a52c059be2eb470db75076d55e1d932173fcc3: ASoC: Intel: Add help text for SND_SOC_INTEL_SST_TOPLEVEL (2017-11-15 11:43:41 +0000) ---------------------------------------------------------------- ASoC: Fix for Intel Kconfig Fix the Intel Kconfig issue reported by Linus. ---------------------------------------------------------------- Vinod Koul (1): ASoC: Intel: Add help text for SND_SOC_INTEL_SST_TOPLEVEL sound/soc/intel/Kconfig | 5 +++++ 1 file changed, 5 insertions(+)

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[alsa-devel] [PATCH] ALSA: ice1712: Use common error handling code in two functions
by SF Markus Elfring 15 Nov '17

15 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 15 Nov 2017 15:50:06 +0100 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ice1712/ice1712.c | 114 +++++++++++++++++++------------------------- 1 file changed, 50 insertions(+), 64 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0e66afa403a3..691b8844465f 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2534,14 +2534,14 @@ static int snd_ice1712_create(struct snd_card *card, dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(28)) < 0) { dev_err(card->dev, "architecture does not support 28bit PCI busmaster DMA\n"); - pci_disable_device(pci); - return -ENXIO; + err = -ENXIO; + goto disable_device; } ice = kzalloc(sizeof(*ice), GFP_KERNEL); if (ice == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } ice->omni = omni ? 1 : 0; if (cs8427_timeout < 1) @@ -2580,8 +2580,7 @@ static int snd_ice1712_create(struct snd_card *card, err = pci_request_regions(pci, "ICE1712"); if (err < 0) { kfree(ice); - pci_disable_device(pci); - return err; + goto disable_device; } ice->port = pci_resource_start(pci, 0); ice->ddma_port = pci_resource_start(pci, 1); @@ -2591,29 +2590,31 @@ static int snd_ice1712_create(struct snd_card *card, if (request_irq(pci->irq, snd_ice1712_interrupt, IRQF_SHARED, KBUILD_MODNAME, ice)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_ice1712_free(ice); - return -EIO; + goto e_io; } ice->irq = pci->irq; - if (snd_ice1712_read_eeprom(ice, modelname) < 0) { - snd_ice1712_free(ice); - return -EIO; - } - if (snd_ice1712_chip_init(ice) < 0) { - snd_ice1712_free(ice); - return -EIO; - } + if (snd_ice1712_read_eeprom(ice, modelname) < 0 || + snd_ice1712_chip_init(ice) < 0) + goto e_io; err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); - if (err < 0) { - snd_ice1712_free(ice); - return err; - } + if (err < 0) + goto free_sound_chip; *r_ice1712 = ice; return 0; + +disable_device: + pci_disable_device(pci); + return err; + +e_io: + err = -EIO; +free_sound_chip: + snd_ice1712_free(ice); + return err; } @@ -2651,10 +2652,8 @@ static int snd_ice1712_probe(struct pci_dev *pci, err = snd_ice1712_create(card, pci, model[dev], omni[dev], cs8427_timeout[dev], dxr_enable[dev], &ice); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; for (tbl = card_tables; *tbl; tbl++) { for (c = *tbl; c->subvendor; c++) { @@ -2665,10 +2664,8 @@ static int snd_ice1712_probe(struct pci_dev *pci, strcpy(card->driver, c->driver); if (c->chip_init) { err = c->chip_init(ice); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; } goto __found; } @@ -2678,45 +2675,33 @@ static int snd_ice1712_probe(struct pci_dev *pci, __found: err = snd_ice1712_pcm_profi(ice, pcm_dev++); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm(ice, pcm_dev++); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; } err = snd_ice1712_ac97_mixer(ice); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; err = snd_ice1712_build_controls(ice); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; if (c->build_controls) { err = c->build_controls(ice); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; } if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm_ds(ice, pcm_dev++); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; } if (!c->no_mpu401) { @@ -2725,10 +2710,9 @@ static int snd_ice1712_probe(struct pci_dev *pci, c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, -1, &ice->rmidi[0]); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; + if (c->mpu401_1_name) /* Preferred name available in card_info */ snprintf(ice->rmidi[0]->name, @@ -2743,10 +2727,9 @@ static int snd_ice1712_probe(struct pci_dev *pci, MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, -1, &ice->rmidi[1]); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; + if (c->mpu401_2_name) /* Preferred name available in card_info */ snprintf(ice->rmidi[1]->name, @@ -2762,13 +2745,16 @@ static int snd_ice1712_probe(struct pci_dev *pci, card->shortname, ice->port, ice->irq); err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; + pci_set_drvdata(pci, card); dev++; return 0; + +free_card: + snd_card_free(card); + return err; } static void snd_ice1712_remove(struct pci_dev *pci) -- 2.15.0

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[alsa-devel] [PATCH] ALSA: HDA: Improve unlocking of a mutex in hdmi_pcm_open()
by SF Markus Elfring 15 Nov '17

15 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 15 Nov 2017 14:46:19 +0100 * Add a jump target so that a call of the function "mutex_unlock" is stored only twice in this function implementation. * Replace four calls by goto statements. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c19c81d230bd..f1c0b6cab90a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1205,8 +1205,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, pin_idx = hinfo_to_pin_index(codec, hinfo); if (!spec->dyn_pcm_assign) { if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } } else { /* no pin is assigned to the PCM @@ -1214,16 +1214,13 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, */ if (pin_idx < 0) { err = hdmi_pcm_open_no_pin(hinfo, codec, substream); - mutex_unlock(&spec->pcm_lock); - return err; + goto unlock; } } err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); - if (err < 0) { - mutex_unlock(&spec->pcm_lock); - return err; - } + if (err < 0) + goto unlock; per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ @@ -1260,8 +1257,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - mutex_unlock(&spec->pcm_lock); - return -ENODEV; + err = -ENODEV; + goto unlock; } } @@ -1275,6 +1272,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); return 0; + +unlock: + mutex_unlock(&spec->pcm_lock); + return err; } /* -- 2.15.0

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[alsa-devel] [PATCH 0/3] ALSA: fm801: Fine-tuning for seven function implementations
by SF Markus Elfring 15 Nov '17

15 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 15 Nov 2017 13:40:43 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 17 function calls together with a variable assignment Use common error handling code in two functions Rename a jump label in snd_card_fm801_probe() sound/pci/fm801.c | 140 +++++++++++++++++++++++++++++------------------------- 1 file changed, 76 insertions(+), 64 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH] ASoC: Intel: Add help text for SND_SOC_INTEL_SST_TOPLEVEL
by Vinod Koul 15 Nov '17

15 Nov '17

Symbol SND_SOC_INTEL_SST_TOPLEVEL is user selectable so add the help text for this symbol. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index bb8be10b8437..7b49d04e3c60 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -34,6 +34,11 @@ config SND_SOC_INTEL_SST_TOPLEVEL depends on X86 || COMPILE_TEST select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON + help + Intel ASoC Audio Drivers. If you have a Intel machine that + has audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y or M + If unsure select "N". config SND_SOC_INTEL_HASWELL tristate "Intel ASoC SST driver for Haswell/Broadwell" -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: mfld: allow mfld machine to be build
by Vinod Koul 15 Nov '17

15 Nov '17

Commit b97169da0699: ("ASoC: Intel: create atom folder and move atom platform files in") missed adding Kconfig and Makefile support for MFLD machine driver, so add it while alos fixing the relative patch to codec header Fixes: b97169da0699: ("ASoC: Intel: create atom folder and move atom platform files in") Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/Makefile | 2 ++ sound/soc/intel/boards/mfld_machine.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a5c5bc5732a2..070cb1afefe1 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -1,3 +1,4 @@ +snd-soc-sst-mfld-objs := mfld_machine.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o @@ -19,6 +20,7 @@ snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-sst-mfld.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 6f44acfb4aae..c3d80123610e 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -33,7 +33,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> -#include "../codecs/sn95031.h" +#include "../../codecs/sn95031.h" #define MID_MONO 1 #define MID_STEREO 2 -- 2.7.4

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[alsa-devel] [PATCH v2] Add support for Tempo Semiconductor's TSCS42xx audio CODEC
by Steven Eckhoff 15 Nov '17

15 Nov '17

Currently there is no support for the TSCS42xx audio CODEC. Add support for it. Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> Cc: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- .../ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt | 185 ++ .../devicetree/bindings/sound/tscs42xx.txt | 24 + .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 5 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tscs42xx.c | 2006 +++++++++++++++ sound/soc/codecs/tscs42xx.h | 2693 ++++++++++++++++++++ 8 files changed, 4924 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt create mode 100644 Documentation/devicetree/bindings/sound/tscs42xx.txt create mode 100644 sound/soc/codecs/tscs42xx.c create mode 100644 sound/soc/codecs/tscs42xx.h diff --git a/Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt b/Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt new file mode 100644 index 000000000000..ed409a97e812 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt @@ -0,0 +1,185 @@ +What: /sys/bus/i2c/devices/.../dsp/eq[n]_ch[n]_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the equalizer biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/eq[n]_ch[n]_prescale +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the prescaler attenuation of the equalizer + biquad cascade + +What: /sys/bus/i2c/devices/.../dsp/bass_ext[n]_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the bass extraction biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/bass_lmt_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the bass non linear function coefficient + +What: /sys/bus/i2c/devices/.../dsp/bass_cto_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the bass cut-off biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_ext[n]_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the treb extraction biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_lmt_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the treb non linear function coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_cto_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the treb cut-off biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/3d_coef +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the 3d coefficient + +What: /sys/bus/i2c/devices/.../dsp/3d_mix +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets 3d mix coefficient + +What: /sys/bus/i2c/devices/.../dsp/bass_mix +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets bass mix coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_mix +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets treb mix coefficient + +What: /sys/bus/i2c/devices/.../controls/eq1_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one + +What: /sys/bus/i2c/devices/.../controls/eq1_bands_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one bands: + 0h = Pre-scale only + 1h = Pre-scale + EQ band 0 + 2h = Pre-scale + EQ bands 0 to 1 + ... + 6h = Pre-scale + EQ bands 0 to 5 + +What: /sys/bus/i2c/devices/.../controls/eq2_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one + +What: /sys/bus/i2c/devices/.../controls/eq2_bands_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one bands: + 0h = Pre-scale only + 1h = Pre-scale + EQ band 0 + 2h = Pre-scale + EQ bands 0 to 1 + ... + 6h = Pre-scale + EQ bands 0 to 5 + +What: /sys/bus/i2c/devices/.../controls/cle_level_mode +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the compressor, limiter and expander level detection mode: + 0h = Average + 1h = Peak + +What: /sys/bus/i2c/devices/.../controls/cle_level_detect_window +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Sets the compressor, limiter and expander level detection mode + window: + 0h = Equivalent of 512 samples + at the selected Base Rate (~10-16ms) + 1h = Equivalent of 64 samples + at the selected Base Rate (~1.3-2ms) + +What: /sys/bus/i2c/devices/.../controls/exp_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the expander + +What: /sys/bus/i2c/devices/.../controls/limit_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the limiter + +What: /sys/bus/i2c/devices/.../controls/comp_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the compressor + +What: /sys/bus/i2c/devices/.../controls/3d_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the 3D enhancer + +What: /sys/bus/i2c/devices/.../controls/treb_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the treble enhancer + +What: /sys/bus/i2c/devices/.../controls/treb_nlf_bypass +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the treble non-linear function bypass + +What: /sys/bus/i2c/devices/.../controls/bass_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the bass enhancer + +What: /sys/bus/i2c/devices/.../controls/bass_nlf_bypass +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Enables the bass non-linear function bypass + +What: /sys/bus/i2c/devices/.../control_regs/export +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Adds the specified control register to the sysfs interface + +What: /sys/bus/i2c/devices/.../control_regs/unexport +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +Description: Removes the specified control register to the sysfs interface diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt new file mode 100644 index 000000000000..2f9c8affde48 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -0,0 +1,24 @@ +TSCS42XX Audio CODEC + +Required Properties: + + - compatible : "tscs:tscs42xx" + + - reg : The I2C address of the device + + - mclk-src : The source of the master clock. This can be "mclk" or "xtal". + + - mclk-src-freq : The frequency of the master clock + + - clocks : The phandle of the master clock + +Example: + +codec: tscs42xx@69 { + compatible = "tscs,tscs42xx"; + reg = <0x69>; + mclk-src = "mclk"; + mclk-src-freq = <25000000>; + clocks = <&cprman BCM2835_CLOCK_GP1>; + status = "okay"; +}; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index b1eeca851d6f..24036f503aa0 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -349,6 +349,7 @@ tpo TPO tronfy Tronfy tronsmart Tronsmart truly Truly Semiconductors Limited +tscs Tempo Semiconductor tsd Theobroma Systems Design und Consulting GmbH tyan Tyan Computer Corporation ucrobotics uCRobotics diff --git a/MAINTAINERS b/MAINTAINERS index 650aa0e0e9d8..63d1d892842a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13674,6 +13674,11 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial.git S: Maintained K: ^Subject:.*(?i)trivial +TEMPO SEMICONDUCTOR DRIVERS +M: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +S: Maintained +F: sound/soc/codecs/tscs42xx.* + TTY LAYER M: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> M: Jiri Slaby <jslaby(a)suse.com> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c367d11079bc..34e911c1e082 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -217,6 +217,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_TSCS42XX if I2C help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -1142,6 +1143,13 @@ config SND_SOC_ZX_AUD96P22 depends on I2C select REGMAP_I2C +config SND_SOC_TSCS42XX + tristate "Tempo Semiconductor TSCS42xx CODEC" + depends on I2C + select REGMAP_I2C + help + Add support for Tempo Semiconductor's TSCS42xx audio CODEC. + # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 05018b7ca72b..a20e643b4067 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -230,6 +230,7 @@ snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o +snd-soc-tscs42xx-objs := tscs42xx.o # Amp snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o @@ -468,6 +469,7 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o +obj-$(CONFIG_SND_SOC_TSCS42XX) += snd-soc-tscs42xx.o # Amp obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c new file mode 100644 index 000000000000..dad874a19c58 --- /dev/null +++ b/sound/soc/codecs/tscs42xx.c @@ -0,0 +1,2006 @@ +/* + * tscs42xx.c -- TSCS42xx ALSA SoC Audio driver + * + * Copyright 2017 Tempo Semiconductor, Inc. + * + * Author: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/moduleparam.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/mutex.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/firmware.h> +#include <linux/sysfs.h> + +#include "tscs42xx.h" + +enum { + PLL_SRC_CLK_XTAL, + PLL_SRC_CLK_MCLK2, +}; + +/* + * Functions that receive a pointer to data + * should assume that the data is locked. + */ +struct tscs42xx_priv { + struct regmap *regmap; + struct clk *mclk; + int mclk_src_freq; + int pll_src_clk; + int bclk_ratio; + int samplerate; + int pll_users; + struct snd_soc_codec *codec; + struct mutex lock; +}; + +static bool tscs42xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case R_DACCRWRL: + case R_DACCRWRM: + case R_DACCRWRH: + case R_DACCRRDL: + case R_DACCRRDM: + case R_DACCRRDH: + case R_DACCRSTAT: + case R_DACCRADDR: + case R_PLLCTL0: + return true; + default: + return false; + }; +} + +static bool tscs42xx_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case R_DACCRWRL: + case R_DACCRWRM: + case R_DACCRWRH: + case R_DACCRRDL: + case R_DACCRRDM: + case R_DACCRRDH: + return true; + default: + return false; + }; +} + +static const struct regmap_config tscs42xx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .volatile_reg = tscs42xx_volatile, + .precious_reg = tscs42xx_precious, + .max_register = R_DACMBCREL3H, + + .cache_type = REGCACHE_RBTREE, +}; + +static struct reg_default r_inits[] = { + { .reg = R_ADCSR, .def = RV_ADCSR_ABCM_64 }, + { .reg = R_DACSR, .def = RV_DACSR_DBCM_64 }, + { .reg = R_AIC2, .def = RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, +}; + +#define NUM_DACCR_BYTES 3 +static int load_dac_coefficient_ram(struct snd_soc_codec *codec) +{ + const struct firmware *fw = NULL; + int ret; + int i; + int j; + int addr; + + ret = request_firmware_direct(&fw, "tscs42xx_daccram.dfw", codec->dev); + if (ret) { + dev_info(codec->dev, + "No tscs42xx_daccram.dfw file found (%d)\n", ret); + return 0; + } + + if (fw->size % NUM_DACCR_BYTES != 0) { + ret = -EINVAL; + dev_err(codec->dev, "Malformed daccram file (%d)\n", ret); + return ret; + } + + for (i = 0, addr = 0; i < fw->size; i += NUM_DACCR_BYTES, addr++) { + + do { + ret = snd_soc_read(codec, R_DACCRSTAT); + } while (ret > 0); + + if (ret < 0) { + dev_err(codec->dev, + "Failed to read daccrstat (%d)\n", ret); + return ret; + } + + /* Explicit address update */ + ret = snd_soc_write(codec, R_DACCRADDR, addr); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write DACCRADDR (%d)\n", ret); + return ret; + } + + for (j = 0; j < NUM_DACCR_BYTES; j++) { /* FW in big endian */ + + do { + ret = snd_soc_read(codec, R_DACCRSTAT); + } while (ret > 0); + + if (ret < 0) { + dev_err(codec->dev, + "Failed to read daccrstat (%d)\n", ret); + return ret; + } + + /* Write LSB first due to auto increment on MSB */ + ret = snd_soc_write(codec, R_DACCRWRL + j, + fw->data[i + NUM_DACCR_BYTES - 1 - j]); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write coefficient (%d)\n", + ret); + return ret; + } + } + } + + dev_info(codec->dev, "Loaded tscs42xx_daccram.dfw\n"); + + return 0; +} + +#define NUM_CONTROL_BYTES 2 +static int load_control_regs(struct snd_soc_codec *codec) +{ + const struct firmware *fw = NULL; + int ret; + int i; + + ret = request_firmware_direct(&fw, "tscs42xx_controls.dfw", codec->dev); + if (ret) { + dev_info(codec->dev, + "No tscs42xx_controls.dfw file found (%d)\n", ret); + return 0; + } + + if (fw->size % NUM_CONTROL_BYTES != 0) { + ret = -EINVAL; + dev_err(codec->dev, "Malformed controls file (%d)\n", ret); + return ret; + } + + for (i = 0; i < fw->size; i += NUM_CONTROL_BYTES) { + ret = snd_soc_write(codec, fw->data[i], fw->data[i + 1]); + if (ret < 0) { + dev_err(codec->dev, "Failed to write control (%d)\n", + ret); + return ret; + } + } + + dev_info(codec->dev, "Loaded tscs42xx_controls.dfw\n"); + + return 0; +} + +#define PLL_LOCK_TIME_MAX 10 +static bool plls_locked(struct snd_soc_codec *codec) +{ + int reg; + int count; + bool ret = false; + + for (count = 0; count < PLL_LOCK_TIME_MAX; count++) { + reg = snd_soc_read(codec, R_PLLCTL0); + if (reg < 0) { + dev_err(codec->dev, + "Failed to read PLL lock status (%d)\n", ret); + break; + } else if (reg > 0) { + ret = true; + break; + } + mdelay(1); + } + + return ret; +} + +/* D2S Input Select */ +static const char * const d2s_input_select_text[] = { + "Line 1", "Line 2" +}; + +static const struct soc_enum d2s_input_select_enum = +SOC_ENUM_SINGLE(R_INMODE, FB_INMODE_DS, ARRAY_SIZE(d2s_input_select_text), + d2s_input_select_text); + +static const struct snd_kcontrol_new d2s_input_mux = +SOC_DAPM_ENUM("D2S_IN_MUX", d2s_input_select_enum); + +/* Input L Capture Route */ +static const char * const input_select_text[] = { + "Line 1", "Line 2", "Line 3", "D2S" +}; + +static const struct soc_enum left_input_select_enum = +SOC_ENUM_SINGLE(R_INSELL, FB_INSELL, ARRAY_SIZE(input_select_text), + input_select_text); + +static const struct snd_kcontrol_new left_input_select = +SOC_DAPM_ENUM("LEFT_INPUT_SELECT_ENUM", left_input_select_enum); + +/* Input R Capture Route */ +static const struct soc_enum right_input_select_enum = +SOC_ENUM_SINGLE(R_INSELR, FB_INSELR, ARRAY_SIZE(input_select_text), + input_select_text); + +static const struct snd_kcontrol_new right_input_select = +SOC_DAPM_ENUM("RIGHT_INPUT_SELECT_ENUM", right_input_select_enum); + +/* Input Channel Mapping */ +static const char * const ch_map_select_text[] = { + "Normal", "Left to Right", "Right to Left", "Swap" +}; + +static const struct soc_enum ch_map_select_enum = +SOC_ENUM_SINGLE(R_AIC2, FB_AIC2_ADCDSEL, ARRAY_SIZE(ch_map_select_text), + ch_map_select_text); + +static int dapm_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + mdelay(5); + return 0; +} + +static int dapm_micb_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + mdelay(5); + return 0; +} + +static const struct snd_soc_dapm_widget tscs42xx_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("Vref", 1, R_PWRM2, FB_PWRM2_VREF, 0, + dapm_vref_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + + /* Headphone */ + SND_SOC_DAPM_DAC("DAC L", "HiFi Playback", R_PWRM2, FB_PWRM2_HPL, 0), + SND_SOC_DAPM_DAC("DAC R", "HiFi Playback", R_PWRM2, FB_PWRM2_HPR, 0), + SND_SOC_DAPM_OUTPUT("Headphone L"), + SND_SOC_DAPM_OUTPUT("Headphone R"), + + /* Speaker */ + SND_SOC_DAPM_DAC("ClassD L", "HiFi Playback", + R_PWRM2, FB_PWRM2_SPKL, 0), + SND_SOC_DAPM_DAC("ClassD R", "HiFi Playback", + R_PWRM2, FB_PWRM2_SPKR, 0), + SND_SOC_DAPM_OUTPUT("Speaker L"), + SND_SOC_DAPM_OUTPUT("Speaker R"), + + /* Capture */ + SND_SOC_DAPM_PGA("Analog In PGA L", R_PWRM1, FB_PWRM1_PGAL, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog In PGA R", R_PWRM1, FB_PWRM1_PGAR, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog Boost L", R_PWRM1, FB_PWRM1_BSTL, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog Boost R", R_PWRM1, FB_PWRM1_BSTR, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC Mute", R_CNVRTR0, FB_CNVRTR0_HPOR, true, NULL, 0), + SND_SOC_DAPM_ADC("ADC L", "HiFi Capture", R_PWRM1, FB_PWRM1_ADCL, 0), + SND_SOC_DAPM_ADC("ADC R", "HiFi Capture", R_PWRM1, FB_PWRM1_ADCR, 0), + + /* Capture Input */ + SND_SOC_DAPM_MUX("Input L Capture Route", R_PWRM2, + FB_PWRM2_INSELL, 0, &left_input_select), + SND_SOC_DAPM_MUX("Input R Capture Route", R_PWRM2, + FB_PWRM2_INSELR, 0, &right_input_select), + + /* Digital Mic */ + SND_SOC_DAPM_SUPPLY_S("Digital Mic Enable", 2, R_DMICCTL, + FB_DMICCTL_DMICEN, 0, NULL, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_INPUT("Digital Mic L"), + SND_SOC_DAPM_INPUT("Digital Mic R"), + + /* Analog Mic */ + SND_SOC_DAPM_SUPPLY_S("Mic Bias", 2, R_PWRM1, FB_PWRM1_MICB, + 0, dapm_micb_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + + /* Line In */ + SND_SOC_DAPM_INPUT("Line In 1 L"), + SND_SOC_DAPM_INPUT("Line In 1 R"), + SND_SOC_DAPM_INPUT("Line In 2 L"), + SND_SOC_DAPM_INPUT("Line In 2 R"), + SND_SOC_DAPM_INPUT("Line In 3 L"), + SND_SOC_DAPM_INPUT("Line In 3 R"), +}; + +static const struct snd_soc_dapm_route tscs42xx_intercon[] = { + {"DAC L", NULL, "Vref"}, + {"DAC R", NULL, "Vref"}, + {"Headphone L", NULL, "DAC L"}, + {"Headphone R", NULL, "DAC R"}, + + {"ClassD L", NULL, "Vref"}, + {"ClassD R", NULL, "Vref"}, + {"Speaker L", NULL, "ClassD L"}, + {"Speaker R", NULL, "ClassD R"}, + + {"Input L Capture Route", NULL, "Vref"}, + {"Input R Capture Route", NULL, "Vref"}, + + {"Mic Bias", NULL, "Vref"}, + + {"Input L Capture Route", "Line 1", "Line In 1 L"}, + {"Input R Capture Route", "Line 1", "Line In 1 R"}, + {"Input L Capture Route", "Line 2", "Line In 2 L"}, + {"Input R Capture Route", "Line 2", "Line In 2 R"}, + {"Input L Capture Route", "Line 3", "Line In 3 L"}, + {"Input R Capture Route", "Line 3", "Line In 3 R"}, + + {"Analog In PGA L", NULL, "Input L Capture Route"}, + {"Analog In PGA R", NULL, "Input R Capture Route"}, + {"Analog Boost L", NULL, "Analog In PGA L"}, + {"Analog Boost R", NULL, "Analog In PGA R"}, + {"ADC Mute", NULL, "Analog Boost L"}, + {"ADC Mute", NULL, "Analog Boost R"}, + {"ADC L", NULL, "ADC Mute"}, + {"ADC R", NULL, "ADC Mute"}, +}; + +/*********** + * Volumes * + ***********/ + +static DECLARE_TLV_DB_SCALE(hpvol, -8850, 75, 0); +static DECLARE_TLV_DB_SCALE(spkvol, -7725, 75, 0); +static DECLARE_TLV_DB_SCALE(dacvol, -9563, 38, 0); +static DECLARE_TLV_DB_SCALE(adcvol, -7125, 38, 0); +static DECLARE_TLV_DB_SCALE(invol, -1725, 75, 0); + +/********* + * INSEL * + *********/ + +static DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 1000, 0); + +/************ + * CONTROLS * + ************/ +static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { + /* Volumes */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR, + FB_HPVOLL, 0x7F, 0, hpvol), + SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR, + FB_SPKVOLL, 0x7F, 0, spkvol), + SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR, + FB_DACVOLL, 0xFF, 0, dacvol), + SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR, + FB_ADCVOLL, 0xFF, 0, adcvol), + SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR, + FB_INVOLL, 0x3F, 0, invol), + + /* INSEL */ + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR, + FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB, + 0, mic_boost_tlv), + + /* Input Channel Map */ + SOC_ENUM("Input Channel Map Switch", ch_map_select_enum), +}; + +#define TSCS42XX_RATES SNDRV_PCM_RATE_8000_96000 + +#define TSCS42XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int setup_sample_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + unsigned int width; + int ret; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + width = RV_AIC1_WL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + width = RV_AIC1_WL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + width = RV_AIC1_WL_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + width = RV_AIC1_WL_32; + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unsupported format width (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_AIC1, RM_AIC1_WL, width); + if (ret < 0) { + dev_err(codec->dev, "Failed to set sample width (%d)\n", ret); + return ret; + } + + return 0; +} + +static int setup_sample_rate(struct snd_soc_codec *codec, + unsigned int rate, + struct tscs42xx_priv *tscs42xx) +{ + unsigned int br, bm; + int ret; + + switch (rate) { + case 8000: + br = RV_DACSR_DBR_32; + bm = RV_DACSR_DBM_PT25; + break; + case 16000: + br = RV_DACSR_DBR_32; + bm = RV_DACSR_DBM_PT5; + break; + case 24000: + br = RV_DACSR_DBR_48; + bm = RV_DACSR_DBM_PT5; + break; + case 32000: + br = RV_DACSR_DBR_32; + bm = RV_DACSR_DBM_1; + break; + case 48000: + br = RV_DACSR_DBR_48; + bm = RV_DACSR_DBM_1; + break; + case 96000: + br = RV_DACSR_DBR_48; + bm = RV_DACSR_DBM_2; + break; + case 11025: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_PT25; + break; + case 22050: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_PT5; + break; + case 44100: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_1; + break; + case 88200: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_2; + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", rate); + return -EINVAL; + } + + /* DAC and ADC share bit and frame clock */ + ret = snd_soc_update_bits(codec, R_DACSR, RM_DACSR_DBR, br); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_DACSR, RM_DACSR_DBM, bm); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_ADCSR, RM_DACSR_DBR, br); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_ADCSR, RM_DACSR_DBM, bm); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + + tscs42xx->samplerate = rate; + + return 0; +} + +struct reg_setting { + unsigned int addr; + unsigned int val; + unsigned int mask; +}; + +#define PLL_REG_SETTINGS_COUNT 13 +struct pll_ctl { + int input_freq; + struct reg_setting settings[PLL_REG_SETTINGS_COUNT]; +}; + +#define PLL_CTL(f, rt, rd, r1b_l, r9, ra, rb, \ + rc, r12, r1b_h, re, rf, r10, r11) \ + { \ + .input_freq = f, \ + .settings = { \ + {R_TIMEBASE, rt, 0xFF}, \ + {R_PLLCTLD, rd, 0xFF}, \ + {R_PLLCTL1B, r1b_l, 0x0F}, \ + {R_PLLCTL9, r9, 0xFF}, \ + {R_PLLCTLA, ra, 0xFF}, \ + {R_PLLCTLB, rb, 0xFF}, \ + {R_PLLCTLC, rc, 0xFF}, \ + {R_PLLCTL12, r12, 0xFF}, \ + {R_PLLCTL1B, r1b_h, 0xF0}, \ + {R_PLLCTLE, re, 0xFF}, \ + {R_PLLCTLF, rf, 0xFF}, \ + {R_PLLCTL10, r10, 0xFF}, \ + {R_PLLCTL11, r11, 0xFF}, \ + }, \ + } + +static const struct pll_ctl pll_ctls[] = { + PLL_CTL(1411200, 0x05, + 0x39, 0x04, 0x07, 0x02, 0xC3, 0x04, + 0x1B, 0x10, 0x03, 0x03, 0xD0, 0x02), + PLL_CTL(1536000, 0x05, + 0x1A, 0x04, 0x02, 0x03, 0xE0, 0x01, + 0x1A, 0x10, 0x02, 0x03, 0xB9, 0x01), + PLL_CTL(2822400, 0x0A, + 0x23, 0x04, 0x07, 0x04, 0xC3, 0x04, + 0x22, 0x10, 0x05, 0x03, 0x58, 0x02), + PLL_CTL(3072000, 0x0B, + 0x22, 0x04, 0x07, 0x03, 0x48, 0x03, + 0x1A, 0x10, 0x04, 0x03, 0xB9, 0x01), + PLL_CTL(5644800, 0x15, + 0x23, 0x04, 0x0E, 0x04, 0xC3, 0x04, + 0x1A, 0x10, 0x08, 0x03, 0xE0, 0x01), + PLL_CTL(6144000, 0x17, + 0x1A, 0x04, 0x08, 0x03, 0xE0, 0x01, + 0x1A, 0x10, 0x08, 0x03, 0xB9, 0x01), + PLL_CTL(12000000, 0x2E, + 0x1B, 0x04, 0x19, 0x03, 0x00, 0x03, + 0x2A, 0x10, 0x19, 0x05, 0x98, 0x04), + PLL_CTL(19200000, 0x4A, + 0x13, 0x04, 0x14, 0x03, 0x80, 0x01, + 0x1A, 0x10, 0x19, 0x03, 0xB9, 0x01), + PLL_CTL(22000000, 0x55, + 0x2A, 0x04, 0x37, 0x05, 0x00, 0x06, + 0x22, 0x10, 0x26, 0x03, 0x49, 0x02), + PLL_CTL(22579200, 0x57, + 0x22, 0x04, 0x31, 0x03, 0x20, 0x03, + 0x1A, 0x10, 0x1D, 0x03, 0xB3, 0x01), + PLL_CTL(24000000, 0x5D, + 0x13, 0x04, 0x19, 0x03, 0x80, 0x01, + 0x1B, 0x10, 0x19, 0x05, 0x4C, 0x02), + PLL_CTL(24576000, 0x5F, + 0x13, 0x04, 0x1D, 0x03, 0xB3, 0x01, + 0x22, 0x10, 0x40, 0x03, 0x72, 0x03), + PLL_CTL(27000000, 0x68, + 0x22, 0x04, 0x4B, 0x03, 0x00, 0x04, + 0x2A, 0x10, 0x7D, 0x03, 0x20, 0x06), + PLL_CTL(36000000, 0x8C, + 0x1B, 0x04, 0x4B, 0x03, 0x00, 0x03, + 0x2A, 0x10, 0x7D, 0x03, 0x98, 0x04), + PLL_CTL(25000000, 0x61, + 0x1B, 0x04, 0x37, 0x03, 0x2B, 0x03, + 0x1A, 0x10, 0x2A, 0x03, 0x39, 0x02), + PLL_CTL(26000000, 0x65, + 0x23, 0x04, 0x41, 0x05, 0x00, 0x06, + 0x1A, 0x10, 0x26, 0x03, 0xEF, 0x01), + PLL_CTL(12288000, 0x2F, + 0x1A, 0x04, 0x12, 0x03, 0x1C, 0x02, + 0x22, 0x10, 0x20, 0x03, 0x72, 0x03), + PLL_CTL(40000000, 0x9B, + 0x22, 0x08, 0x7D, 0x03, 0x80, 0x04, + 0x23, 0x10, 0x7D, 0x05, 0xE4, 0x06), + PLL_CTL(512000, 0x01, + 0x22, 0x04, 0x01, 0x03, 0xD0, 0x02, + 0x1B, 0x10, 0x01, 0x04, 0x72, 0x03), + PLL_CTL(705600, 0x02, + 0x22, 0x04, 0x02, 0x03, 0x15, 0x04, + 0x22, 0x10, 0x01, 0x04, 0x80, 0x02), + PLL_CTL(1024000, 0x03, + 0x22, 0x04, 0x02, 0x03, 0xD0, 0x02, + 0x1B, 0x10, 0x02, 0x04, 0x72, 0x03), + PLL_CTL(2048000, 0x07, + 0x22, 0x04, 0x04, 0x03, 0xD0, 0x02, + 0x1B, 0x10, 0x04, 0x04, 0x72, 0x03), + PLL_CTL(2400000, 0x08, + 0x22, 0x04, 0x05, 0x03, 0x00, 0x03, + 0x23, 0x10, 0x05, 0x05, 0x98, 0x04), +}; + +static const struct pll_ctl *get_pll_ctl(int input_freq) +{ + int i; + const struct pll_ctl *pll_ctl = NULL; + + for (i = 0; i < ARRAY_SIZE(pll_ctls); ++i) + if (input_freq == pll_ctls[i].input_freq) { + pll_ctl = &pll_ctls[i]; + break; + } + + return pll_ctl; +} + +static int sample_rate_to_pll_freq_out(int sample_rate) +{ + switch (sample_rate) { + case 11025: + case 22050: + case 44100: + case 88200: + return 112896000; + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + return 122880000; + default: + return -EINVAL; + } +} + + +static int set_pll_ctl_from_input_freq(struct snd_soc_codec *codec, + const int input_freq) +{ + int ret; + int i; + const struct pll_ctl *pll_ctl; + + pll_ctl = get_pll_ctl(input_freq); + if (!pll_ctl) { + ret = -EINVAL; + dev_err(codec->dev, "No PLL input entry for %d (%d)\n", + input_freq, ret); + return ret; + } + + for (i = 0; i < PLL_REG_SETTINGS_COUNT; ++i) { + ret = snd_soc_update_bits(codec, + pll_ctl->settings[i].addr, + pll_ctl->settings[i].mask, + pll_ctl->settings[i].val); + if (ret < 0) { + dev_err(codec->dev, "Failed to set pll ctl (%d)\n", + ret); + return ret; + } + } + + return 0; +} + +static int configure_clocks(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = set_pll_ctl_from_input_freq(codec, tscs42xx->mclk_src_freq); + if (ret < 0) { + dev_err(codec->dev, "Failed to setup PLL input (%d)\n", ret); + return ret; + } + + switch (tscs42xx->pll_src_clk) { + case PLL_SRC_CLK_XTAL: + ret = snd_soc_write(codec, R_PLLREFSEL, + RV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1 | + RV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set pll reference input (%d)\n", + ret); + return ret; + } + break; + case PLL_SRC_CLK_MCLK2: + ret = clk_set_rate(tscs42xx->mclk, tscs42xx->mclk_src_freq); + if (ret < 0) { + dev_err(codec->dev, + "Could not set mclk rate %d (%d)\n", + tscs42xx->mclk_src_freq, ret); + return ret; + } + + ret = clk_prepare_enable(tscs42xx->mclk); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable mclk: (%d)\n", + ret); + return ret; + } + + ret = snd_soc_write(codec, R_PLLREFSEL, + RV_PLLREFSEL_PLL1_REF_SEL_MCLK2 | + RV_PLLREFSEL_PLL2_REF_SEL_MCLK2); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set PLL reference (%d)\n", ret); + return ret; + } + break; + } + + return 0; +} + +static int power_down_audio_plls(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + tscs42xx->pll_users--; + if (tscs42xx->pll_users > 0) + return 0; + + ret = snd_soc_update_bits(codec, R_PLLCTL1C, + RM_PLLCTL1C_PDB_PLL1, + RV_PLLCTL1C_PDB_PLL1_DISABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to turn PLL off (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_PLLCTL1C, + RM_PLLCTL1C_PDB_PLL2, + RV_PLLCTL1C_PDB_PLL2_DISABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to turn PLL off (%d)\n", ret); + return ret; + } + + return 0; +} + +static int power_up_audio_plls(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int freq_out; + int ret; + unsigned int mask; + unsigned int val; + + freq_out = sample_rate_to_pll_freq_out(tscs42xx->samplerate); + switch (freq_out) { + case 122880000: /* 48k */ + mask = RM_PLLCTL1C_PDB_PLL1; + val = RV_PLLCTL1C_PDB_PLL1_ENABLE; + break; + case 112896000: /* 44.1k */ + mask = RM_PLLCTL1C_PDB_PLL2; + val = RV_PLLCTL1C_PDB_PLL2_ENABLE; + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unrecognized PLL output freq (%d)\n", ret); + return ret; + } + + ret = snd_soc_update_bits(codec, R_PLLCTL1C, mask, val); + if (ret < 0) { + dev_err(codec->dev, "Failed to turn PLL on (%d)\n", ret); + return ret; + } + + if (!plls_locked(codec)) { + dev_err(codec->dev, "Failed to lock plls\n"); + return -ENOMSG; + } + + tscs42xx->pll_users++; + + return 0; +} + +static int tscs42xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = setup_sample_format(codec, params_format(params)); + if (ret < 0) { + dev_err(codec->dev, "Failed to setup sample format (%d)\n", + ret); + goto exit; + } + + ret = setup_sample_rate(codec, params_rate(params), tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to setup sample rate (%d)\n", ret); + goto exit; + } + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int dac_mute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = snd_soc_update_bits(codec, R_CNVRTR1, RM_CNVRTR1_DACMU, + RV_CNVRTR1_DACMU_ENABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to mute DAC (%d)\n", + ret); + return ret; + } + + ret = power_down_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power down plls (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int dac_unmute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = power_up_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power up plls (%d)\n", + ret); + return ret; + } + + ret = snd_soc_update_bits(codec, R_CNVRTR1, RM_CNVRTR1_DACMU, + RV_CNVRTR1_DACMU_DISABLE); + if (ret < 0) { + power_down_audio_plls(codec, tscs42xx); + dev_err(codec->dev, "Failed to unmute DAC (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int adc_mute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = snd_soc_update_bits(codec, R_CNVRTR0, RM_CNVRTR0_ADCMU, + RV_CNVRTR0_ADCMU_ENABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to mute ADC (%d)\n", + ret); + return ret; + } + + ret = power_down_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power down plls (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int adc_unmute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = power_up_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power up plls (%d)\n", + ret); + return ret; + } + + ret = snd_soc_update_bits(codec, R_CNVRTR0, RM_CNVRTR0_ADCMU, + RV_CNVRTR0_ADCMU_DISABLE); + if (ret < 0) { + power_down_audio_plls(codec, tscs42xx); + dev_err(codec->dev, "Failed to unmute ADC (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int tscs42xx_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&tscs42xx->lock); + + if (mute) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = dac_mute(codec, tscs42xx); + else + ret = adc_mute(codec, tscs42xx); + else + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = dac_unmute(codec, tscs42xx); + else + ret = adc_unmute(codec, tscs42xx); + + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ret = snd_soc_update_bits(codec, R_AIC1, RM_AIC1_MS, + RV_AIC1_MS_MASTER); + if (ret < 0) + dev_err(codec->dev, + "Failed to set codec DAI master (%d)\n", ret); + else + ret = 0; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ret = -EINVAL; + dev_err(codec->dev, "tscs42xx slave mode not supported (%d)\n", + ret); + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unsupported format (%d)\n", ret); + break; + } + + return ret; +} + +static int tscs42xx_set_bclk_ratio(struct snd_soc_dai *codec_dai, + unsigned int ratio) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int ret = 0; + + mutex_lock(&tscs42xx->lock); + + switch (ratio) { + case 32: + value = RV_DACSR_DBCM_32; + break; + case 40: + value = RV_DACSR_DBCM_40; + break; + case 64: + value = RV_DACSR_DBCM_64; + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unsupported bclk ratio (%d)\n", ret); + goto exit; + } + + ret = snd_soc_update_bits(codec, R_DACSR, RM_DACSR_DBCM, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + goto exit; + } + ret = snd_soc_update_bits(codec, R_ADCSR, RM_ADCSR_ABCM, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + goto exit; + } + + tscs42xx->bclk_ratio = ratio; + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver tscs42xx_dai = { + .name = "tscs42xx-HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TSCS42XX_RATES, + .formats = TSCS42XX_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = TSCS42XX_RATES, + .formats = TSCS42XX_FORMATS,}, + .ops = &tscs42xx_dai_ops, + .symmetric_rates = 1, +}; + +static int tscs42xx_i2c_read(struct i2c_client *i2c, u8 reg, u8 *val) +{ + int ret; + + ret = i2c_smbus_write_byte(i2c, reg); + if (ret < 0) { + dev_err(&i2c->dev, "I2C write failed (%d)\n", ret); + return ret; + } + + ret = i2c_smbus_read_byte(i2c); + if (ret < 0) { + dev_err(&i2c->dev, "I2C read failed (%d)\n", ret); + return ret; + } + + *val = (u8)ret; + + return 0; +} + +static int part_is_valid(struct i2c_client *i2c) +{ + int val; + int ret; + u8 reg; + + ret = tscs42xx_i2c_read(i2c, R_DEVIDH, &reg); + if (ret < 0) + return ret; + + val = reg << 8; + ret = tscs42xx_i2c_read(i2c, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + ret = true; + break; + default: + ret = false; + break; + }; + + if (ret) + dev_info(&i2c->dev, "Found part 0x%04x\n", val); + else + dev_err(&i2c->dev, "0x%04x is not a valid part\n", val); + + return ret; +} + +static int set_data_from_of(struct i2c_client *i2c, + struct tscs42xx_priv *tscs42xx) +{ + struct device_node *np = i2c->dev.of_node; + const char *mclk_src = NULL; + int ret; + + ret = of_property_read_string(np, "mclk-src", &mclk_src); + if (ret) { + dev_err(&i2c->dev, "mclk-src is needed (%d)\n", ret); + return ret; + } + + if (!strncmp(mclk_src, "mclk", 4)) { + tscs42xx->mclk = devm_clk_get(&i2c->dev, NULL); + if (IS_ERR(tscs42xx->mclk)) { + dev_info(&i2c->dev, "mclk not present trying again\n"); + return -EPROBE_DEFER; + } + tscs42xx->pll_src_clk = PLL_SRC_CLK_MCLK2; + } else if (!strncmp(mclk_src, "xtal", 4)) { + tscs42xx->pll_src_clk = PLL_SRC_CLK_XTAL; + } else { + dev_err(&i2c->dev, "mclk-src %s is unsupported\n", mclk_src); + return -EINVAL; + } + + ret = of_property_read_u32(np, "mclk-src-freq", + &tscs42xx->mclk_src_freq); + if (ret) { + dev_err(&i2c->dev, "mclk-src-freq not provided (%d)\n", ret); + return ret; + } + if (!get_pll_ctl(tscs42xx->mclk_src_freq)) { + dev_err(&i2c->dev, "mclk frequency unsupported\n"); + return ret; + } + + return 0; +} + +/******************* + * SYSFS Interface * + *******************/ +#define FMODE 0664 + +/* Control Reg Interface */ + +struct tempo_control_reg { + struct device *dev; + struct kobject *dir_kobj; + struct kobj_attribute val_kobj_attr; + struct kobj_attribute addr_kobj_attr; + const char *name; + const uint8_t addr; +}; + +static ssize_t ctrl_reg_val_show(struct kobject *kobj, + struct kobj_attribute *attr, char *buf) +{ + struct tempo_control_reg *control_reg = + container_of(attr, struct tempo_control_reg, val_kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control_reg->dev); + unsigned int show; + int ret = 0; + + ret = regmap_read(tscs42xx->regmap, + control_reg->addr, + &show); + if (ret < 0) + return ret; + + ret = sprintf(buf, "0x%02x\n", show); + + return ret; +} + +static ssize_t ctrl_reg_val_store(struct kobject *kobj, + struct kobj_attribute *attr, + const char *buf, size_t count) +{ + struct tempo_control_reg *control_reg = + container_of(attr, struct tempo_control_reg, val_kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control_reg->dev); + unsigned int store; + int ret; + + ret = kstrtoint(buf, 0, &store); + if (ret < 0) + return ret; + + ret = regmap_write(tscs42xx->regmap, control_reg->addr, store); + if (ret < 0) + return ret; + + return count; +} + +static ssize_t ctrl_reg_addr_show(struct kobject *kobj, + struct kobj_attribute *attr, char *buf) +{ + struct tempo_control_reg *control_reg = + container_of(attr, struct tempo_control_reg, addr_kobj_attr); + + return sprintf(buf, "0x%02x\n", control_reg->addr); +} + +#define TEMPO_CONTROL_REG(_name, _addr) \ + { \ + NULL, \ + NULL, \ + __ATTR(value, FMODE, ctrl_reg_val_show, \ + ctrl_reg_val_store), \ + __ATTR(address, FMODE, ctrl_reg_addr_show, \ + NULL), \ + .name = __stringify(_name), \ + .addr = _addr, \ + } + +static struct tempo_control_reg control_regs[] = { + TEMPO_CONTROL_REG(config0, R_CONFIG0), /* 0x1F */ + TEMPO_CONTROL_REG(config1, R_CONFIG1), /* 0x20 */ + TEMPO_CONTROL_REG(clectl, R_CLECTL), /* 0x25 */ + TEMPO_CONTROL_REG(mugain, R_MUGAIN), /* 0x26 */ + TEMPO_CONTROL_REG(compth, R_COMPTH), /* 0x27 */ + TEMPO_CONTROL_REG(cmprat, R_CMPRAT), /* 0x28 */ + TEMPO_CONTROL_REG(catktcl, R_CATKTCL), /* 0x29 */ + TEMPO_CONTROL_REG(catktch, R_CATKTCH), /* 0x2A */ + TEMPO_CONTROL_REG(creltcl, R_CRELTCL), /* 0x2B */ + TEMPO_CONTROL_REG(creltch, R_CRELTCH), /* 0x2C */ + TEMPO_CONTROL_REG(limth, R_LIMTH), /* 0x2D */ + TEMPO_CONTROL_REG(limtgt, R_LIMTGT), /* 0x2E */ + TEMPO_CONTROL_REG(latktcl, R_LATKTCL), /* 0x2F */ + TEMPO_CONTROL_REG(latktch, R_LATKTCH), /* 0x30 */ + TEMPO_CONTROL_REG(lreltcl, R_LRELTCL), /* 0x31 */ + TEMPO_CONTROL_REG(lreltch, R_LRELTCH), /* 0x32 */ + TEMPO_CONTROL_REG(expth, R_EXPTH), /* 0x33 */ + TEMPO_CONTROL_REG(exprat, R_EXPRAT), /* 0x34 */ + TEMPO_CONTROL_REG(xatktcl, R_XATKTCL), /* 0x35 */ + TEMPO_CONTROL_REG(xatktch, R_XATKTCH), /* 0x36 */ + TEMPO_CONTROL_REG(xreltcl, R_XRELTCL), /* 0x37 */ + TEMPO_CONTROL_REG(xreltch, R_XRELTCH), /* 0x38 */ + TEMPO_CONTROL_REG(fxctl, R_FXCTL), /* 0x39 */ + TEMPO_CONTROL_REG(daccrwrl, R_DACCRWRL), /* 0x3A */ + TEMPO_CONTROL_REG(daccrwrm, R_DACCRWRM), /* 0x3B */ + TEMPO_CONTROL_REG(daccrwrh, R_DACCRWRH), /* 0x3C */ + TEMPO_CONTROL_REG(daccrrdl, R_DACCRRDL), /* 0x3D */ + TEMPO_CONTROL_REG(daccrrdm, R_DACCRRDM), /* 0x3E */ + TEMPO_CONTROL_REG(daccrrdh, R_DACCRRDH), /* 0x3F */ + TEMPO_CONTROL_REG(daccraddr, R_DACCRADDR), /* 0x40 */ + TEMPO_CONTROL_REG(dcofsel, R_DCOFSEL), /* 0x41 */ + TEMPO_CONTROL_REG(daccrstat, R_DACCRSTAT), /* 0x8A */ + TEMPO_CONTROL_REG(dacmbcen, R_DACMBCEN), /* 0xC7 */ + TEMPO_CONTROL_REG(dacmbcctl, R_DACMBCCTL), /* 0xC8 */ + TEMPO_CONTROL_REG(dacmbcmug1, R_DACMBCMUG1), /* 0xC9 */ + TEMPO_CONTROL_REG(dacmbcthr1, R_DACMBCTHR1), /* 0xCA */ + TEMPO_CONTROL_REG(dacmbcrat1, R_DACMBCRAT1), /* 0xCB */ + TEMPO_CONTROL_REG(dacmbcatk1l, R_DACMBCATK1L), /* 0xCC */ + TEMPO_CONTROL_REG(dacmbcatk1h, R_DACMBCATK1H), /* 0xCD */ + TEMPO_CONTROL_REG(dacmbcrel1l, R_DACMBCREL1L), /* 0xCE */ + TEMPO_CONTROL_REG(dacmbcrel1h, R_DACMBCREL1H), /* 0xCF */ + TEMPO_CONTROL_REG(dacmbcmug2, R_DACMBCMUG2), /* 0xD0 */ + TEMPO_CONTROL_REG(dacmbcthr2, R_DACMBCTHR2), /* 0xD1 */ + TEMPO_CONTROL_REG(dacmbcrat2, R_DACMBCRAT2), /* 0xD2 */ + TEMPO_CONTROL_REG(dacmbcatk2l, R_DACMBCATK2L), /* 0xD3 */ + TEMPO_CONTROL_REG(dacmbcatk2h, R_DACMBCATK2H), /* 0xD4 */ + TEMPO_CONTROL_REG(dacmbcrel2l, R_DACMBCREL2L), /* 0xD5 */ + TEMPO_CONTROL_REG(dacmbcrel2h, R_DACMBCREL2H), /* 0xD6 */ + TEMPO_CONTROL_REG(dacmbcmug3, R_DACMBCMUG3), /* 0xD7 */ + TEMPO_CONTROL_REG(dacmbcthr3, R_DACMBCTHR3), /* 0xD8 */ + TEMPO_CONTROL_REG(dacmbcrat3, R_DACMBCRAT3), /* 0xD9 */ + TEMPO_CONTROL_REG(dacmbcatk3l, R_DACMBCATK3L), /* 0xDA */ + TEMPO_CONTROL_REG(dacmbcatk3h, R_DACMBCATK3H), /* 0xDB */ + TEMPO_CONTROL_REG(dacmbcrel3l, R_DACMBCREL3L), /* 0xDC */ + TEMPO_CONTROL_REG(dacmbcrel3h, R_DACMBCREL3H), /* 0xDD */ +}; + +static ssize_t control_reg_export_store(struct kobject *kobj, + struct kobj_attribute *attr, + const char *buf, size_t count) +{ + int i; + int ret; + char *nl; + + /* Strip the newline */ + nl = strchr(buf, '\n'); + if (nl != NULL) + *nl = '\0'; + + for (i = 0; i < ARRAY_SIZE(control_regs); i++) { + if (strcmp(buf, control_regs[i].name) == 0) { + + if (control_regs[i].dir_kobj) + break; + + control_regs[i].dev = kobj_to_dev(kobj->parent); + if (!control_regs[i].dev) + return -ENODEV; + control_regs[i].dir_kobj = kobject_create_and_add( + control_regs[i].name, kobj); + if (!control_regs[i].dir_kobj) + return -ENOMEM; + ret = sysfs_create_file(control_regs[i].dir_kobj, + &control_regs[i].val_kobj_attr.attr); + if (ret < 0) + return ret; + ret = sysfs_create_file(control_regs[i].dir_kobj, + &control_regs[i].addr_kobj_attr.attr); + if (ret < 0) + return ret; + + return count; + } + } + + return count; +} + +static ssize_t control_reg_unexport_store(struct kobject *kobj, + struct kobj_attribute *attr, + const char *buf, size_t count) +{ + int i; + char *nl; + + /* Strip the newline */ + nl = strchr(buf, '\n'); + if (nl != NULL) + *nl = '\0'; + + for (i = 0; i < ARRAY_SIZE(control_regs); i++) { + if (strcmp(buf, control_regs[i].name) == 0) { + if (control_regs[i].dir_kobj) { + sysfs_remove_file(control_regs[i].dir_kobj, + &control_regs[i].val_kobj_attr.attr); + sysfs_remove_file(control_regs[i].dir_kobj, + &control_regs[i].addr_kobj_attr.attr); + kobject_put(control_regs[i].dir_kobj); + control_regs[i].dir_kobj = NULL; + return count; + } + } + } + + return count; +} + +static struct kobj_attribute control_reg_export = + __ATTR(export, 0664, NULL, control_reg_export_store); +static struct kobj_attribute control_reg_unexport = + __ATTR(unexport, 0664, NULL, control_reg_unexport_store); + +/* Control Interface */ + +struct tempo_control { + struct device *dev; + struct kobj_attribute kobj_attr; + uint8_t addr; + uint8_t mask; + uint8_t shift; +}; + +static ssize_t ctrl_show(struct kobject *kobj, struct kobj_attribute *attr, + char *buf) +{ + struct tempo_control *control = + container_of(attr, struct tempo_control, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control->dev); + unsigned int val, show; + int ret; + + ret = regmap_read(tscs42xx->regmap, + control->addr, + &val); + if (ret < 0) + return ret; + show = (val & control->mask) >> control->shift; + + return sprintf(buf, "0x%02x\n", show); +} + +static ssize_t ctrl_store(struct kobject *kobj, struct kobj_attribute *attr, + const char *buf, size_t count) +{ + struct tempo_control *control = + container_of(attr, struct tempo_control, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control->dev); + unsigned int store; + int ret; + + ret = kstrtoint(buf, 0, &store); + if (ret < 0) + return ret; + + store = store << control->shift; + ret = regmap_update_bits(tscs42xx->regmap, + control->addr, control->mask, store); + if (ret < 0) + return ret; + + return count; +} + +#define TEMPO_CONTROL(_name, _addr, _mask, _shift) \ + { \ + NULL, \ + __ATTR(_name, FMODE, ctrl_show, ctrl_store), \ + .addr = _addr, \ + .mask = _mask, \ + .shift = _shift, \ + } + +/* Coefficient Interface */ + +struct tempo_coefficient { + struct device *dev; + struct kobj_attribute kobj_attr; + uint8_t addr; +}; + +static int enable_daccram_access(struct tscs42xx_priv *tscs42xx) +{ + struct snd_soc_codec *codec = tscs42xx->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + /* DAC needs to be powered */ + ret = snd_soc_dapm_force_enable_pin(dapm, "DAC L"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable DAC for DACCRAM access (%d)\n", ret); + return ret; + } + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync dapm context (%d)\n", ret); + return ret; + } + + /* If no one is using the PLL make sure there is a valid rate */ + if (tscs42xx->pll_users == 0) + tscs42xx->samplerate = 48000; + + return power_up_audio_plls(tscs42xx->codec, tscs42xx); +} + +static int disable_daccram_access(struct tscs42xx_priv *tscs42xx) +{ + struct snd_soc_codec *codec = tscs42xx->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + /* DAC needs to be powered */ + ret = snd_soc_dapm_disable_pin(dapm, "DAC L"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to disable DAC after DACCRAM access (%d)\n", + ret); + return ret; + } + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync dapm context (%d)\n", ret); + return ret; + } + + return power_down_audio_plls(tscs42xx->codec, tscs42xx); +} + +static ssize_t cff_show(struct kobject *kobj, struct kobj_attribute *attr, + char *buf) +{ + struct tempo_coefficient *coefficient = + container_of(attr, struct tempo_coefficient, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(coefficient->dev); + unsigned int val, show; + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = enable_daccram_access(tscs42xx); + if (ret < 0) + goto early_exit; + + ret = regmap_write(tscs42xx->regmap, + R_DACCRADDR, + coefficient->addr); + if (ret < 0) + goto exit; + ret = regmap_read(tscs42xx->regmap, + R_DACCRRDL, + &val); + if (ret < 0) + goto exit; + show = val; + ret = regmap_read(tscs42xx->regmap, + R_DACCRRDM, + &val); + if (ret < 0) + goto exit; + show |= (val << 8); + ret = regmap_read(tscs42xx->regmap, + R_DACCRRDH, + &val); + if (ret < 0) + goto exit; + show |= (val << 16); + + ret = sprintf(buf, "0x%06x\n", show); +exit: + disable_daccram_access(tscs42xx); + +early_exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static ssize_t cff_store(struct kobject *kobj, struct kobj_attribute *attr, + const char *buf, size_t count) +{ + struct tempo_coefficient *coefficient = + container_of(attr, struct tempo_coefficient, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(coefficient->dev); + unsigned int val, store; + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = enable_daccram_access(tscs42xx); + if (ret < 0) + goto early_exit; + + ret = kstrtoint(buf, 0, &store); + if (ret < 0) + goto exit; + + ret = regmap_write(tscs42xx->regmap, + R_DACCRADDR, + coefficient->addr); + if (ret < 0) + goto exit; + val = store & 0xff; + ret = regmap_write(tscs42xx->regmap, + R_DACCRWRL, + val); + if (ret < 0) + goto exit; + val = (store >> 8) & 0xff; + ret = regmap_write(tscs42xx->regmap, + R_DACCRWRM, + val); + if (ret < 0) + goto exit; + val = (store >> 16) & 0xff; + ret = regmap_write(tscs42xx->regmap, + R_DACCRWRH, + val); + if (ret < 0) + goto exit; + + ret = count; +exit: + disable_daccram_access(tscs42xx); + +early_exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +#define TEMPO_COEFFICIENT(_name, _addr) \ + { \ + NULL, \ + __ATTR(_name, FMODE, cff_show, cff_store), \ + .addr = _addr, \ + } + +#define BQC_COUNT 5 + +struct tempo_biquad { + struct tempo_coefficient coefficients[BQC_COUNT]; +}; + +#define TEMPO_BIQUAD(name, addr) \ + { \ + .coefficients = { \ + TEMPO_COEFFICIENT(name ## _b0, ((addr) + 0)), \ + TEMPO_COEFFICIENT(name ## _b1, ((addr) + 1)), \ + TEMPO_COEFFICIENT(name ## _b2, ((addr) + 2)), \ + TEMPO_COEFFICIENT(name ## _a1, ((addr) + 3)), \ + TEMPO_COEFFICIENT(name ## _a2, ((addr) + 4)) \ + }, \ + } + +static struct tempo_biquad dsp_biquads[] = { + /* EQ1 */ + TEMPO_BIQUAD(eq1_ch0_band1, 0x00), + TEMPO_BIQUAD(eq1_ch0_band2, 0x05), + TEMPO_BIQUAD(eq1_ch0_band3, 0x0a), + TEMPO_BIQUAD(eq1_ch0_band4, 0x0f), + TEMPO_BIQUAD(eq1_ch0_band5, 0x14), + TEMPO_BIQUAD(eq1_ch0_band6, 0x19), + TEMPO_BIQUAD(eq1_ch1_band1, 0x20), + TEMPO_BIQUAD(eq1_ch1_band2, 0x25), + TEMPO_BIQUAD(eq1_ch1_band3, 0x2a), + TEMPO_BIQUAD(eq1_ch1_band4, 0x2f), + TEMPO_BIQUAD(eq1_ch1_band5, 0x34), + TEMPO_BIQUAD(eq1_ch1_band6, 0x39), + /* EQ2 */ + TEMPO_BIQUAD(eq2_ch0_band1, 0x40), + TEMPO_BIQUAD(eq2_ch0_band2, 0x45), + TEMPO_BIQUAD(eq2_ch0_band3, 0x4a), + TEMPO_BIQUAD(eq2_ch0_band4, 0x4f), + TEMPO_BIQUAD(eq2_ch0_band5, 0x54), + TEMPO_BIQUAD(eq2_ch0_band6, 0x59), + TEMPO_BIQUAD(eq2_ch1_band1, 0x60), + TEMPO_BIQUAD(eq2_ch1_band2, 0x65), + TEMPO_BIQUAD(eq2_ch1_band3, 0x6a), + TEMPO_BIQUAD(eq2_ch1_band4, 0x6f), + TEMPO_BIQUAD(eq2_ch1_band5, 0x74), + TEMPO_BIQUAD(eq2_ch1_band6, 0x79), + /* Bass */ + TEMPO_BIQUAD(bass_ext1, 0x80), + TEMPO_BIQUAD(bass_ext2, 0x85), + TEMPO_BIQUAD(bass_lmt, 0x8c), + TEMPO_BIQUAD(bass_cto, 0x91), + /* Treble */ + TEMPO_BIQUAD(treb_ext1, 0x97), + TEMPO_BIQUAD(treb_ext2, 0x9c), + TEMPO_BIQUAD(treb_lmt, 0xa3), + TEMPO_BIQUAD(treb_cto, 0xa8), + /* Multi Band Compressor */ + TEMPO_BIQUAD(mbc_1_bq1, 0xb0), + TEMPO_BIQUAD(mbc_1_bq2, 0xb5), + TEMPO_BIQUAD(mbc_2_bq1, 0xba), + TEMPO_BIQUAD(mbc_2_bq2, 0xbf), + TEMPO_BIQUAD(mbc_3_bq1, 0xc4), + TEMPO_BIQUAD(mbc_3_bq2, 0xc9), +}; + +static struct tempo_coefficient gen_coefficients[] = { + /* 3D */ + TEMPO_COEFFICIENT(3d_coef, 0xae), + TEMPO_COEFFICIENT(3d_mix, 0xaf), + /* EQ1 */ + TEMPO_COEFFICIENT(eq1_ch0_prescale, 0x1f), + TEMPO_COEFFICIENT(eq1_ch1_prescale, 0x3f), + /* EQ2 */ + TEMPO_COEFFICIENT(eq2_ch0_prescale, 0x5f), + TEMPO_COEFFICIENT(eq2_ch1_prescale, 0x7f), + /* Bass */ + TEMPO_COEFFICIENT(bass_nlf_m1, 0x8a), + TEMPO_COEFFICIENT(bass_nlf_m2, 0x8b), + TEMPO_COEFFICIENT(bass_mix, 0x96), + /* Treble */ + TEMPO_COEFFICIENT(treb_nlf_m1, 0xa1), + TEMPO_COEFFICIENT(treb_nlf_m2, 0xa2), + TEMPO_COEFFICIENT(treb_mix, 0xad), +}; + +static struct tempo_control controls[] = { + TEMPO_CONTROL(eq1_en, R_CONFIG1, + RM_CONFIG1_EQ1_EN, FB_CONFIG1_EQ1_EN), + + TEMPO_CONTROL(eq1_bands_en, R_CONFIG1, + RM_CONFIG1_EQ1_BE, FB_CONFIG1_EQ1_BE), + + TEMPO_CONTROL(eq2_en, R_CONFIG1, + RM_CONFIG1_EQ2_EN, FB_CONFIG1_EQ2_EN), + + TEMPO_CONTROL(eq2_bands_en, R_CONFIG1, + RM_CONFIG1_EQ2_BE, FB_CONFIG1_EQ2_BE), + + TEMPO_CONTROL(cle_level_mode, R_CLECTL, + RM_CLECTL_LVL_MODE, FB_CLECTL_LVL_MODE), + + TEMPO_CONTROL(cle_level_detect_window, R_CLECTL, + RM_CLECTL_WINDOWSEL, FB_CLECTL_WINDOWSEL), + + TEMPO_CONTROL(exp_en, R_CLECTL, RM_CLECTL_EXP_EN, FB_CLECTL_EXP_EN), + + TEMPO_CONTROL(limit_en, R_CLECTL, RM_CLECTL_LIMIT_EN, + FB_CLECTL_LIMIT_EN), + + TEMPO_CONTROL(comp_en, R_CLECTL, RM_CLECTL_COMP_EN, FB_CLECTL_COMP_EN), + + TEMPO_CONTROL(3d_en, R_FXCTL, RM_FXCTL_3DEN, FB_FXCTL_3DEN), + + TEMPO_CONTROL(treb_en, R_FXCTL, RM_FXCTL_TEEN, FB_FXCTL_TEEN), + + TEMPO_CONTROL(treb_nlf_bypass, R_FXCTL, RM_FXCTL_TNLFBYPASS, + FB_FXCTL_TNLFBYPASS), + + TEMPO_CONTROL(bass_en, R_FXCTL, RM_FXCTL_BEEN, FB_FXCTL_BEEN), + + TEMPO_CONTROL(bass_nlf_bypass, R_FXCTL, RM_FXCTL_BNLFBYPASS, + FB_FXCTL_BNLFBYPASS), +}; + +static int create_sysfs_interface(struct kobject *parent_kobj) +{ + struct kobject *dsp_kobj; + struct kobject *controls_kobj; + struct kobject *control_regs_dir_kobj; + int i, j; + int ret; + + dsp_kobj = kobject_create_and_add("dsp", parent_kobj); + if (!dsp_kobj) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(dsp_biquads); i++) { + for (j = 0; j < BQC_COUNT; j++) { + dsp_biquads[i].coefficients[j].dev = + kobj_to_dev(parent_kobj); + if (!dsp_biquads[i].coefficients[j].dev) + return -ENODEV; + ret = sysfs_create_file(dsp_kobj, + &dsp_biquads[i].coefficients[j].kobj_attr.attr); + if (ret < 0) + return ret; + } + } + + for (i = 0; i < ARRAY_SIZE(gen_coefficients); i++) { + gen_coefficients[i].dev = kobj_to_dev(parent_kobj); + if (!gen_coefficients[i].dev) + return -ENODEV; + ret = sysfs_create_file(dsp_kobj, + &gen_coefficients[i].kobj_attr.attr); + if (ret < 0) + return ret; + } + + controls_kobj = kobject_create_and_add("controls", + parent_kobj); + if (!controls_kobj) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(controls); i++) { + controls[i].dev = kobj_to_dev(parent_kobj); + if (!controls[i].dev) + return -ENODEV; + ret = sysfs_create_file(controls_kobj, + &controls[i].kobj_attr.attr); + if (ret < 0) + return ret; + } + + control_regs_dir_kobj = kobject_create_and_add("control_regs", + parent_kobj); + + ret = sysfs_create_file(control_regs_dir_kobj, + &control_reg_export.attr); + if (ret < 0) + return ret; + + ret = sysfs_create_file(control_regs_dir_kobj, + &control_reg_unexport.attr); + if (ret < 0) + return ret; + + return 0; +} + +static int tscs42xx_probe(struct snd_soc_codec *codec) +{ + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + int i; + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = create_sysfs_interface(&codec->dev->kobj); + if (ret < 0) { + dev_info(codec->dev, "Failed to create dsp interface (%d)\n", + ret); + goto exit; + } + + tscs42xx->codec = codec; + + ret = configure_clocks(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to configure clocks (%d)\n", ret); + goto exit; + } + + for (i = 0; i < ARRAY_SIZE(r_inits); ++i) { + ret = snd_soc_write(codec, r_inits[i].reg, r_inits[i].def); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write codec defaults (%d)\n", ret); + goto exit; + } + } + + /* Power up an interface so the daccram can be accessed */ + ret = snd_soc_update_bits(codec, R_PWRM2, RM_PWRM2_HPL, + RV_PWRM2_HPL_ENABLE); + if (ret < 0) { + dev_err(codec->dev, + "Failed to power up interface (%d)\n", ret); + goto exit; + } + + /* PLLs also needed to be powered */ + tscs42xx->samplerate = 48000; /* No valid rate exist yet */ + ret = power_up_audio_plls(codec, tscs42xx); + if (ret < 0) { + goto exit; + snd_soc_update_bits(codec, R_PWRM2, + RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + } + + ret = load_dac_coefficient_ram(codec); + if (ret < 0) { + dev_info(codec->dev, "Failed to load DAC Coefficients (%d)\n", + ret); + power_down_audio_plls(codec, tscs42xx); + snd_soc_update_bits(codec, R_PWRM2, + RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + goto exit; + } + + ret = load_control_regs(codec); + if (ret < 0) { + dev_info(codec->dev, "Failed to load controls (%d)\n", + ret); + power_down_audio_plls(codec, tscs42xx); + snd_soc_update_bits(codec, R_PWRM2, + RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + goto exit; + } + + power_down_audio_plls(codec, tscs42xx); + snd_soc_update_bits(codec, R_PWRM2, RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int tscs42xx_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, + .remove = tscs42xx_remove, + .component_driver = { + .dapm_widgets = tscs42xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), + .dapm_routes = tscs42xx_intercon, + .num_dapm_routes = ARRAY_SIZE(tscs42xx_intercon), + .controls = tscs42xx_snd_controls, + .num_controls = ARRAY_SIZE(tscs42xx_snd_controls), + }, +}; + +static int tscs42xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tscs42xx_priv *tscs42xx; + int ret = 0; + + tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); + if (!tscs42xx) + return -ENOMEM; + + mutex_init(&tscs42xx->lock); + mutex_lock(&tscs42xx->lock); + + tscs42xx->pll_users = 0; + + ret = set_data_from_of(i2c, tscs42xx); + if (ret < 0) { + dev_err(&i2c->dev, "Error parsing device tree info (%d)", ret); + goto exit; + } + + ret = part_is_valid(i2c); + if (ret <= 0) { + dev_err(&i2c->dev, "No valid part (%d)\n", ret); + ret = -ENODEV; + goto exit; + } + + /* Reset device */ + ret = i2c_smbus_write_byte_data(i2c, R_RESET, + RV_RESET_ENABLE); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to reset device (%d)\n", ret); + goto exit; + } + + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); + if (IS_ERR(tscs42xx->regmap)) { + ret = PTR_ERR(tscs42xx->regmap); + dev_err(&i2c->dev, "Failed to allocat regmap (%d)\n", ret); + goto exit; + } + + i2c_set_clientdata(i2c, tscs42xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tscs42xx, + &tscs42xx_dai, 1); + if (ret) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + goto exit; + } + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int tscs42xx_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id tscs42xx_i2c_id[] = { + { "tscs42xx", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tscs42xx_i2c_id); + +static const struct of_device_id tscs42xx_of_match[] = { + { .compatible = "tscs,tscs42xx", }, + { } +}; +MODULE_DEVICE_TABLE(of, tscs42xx_of_match); + +static struct i2c_driver tscs42xx_i2c_driver = { + .driver = { + .name = "tscs42xx", + .owner = THIS_MODULE, + .of_match_table = tscs42xx_of_match, + }, + .probe = tscs42xx_i2c_probe, + .remove = tscs42xx_i2c_remove, + .id_table = tscs42xx_i2c_id, +}; + +static int __init tscs42xx_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&tscs42xx_i2c_driver); + if (ret < 0) + pr_err("Failed to register TSCS42xx I2C driver (%d)\n", ret); + + return ret; +} +module_init(tscs42xx_modinit); + +static void __exit tscs42xx_exit(void) +{ + i2c_del_driver(&tscs42xx_i2c_driver); +} +module_exit(tscs42xx_exit); + +MODULE_AUTHOR("Tempo Semiconductor <steven.eckhoff.opensource(a)gmail.com"); +MODULE_DESCRIPTION("ASoC TSCS42xx driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h new file mode 100644 index 000000000000..e38ff8c7b6c9 --- /dev/null +++ b/sound/soc/codecs/tscs42xx.h @@ -0,0 +1,2693 @@ +/* + * tscs42xx.c -- TSCS42xx ALSA SoC Audio driver + * + * Copyright 2017 Tempo Semiconductor, Inc. + * + * Author: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __WOOKIE_H__ +#define __WOOKIE_H__ + +#define R_HPVOLL 0x0 +#define R_HPVOLR 0x1 +#define R_SPKVOLL 0x2 +#define R_SPKVOLR 0x3 +#define R_DACVOLL 0x4 +#define R_DACVOLR 0x5 +#define R_ADCVOLL 0x6 +#define R_ADCVOLR 0x7 +#define R_INVOLL 0x8 +#define R_INVOLR 0x9 +#define R_INMODE 0x0B +#define R_INSELL 0x0C +#define R_INSELR 0x0D +#define R_AIC1 0x13 +#define R_AIC2 0x14 +#define R_CNVRTR0 0x16 +#define R_ADCSR 0x17 +#define R_CNVRTR1 0x18 +#define R_DACSR 0x19 +#define R_PWRM1 0x1A +#define R_PWRM2 0x1B +#define R_CONFIG0 0x1F +#define R_CONFIG1 0x20 +#define R_DMICCTL 0x24 +#define R_CLECTL 0x25 +#define R_MUGAIN 0x26 +#define R_COMPTH 0x27 +#define R_CMPRAT 0x28 +#define R_CATKTCL 0x29 +#define R_CATKTCH 0x2A +#define R_CRELTCL 0x2B +#define R_CRELTCH 0x2C +#define R_LIMTH 0x2D +#define R_LIMTGT 0x2E +#define R_LATKTCL 0x2F +#define R_LATKTCH 0x30 +#define R_LRELTCL 0x31 +#define R_LRELTCH 0x32 +#define R_EXPTH 0x33 +#define R_EXPRAT 0x34 +#define R_XATKTCL 0x35 +#define R_XATKTCH 0x36 +#define R_XRELTCL 0x37 +#define R_XRELTCH 0x38 +#define R_FXCTL 0x39 +#define R_DACCRWRL 0x3A +#define R_DACCRWRM 0x3B +#define R_DACCRWRH 0x3C +#define R_DACCRRDL 0x3D +#define R_DACCRRDM 0x3E +#define R_DACCRRDH 0x3F +#define R_DACCRADDR 0x40 +#define R_DCOFSEL 0x41 +#define R_PLLCTL9 0x4E +#define R_PLLCTLA 0x4F +#define R_PLLCTLB 0x50 +#define R_PLLCTLC 0x51 +#define R_PLLCTLD 0x52 +#define R_PLLCTLE 0x53 +#define R_PLLCTLF 0x54 +#define R_PLLCTL10 0x55 +#define R_PLLCTL11 0x56 +#define R_PLLCTL12 0x57 +#define R_PLLCTL1B 0x60 +#define R_PLLCTL1C 0x61 +#define R_TIMEBASE 0x77 +#define R_DEVIDL 0x7D +#define R_DEVIDH 0x7E +#define R_RESET 0x80 +#define R_DACCRSTAT 0x8A +#define R_PLLCTL0 0x8E +#define R_PLLREFSEL 0x8F +#define R_DACMBCEN 0xC7 +#define R_DACMBCCTL 0xC8 +#define R_DACMBCMUG1 0xC9 +#define R_DACMBCTHR1 0xCA +#define R_DACMBCRAT1 0xCB +#define R_DACMBCATK1L 0xCC +#define R_DACMBCATK1H 0xCD +#define R_DACMBCREL1L 0xCE +#define R_DACMBCREL1H 0xCF +#define R_DACMBCMUG2 0xD0 +#define R_DACMBCTHR2 0xD1 +#define R_DACMBCRAT2 0xD2 +#define R_DACMBCATK2L 0xD3 +#define R_DACMBCATK2H 0xD4 +#define R_DACMBCREL2L 0xD5 +#define R_DACMBCREL2H 0xD6 +#define R_DACMBCMUG3 0xD7 +#define R_DACMBCTHR3 0xD8 +#define R_DACMBCRAT3 0xD9 +#define R_DACMBCATK3L 0xDA +#define R_DACMBCATK3H 0xDB +#define R_DACMBCREL3L 0xDC +#define R_DACMBCREL3H 0xDD + +/* Helpers */ +#define RM(m, b) ((m)<<(b)) +#define RV(v, b) ((v)<<(b)) + +/**************************** + * R_HPVOLL (0x0) * + ****************************/ + +/* Field Offsets */ +#define FB_HPVOLL 0 + +/* Field Masks */ +#define FM_HPVOLL 0X7F + +/* Field Values */ +#define FV_HPVOLL_P6DB 0x7F +#define FV_HPVOLL_N88PT5DB 0x1 +#define FV_HPVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_HPVOLL RM(FM_HPVOLL, FB_HPVOLL) + +/* Register Values */ +#define RV_HPVOLL_P6DB RV(FV_HPVOLL_P6DB, FB_HPVOLL) +#define RV_HPVOLL_N88PT5DB RV(FV_HPVOLL_N88PT5DB, FB_HPVOLL) +#define RV_HPVOLL_MUTE RV(FV_HPVOLL_MUTE, FB_HPVOLL) + +/**************************** + * R_HPVOLR (0x1) * + ****************************/ + +/* Field Offsets */ +#define FB_HPVOLR 0 + +/* Field Masks */ +#define FM_HPVOLR 0X7F + +/* Field Values */ +#define FV_HPVOLR_P6DB 0x7F +#define FV_HPVOLR_N88PT5DB 0x1 +#define FV_HPVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_HPVOLR RM(FM_HPVOLR, FB_HPVOLR) + +/* Register Values */ +#define RV_HPVOLR_P6DB RV(FV_HPVOLR_P6DB, FB_HPVOLR) +#define RV_HPVOLR_N88PT5DB RV(FV_HPVOLR_N88PT5DB, FB_HPVOLR) +#define RV_HPVOLR_MUTE RV(FV_HPVOLR_MUTE, FB_HPVOLR) + +/***************************** + * R_SPKVOLL (0x2) * + *****************************/ + +/* Field Offsets */ +#define FB_SPKVOLL 0 + +/* Field Masks */ +#define FM_SPKVOLL 0X7F + +/* Field Values */ +#define FV_SPKVOLL_P12DB 0x7F +#define FV_SPKVOLL_N77PT25DB 0x8 +#define FV_SPKVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_SPKVOLL RM(FM_SPKVOLL, FB_SPKVOLL) + +/* Register Values */ +#define RV_SPKVOLL_P12DB RV(FV_SPKVOLL_P12DB, FB_SPKVOLL) +#define RV_SPKVOLL_N77PT25DB \ + RV(FV_SPKVOLL_N77PT25DB, FB_SPKVOLL) + +#define RV_SPKVOLL_MUTE RV(FV_SPKVOLL_MUTE, FB_SPKVOLL) + +/***************************** + * R_SPKVOLR (0x3) * + *****************************/ + +/* Field Offsets */ +#define FB_SPKVOLR 0 + +/* Field Masks */ +#define FM_SPKVOLR 0X7F + +/* Field Values */ +#define FV_SPKVOLR_P12DB 0x7F +#define FV_SPKVOLR_N77PT25DB 0x8 +#define FV_SPKVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_SPKVOLR RM(FM_SPKVOLR, FB_SPKVOLR) + +/* Register Values */ +#define RV_SPKVOLR_P12DB RV(FV_SPKVOLR_P12DB, FB_SPKVOLR) +#define RV_SPKVOLR_N77PT25DB \ + RV(FV_SPKVOLR_N77PT25DB, FB_SPKVOLR) + +#define RV_SPKVOLR_MUTE RV(FV_SPKVOLR_MUTE, FB_SPKVOLR) + +/***************************** + * R_DACVOLL (0x4) * + *****************************/ + +/* Field Offsets */ +#define FB_DACVOLL 0 + +/* Field Masks */ +#define FM_DACVOLL 0XFF + +/* Field Values */ +#define FV_DACVOLL_0DB 0xFF +#define FV_DACVOLL_N95PT625DB 0x1 +#define FV_DACVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_DACVOLL RM(FM_DACVOLL, FB_DACVOLL) + +/* Register Values */ +#define RV_DACVOLL_0DB RV(FV_DACVOLL_0DB, FB_DACVOLL) +#define RV_DACVOLL_N95PT625DB \ + RV(FV_DACVOLL_N95PT625DB, FB_DACVOLL) + +#define RV_DACVOLL_MUTE RV(FV_DACVOLL_MUTE, FB_DACVOLL) + +/***************************** + * R_DACVOLR (0x5) * + *****************************/ + +/* Field Offsets */ +#define FB_DACVOLR 0 + +/* Field Masks */ +#define FM_DACVOLR 0XFF + +/* Field Values */ +#define FV_DACVOLR_0DB 0xFF +#define FV_DACVOLR_N95PT625DB 0x1 +#define FV_DACVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_DACVOLR RM(FM_DACVOLR, FB_DACVOLR) + +/* Register Values */ +#define RV_DACVOLR_0DB RV(FV_DACVOLR_0DB, FB_DACVOLR) +#define RV_DACVOLR_N95PT625DB \ + RV(FV_DACVOLR_N95PT625DB, FB_DACVOLR) + +#define RV_DACVOLR_MUTE RV(FV_DACVOLR_MUTE, FB_DACVOLR) + +/***************************** + * R_ADCVOLL (0x6) * + *****************************/ + +/* Field Offsets */ +#define FB_ADCVOLL 0 + +/* Field Masks */ +#define FM_ADCVOLL 0XFF + +/* Field Values */ +#define FV_ADCVOLL_P24DB 0xFF +#define FV_ADCVOLL_N71PT25DB 0x1 +#define FV_ADCVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_ADCVOLL RM(FM_ADCVOLL, FB_ADCVOLL) + +/* Register Values */ +#define RV_ADCVOLL_P24DB RV(FV_ADCVOLL_P24DB, FB_ADCVOLL) +#define RV_ADCVOLL_N71PT25DB \ + RV(FV_ADCVOLL_N71PT25DB, FB_ADCVOLL) + +#define RV_ADCVOLL_MUTE RV(FV_ADCVOLL_MUTE, FB_ADCVOLL) + +/***************************** + * R_ADCVOLR (0x7) * + *****************************/ + +/* Field Offsets */ +#define FB_ADCVOLR 0 + +/* Field Masks */ +#define FM_ADCVOLR 0XFF + +/* Field Values */ +#define FV_ADCVOLR_P24DB 0xFF +#define FV_ADCVOLR_N71PT25DB 0x1 +#define FV_ADCVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_ADCVOLR RM(FM_ADCVOLR, FB_ADCVOLR) + +/* Register Values */ +#define RV_ADCVOLR_P24DB RV(FV_ADCVOLR_P24DB, FB_ADCVOLR) +#define RV_ADCVOLR_N71PT25DB \ + RV(FV_ADCVOLR_N71PT25DB, FB_ADCVOLR) + +#define RV_ADCVOLR_MUTE RV(FV_ADCVOLR_MUTE, FB_ADCVOLR) + +/**************************** + * R_INVOLL (0x8) * + ****************************/ + +/* Field Offsets */ +#define FB_INVOLL_INMUTEL 7 +#define FB_INVOLL_IZCL 6 +#define FB_INVOLL 0 + +/* Field Masks */ +#define FM_INVOLL_INMUTEL 0X1 +#define FM_INVOLL_IZCL 0X1 +#define FM_INVOLL 0X3F + +/* Field Values */ +#define FV_INVOLL_INMUTEL_ENABLE 0x1 +#define FV_INVOLL_INMUTEL_DISABLE 0x0 +#define FV_INVOLL_IZCL_ENABLE 0x1 +#define FV_INVOLL_IZCL_DISABLE 0x0 +#define FV_INVOLL_P30DB 0x3F +#define FV_INVOLL_N17PT25DB 0x0 + +/* Register Masks */ +#define RM_INVOLL_INMUTEL \ + RM(FM_INVOLL_INMUTEL, FB_INVOLL_INMUTEL) + +#define RM_INVOLL_IZCL RM(FM_INVOLL_IZCL, FB_INVOLL_IZCL) +#define RM_INVOLL RM(FM_INVOLL, FB_INVOLL) + +/* Register Values */ +#define RV_INVOLL_INMUTEL_ENABLE \ + RV(FV_INVOLL_INMUTEL_ENABLE, FB_INVOLL_INMUTEL) + +#define RV_INVOLL_INMUTEL_DISABLE \ + RV(FV_INVOLL_INMUTEL_DISABLE, FB_INVOLL_INMUTEL) + +#define RV_INVOLL_IZCL_ENABLE \ + RV(FV_INVOLL_IZCL_ENABLE, FB_INVOLL_IZCL) + +#define RV_INVOLL_IZCL_DISABLE \ + RV(FV_INVOLL_IZCL_DISABLE, FB_INVOLL_IZCL) + +#define RV_INVOLL_P30DB RV(FV_INVOLL_P30DB, FB_INVOLL) +#define RV_INVOLL_N17PT25DB RV(FV_INVOLL_N17PT25DB, FB_INVOLL) + +/**************************** + * R_INVOLR (0x9) * + ****************************/ + +/* Field Offsets */ +#define FB_INVOLR_INMUTER 7 +#define FB_INVOLR_IZCR 6 +#define FB_INVOLR 0 + +/* Field Masks */ +#define FM_INVOLR_INMUTER 0X1 +#define FM_INVOLR_IZCR 0X1 +#define FM_INVOLR 0X3F + +/* Field Values */ +#define FV_INVOLR_INMUTER_ENABLE 0x1 +#define FV_INVOLR_INMUTER_DISABLE 0x0 +#define FV_INVOLR_IZCR_ENABLE 0x1 +#define FV_INVOLR_IZCR_DISABLE 0x0 +#define FV_INVOLR_P30DB 0x3F +#define FV_INVOLR_N17PT25DB 0x0 + +/* Register Masks */ +#define RM_INVOLR_INMUTER \ + RM(FM_INVOLR_INMUTER, FB_INVOLR_INMUTER) + +#define RM_INVOLR_IZCR RM(FM_INVOLR_IZCR, FB_INVOLR_IZCR) +#define RM_INVOLR RM(FM_INVOLR, FB_INVOLR) + +/* Register Values */ +#define RV_INVOLR_INMUTER_ENABLE \ + RV(FV_INVOLR_INMUTER_ENABLE, FB_INVOLR_INMUTER) + +#define RV_INVOLR_INMUTER_DISABLE \ + RV(FV_INVOLR_INMUTER_DISABLE, FB_INVOLR_INMUTER) + +#define RV_INVOLR_IZCR_ENABLE \ + RV(FV_INVOLR_IZCR_ENABLE, FB_INVOLR_IZCR) + +#define RV_INVOLR_IZCR_DISABLE \ + RV(FV_INVOLR_IZCR_DISABLE, FB_INVOLR_IZCR) + +#define RV_INVOLR_P30DB RV(FV_INVOLR_P30DB, FB_INVOLR) +#define RV_INVOLR_N17PT25DB RV(FV_INVOLR_N17PT25DB, FB_INVOLR) + +/***************************** + * R_INMODE (0x0B) * + *****************************/ + +/* Field Offsets */ +#define FB_INMODE_DS 0 + +/* Field Masks */ +#define FM_INMODE_DS 0X1 + +/* Field Values */ +#define FV_INMODE_DS_LRIN1 0x0 +#define FV_INMODE_DS_LRIN2 0x1 + +/* Register Masks */ +#define RM_INMODE_DS RM(FM_INMODE_DS, FB_INMODE_DS) + +/* Register Values */ +#define RV_INMODE_DS_LRIN1 \ + RV(FV_INMODE_DS_LRIN1, FB_INMODE_DS) + +#define RV_INMODE_DS_LRIN2 \ + RV(FV_INMODE_DS_LRIN2, FB_INMODE_DS) + + +/***************************** + * R_INSELL (0x0C) * + *****************************/ + +/* Field Offsets */ +#define FB_INSELL 6 +#define FB_INSELL_MICBSTL 4 + +/* Field Masks */ +#define FM_INSELL 0X3 +#define FM_INSELL_MICBSTL 0X3 + +/* Field Values */ +#define FV_INSELL_IN1 0x0 +#define FV_INSELL_IN2 0x1 +#define FV_INSELL_IN3 0x2 +#define FV_INSELL_D2S 0x3 +#define FV_INSELL_MICBSTL_OFF 0x0 +#define FV_INSELL_MICBSTL_10DB 0x1 +#define FV_INSELL_MICBSTL_20DB 0x2 +#define FV_INSELL_MICBSTL_30DB 0x3 + +/* Register Masks */ +#define RM_INSELL RM(FM_INSELL, FB_INSELL) +#define RM_INSELL_MICBSTL \ + RM(FM_INSELL_MICBSTL, FB_INSELL_MICBSTL) + + +/* Register Values */ +#define RV_INSELL_IN1 RV(FV_INSELL_IN1, FB_INSELL) +#define RV_INSELL_IN2 RV(FV_INSELL_IN2, FB_INSELL) +#define RV_INSELL_IN3 RV(FV_INSELL_IN3, FB_INSELL) +#define RV_INSELL_D2S RV(FV_INSELL_D2S, FB_INSELL) +#define RV_INSELL_MICBSTL_OFF \ + RV(FV_INSELL_MICBSTL_OFF, FB_INSELL_MICBSTL) + +#define RV_INSELL_MICBSTL_10DB \ + RV(FV_INSELL_MICBSTL_10DB, FB_INSELL_MICBSTL) + +#define RV_INSELL_MICBSTL_20DB \ + RV(FV_INSELL_MICBSTL_20DB, FB_INSELL_MICBSTL) + +#define RV_INSELL_MICBSTL_30DB \ + RV(FV_INSELL_MICBSTL_30DB, FB_INSELL_MICBSTL) + + +/***************************** + * R_INSELR (0x0D) * + *****************************/ + +/* Field Offsets */ +#define FB_INSELR 6 +#define FB_INSELR_MICBSTR 4 + +/* Field Masks */ +#define FM_INSELR 0X3 +#define FM_INSELR_MICBSTR 0X3 + +/* Field Values */ +#define FV_INSELR_IN1 0x0 +#define FV_INSELR_IN2 0x1 +#define FV_INSELR_IN3 0x2 +#define FV_INSELR_D2S 0x3 +#define FV_INSELR_MICBSTR_OFF 0x0 +#define FV_INSELR_MICBSTR_10DB 0x1 +#define FV_INSELR_MICBSTR_20DB 0x2 +#define FV_INSELR_MICBSTR_30DB 0x3 + +/* Register Masks */ +#define RM_INSELR RM(FM_INSELR, FB_INSELR) +#define RM_INSELR_MICBSTR \ + RM(FM_INSELR_MICBSTR, FB_INSELR_MICBSTR) + + +/* Register Values */ +#define RV_INSELR_IN1 RV(FV_INSELR_IN1, FB_INSELR) +#define RV_INSELR_IN2 RV(FV_INSELR_IN2, FB_INSELR) +#define RV_INSELR_IN3 RV(FV_INSELR_IN3, FB_INSELR) +#define RV_INSELR_D2S RV(FV_INSELR_D2S, FB_INSELR) +#define RV_INSELR_MICBSTR_OFF \ + RV(FV_INSELR_MICBSTR_OFF, FB_INSELR_MICBSTR) + +#define RV_INSELR_MICBSTR_10DB \ + RV(FV_INSELR_MICBSTR_10DB, FB_INSELR_MICBSTR) + +#define RV_INSELR_MICBSTR_20DB \ + RV(FV_INSELR_MICBSTR_20DB, FB_INSELR_MICBSTR) + +#define RV_INSELR_MICBSTR_30DB \ + RV(FV_INSELR_MICBSTR_30DB, FB_INSELR_MICBSTR) + + +/*************************** + * R_AIC1 (0x13) * + ***************************/ + +/* Field Offsets */ +#define FB_AIC1_BCLKINV 6 +#define FB_AIC1_MS 5 +#define FB_AIC1_LRP 4 +#define FB_AIC1_WL 2 +#define FB_AIC1_FORMAT 0 + +/* Field Masks */ +#define FM_AIC1_BCLKINV 0X1 +#define FM_AIC1_MS 0X1 +#define FM_AIC1_LRP 0X1 +#define FM_AIC1_WL 0X3 +#define FM_AIC1_FORMAT 0X3 + +/* Field Values */ +#define FV_AIC1_BCLKINV_ENABLE 0x1 +#define FV_AIC1_BCLKINV_DISABLE 0x0 +#define FV_AIC1_MS_MASTER 0x1 +#define FV_AIC1_MS_SLAVE 0x0 +#define FV_AIC1_LRP_INVERT 0x1 +#define FV_AIC1_LRP_NORMAL 0x0 +#define FV_AIC1_WL_16 0x0 +#define FV_AIC1_WL_20 0x1 +#define FV_AIC1_WL_24 0x2 +#define FV_AIC1_WL_32 0x3 +#define FV_AIC1_FORMAT_RIGHT 0x0 +#define FV_AIC1_FORMAT_LEFT 0x1 +#define FV_AIC1_FORMAT_I2S 0x2 + +/* Register Masks */ +#define RM_AIC1_BCLKINV \ + RM(FM_AIC1_BCLKINV, FB_AIC1_BCLKINV) + +#define RM_AIC1_MS RM(FM_AIC1_MS, FB_AIC1_MS) +#define RM_AIC1_LRP RM(FM_AIC1_LRP, FB_AIC1_LRP) +#define RM_AIC1_WL RM(FM_AIC1_WL, FB_AIC1_WL) +#define RM_AIC1_FORMAT RM(FM_AIC1_FORMAT, FB_AIC1_FORMAT) + +/* Register Values */ +#define RV_AIC1_BCLKINV_ENABLE \ + RV(FV_AIC1_BCLKINV_ENABLE, FB_AIC1_BCLKINV) + +#define RV_AIC1_BCLKINV_DISABLE \ + RV(FV_AIC1_BCLKINV_DISABLE, FB_AIC1_BCLKINV) + +#define RV_AIC1_MS_MASTER RV(FV_AIC1_MS_MASTER, FB_AIC1_MS) +#define RV_AIC1_MS_SLAVE RV(FV_AIC1_MS_SLAVE, FB_AIC1_MS) +#define RV_AIC1_LRP_INVERT \ + RV(FV_AIC1_LRP_INVERT, FB_AIC1_LRP) + +#define RV_AIC1_LRP_NORMAL \ + RV(FV_AIC1_LRP_NORMAL, FB_AIC1_LRP) + +#define RV_AIC1_WL_16 RV(FV_AIC1_WL_16, FB_AIC1_WL) +#define RV_AIC1_WL_20 RV(FV_AIC1_WL_20, FB_AIC1_WL) +#define RV_AIC1_WL_24 RV(FV_AIC1_WL_24, FB_AIC1_WL) +#define RV_AIC1_WL_32 RV(FV_AIC1_WL_32, FB_AIC1_WL) +#define RV_AIC1_FORMAT_RIGHT \ + RV(FV_AIC1_FORMAT_RIGHT, FB_AIC1_FORMAT) + +#define RV_AIC1_FORMAT_LEFT \ + RV(FV_AIC1_FORMAT_LEFT, FB_AIC1_FORMAT) + +#define RV_AIC1_FORMAT_I2S \ + RV(FV_AIC1_FORMAT_I2S, FB_AIC1_FORMAT) + + +/*************************** + * R_AIC2 (0x14) * + ***************************/ + +/* Field Offsets */ +#define FB_AIC2_DACDSEL 6 +#define FB_AIC2_ADCDSEL 4 +#define FB_AIC2_TRI 3 +#define FB_AIC2_BLRCM 0 + +/* Field Masks */ +#define FM_AIC2_DACDSEL 0X3 +#define FM_AIC2_ADCDSEL 0X3 +#define FM_AIC2_TRI 0X1 +#define FM_AIC2_BLRCM 0X7 + +/* Field Values */ +#define FV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED 0x3 + +/* Register Masks */ +#define RM_AIC2_DACDSEL \ + RM(FM_AIC2_DACDSEL, FB_AIC2_DACDSEL) + +#define RM_AIC2_ADCDSEL \ + RM(FM_AIC2_ADCDSEL, FB_AIC2_ADCDSEL) + +#define RM_AIC2_TRI RM(FM_AIC2_TRI, FB_AIC2_TRI) +#define RM_AIC2_BLRCM RM(FM_AIC2_BLRCM, FB_AIC2_BLRCM) + +/* Register Values */ +#define RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED \ + RV(FV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED, FB_AIC2_BLRCM) + + +/****************************** + * R_CNVRTR0 (0x16) * + ******************************/ + +/* Field Offsets */ +#define FB_CNVRTR0_ADCPOLR 7 +#define FB_CNVRTR0_ADCPOLL 6 +#define FB_CNVRTR0_AMONOMIX 4 +#define FB_CNVRTR0_ADCMU 3 +#define FB_CNVRTR0_HPOR 2 +#define FB_CNVRTR0_ADCHPDR 1 +#define FB_CNVRTR0_ADCHPDL 0 + +/* Field Masks */ +#define FM_CNVRTR0_ADCPOLR 0X1 +#define FM_CNVRTR0_ADCPOLL 0X1 +#define FM_CNVRTR0_AMONOMIX 0X3 +#define FM_CNVRTR0_ADCMU 0X1 +#define FM_CNVRTR0_HPOR 0X1 +#define FM_CNVRTR0_ADCHPDR 0X1 +#define FM_CNVRTR0_ADCHPDL 0X1 + +/* Field Values */ +#define FV_CNVRTR0_ADCPOLR_INVERT 0x1 +#define FV_CNVRTR0_ADCPOLR_NORMAL 0x0 +#define FV_CNVRTR0_ADCPOLL_INVERT 0x1 +#define FV_CNVRTR0_ADCPOLL_NORMAL 0x0 +#define FV_CNVRTR0_ADCMU_ENABLE 0x1 +#define FV_CNVRTR0_ADCMU_DISABLE 0x0 +#define FV_CNVRTR0_ADCHPDR_ENABLE 0x1 +#define FV_CNVRTR0_ADCHPDR_DISABLE 0x0 +#define FV_CNVRTR0_ADCHPDL_ENABLE 0x1 +#define FV_CNVRTR0_ADCHPDL_DISABLE 0x0 + +/* Register Masks */ +#define RM_CNVRTR0_ADCPOLR \ + RM(FM_CNVRTR0_ADCPOLR, FB_CNVRTR0_ADCPOLR) + +#define RM_CNVRTR0_ADCPOLL \ + RM(FM_CNVRTR0_ADCPOLL, FB_CNVRTR0_ADCPOLL) + +#define RM_CNVRTR0_AMONOMIX \ + RM(FM_CNVRTR0_AMONOMIX, FB_CNVRTR0_AMONOMIX) + +#define RM_CNVRTR0_ADCMU \ + RM(FM_CNVRTR0_ADCMU, FB_CNVRTR0_ADCMU) + +#define RM_CNVRTR0_HPOR \ + RM(FM_CNVRTR0_HPOR, FB_CNVRTR0_HPOR) + +#define RM_CNVRTR0_ADCHPDR \ + RM(FM_CNVRTR0_ADCHPDR, FB_CNVRTR0_ADCHPDR) + +#define RM_CNVRTR0_ADCHPDL \ + RM(FM_CNVRTR0_ADCHPDL, FB_CNVRTR0_ADCHPDL) + + +/* Register Values */ +#define RV_CNVRTR0_ADCPOLR_INVERT \ + RV(FV_CNVRTR0_ADCPOLR_INVERT, FB_CNVRTR0_ADCPOLR) + +#define RV_CNVRTR0_ADCPOLR_NORMAL \ + RV(FV_CNVRTR0_ADCPOLR_NORMAL, FB_CNVRTR0_ADCPOLR) + +#define RV_CNVRTR0_ADCPOLL_INVERT \ + RV(FV_CNVRTR0_ADCPOLL_INVERT, FB_CNVRTR0_ADCPOLL) + +#define RV_CNVRTR0_ADCPOLL_NORMAL \ + RV(FV_CNVRTR0_ADCPOLL_NORMAL, FB_CNVRTR0_ADCPOLL) + +#define RV_CNVRTR0_ADCMU_ENABLE \ + RV(FV_CNVRTR0_ADCMU_ENABLE, FB_CNVRTR0_ADCMU) + +#define RV_CNVRTR0_ADCMU_DISABLE \ + RV(FV_CNVRTR0_ADCMU_DISABLE, FB_CNVRTR0_ADCMU) + +#define RV_CNVRTR0_ADCHPDR_ENABLE \ + RV(FV_CNVRTR0_ADCHPDR_ENABLE, FB_CNVRTR0_ADCHPDR) + +#define RV_CNVRTR0_ADCHPDR_DISABLE \ + RV(FV_CNVRTR0_ADCHPDR_DISABLE, FB_CNVRTR0_ADCHPDR) + +#define RV_CNVRTR0_ADCHPDL_ENABLE \ + RV(FV_CNVRTR0_ADCHPDL_ENABLE, FB_CNVRTR0_ADCHPDL) + +#define RV_CNVRTR0_ADCHPDL_DISABLE \ + RV(FV_CNVRTR0_ADCHPDL_DISABLE, FB_CNVRTR0_ADCHPDL) + + +/**************************** + * R_ADCSR (0x17) * + ****************************/ + +/* Field Offsets */ +#define FB_ADCSR_ABCM 6 +#define FB_ADCSR_ABR 3 +#define FB_ADCSR_ABM 0 + +/* Field Masks */ +#define FM_ADCSR_ABCM 0X3 +#define FM_ADCSR_ABR 0X3 +#define FM_ADCSR_ABM 0X7 + +/* Field Values */ +#define FV_ADCSR_ABCM_AUTO 0x0 +#define FV_ADCSR_ABCM_32 0x1 +#define FV_ADCSR_ABCM_40 0x2 +#define FV_ADCSR_ABCM_64 0x3 +#define FV_ADCSR_ABR_32 0x0 +#define FV_ADCSR_ABR_44_1 0x1 +#define FV_ADCSR_ABR_48 0x2 +#define FV_ADCSR_ABM_PT25 0x0 +#define FV_ADCSR_ABM_PT5 0x1 +#define FV_ADCSR_ABM_1 0x2 +#define FV_ADCSR_ABM_2 0x3 + +/* Register Masks */ +#define RM_ADCSR_ABCM RM(FM_ADCSR_ABCM, FB_ADCSR_ABCM) +#define RM_ADCSR_ABR RM(FM_ADCSR_ABR, FB_ADCSR_ABR) +#define RM_ADCSR_ABM RM(FM_ADCSR_ABM, FB_ADCSR_ABM) + +/* Register Values */ +#define RV_ADCSR_ABCM_AUTO \ + RV(FV_ADCSR_ABCM_AUTO, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABCM_32 \ + RV(FV_ADCSR_ABCM_32, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABCM_40 \ + RV(FV_ADCSR_ABCM_40, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABCM_64 \ + RV(FV_ADCSR_ABCM_64, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABR_32 RV(FV_ADCSR_ABR_32, FB_ADCSR_ABR) +#define RV_ADCSR_ABR_44_1 \ + RV(FV_ADCSR_ABR_44_1, FB_ADCSR_ABR) + +#define RV_ADCSR_ABR_48 RV(FV_ADCSR_ABR_48, FB_ADCSR_ABR) +#define RV_ADCSR_ABR_ RV(FV_ADCSR_ABR_, FB_ADCSR_ABR) +#define RV_ADCSR_ABM_PT25 \ + RV(FV_ADCSR_ABM_PT25, FB_ADCSR_ABM) + +#define RV_ADCSR_ABM_PT5 RV(FV_ADCSR_ABM_PT5, FB_ADCSR_ABM) +#define RV_ADCSR_ABM_1 RV(FV_ADCSR_ABM_1, FB_ADCSR_ABM) +#define RV_ADCSR_ABM_2 RV(FV_ADCSR_ABM_2, FB_ADCSR_ABM) + +/****************************** + * R_CNVRTR1 (0x18) * + ******************************/ + +/* Field Offsets */ +#define FB_CNVRTR1_DACPOLR 7 +#define FB_CNVRTR1_DACPOLL 6 +#define FB_CNVRTR1_DMONOMIX 4 +#define FB_CNVRTR1_DACMU 3 +#define FB_CNVRTR1_DEEMPH 2 +#define FB_CNVRTR1_DACDITH 0 + +/* Field Masks */ +#define FM_CNVRTR1_DACPOLR 0X1 +#define FM_CNVRTR1_DACPOLL 0X1 +#define FM_CNVRTR1_DMONOMIX 0X3 +#define FM_CNVRTR1_DACMU 0X1 +#define FM_CNVRTR1_DEEMPH 0X1 +#define FM_CNVRTR1_DACDITH 0X3 + +/* Field Values */ +#define FV_CNVRTR1_DACPOLR_INVERT 0x1 +#define FV_CNVRTR1_DACPOLR_NORMAL 0x0 +#define FV_CNVRTR1_DACPOLL_INVERT 0x1 +#define FV_CNVRTR1_DACPOLL_NORMAL 0x0 +#define FV_CNVRTR1_DMONOMIX_ENABLE 0x1 +#define FV_CNVRTR1_DMONOMIX_DISABLE 0x0 +#define FV_CNVRTR1_DACMU_ENABLE 0x1 +#define FV_CNVRTR1_DACMU_DISABLE 0x0 + +/* Register Masks */ +#define RM_CNVRTR1_DACPOLR \ + RM(FM_CNVRTR1_DACPOLR, FB_CNVRTR1_DACPOLR) + +#define RM_CNVRTR1_DACPOLL \ + RM(FM_CNVRTR1_DACPOLL, FB_CNVRTR1_DACPOLL) + +#define RM_CNVRTR1_DMONOMIX \ + RM(FM_CNVRTR1_DMONOMIX, FB_CNVRTR1_DMONOMIX) + +#define RM_CNVRTR1_DACMU \ + RM(FM_CNVRTR1_DACMU, FB_CNVRTR1_DACMU) + +#define RM_CNVRTR1_DEEMPH \ + RM(FM_CNVRTR1_DEEMPH, FB_CNVRTR1_DEEMPH) + +#define RM_CNVRTR1_DACDITH \ + RM(FM_CNVRTR1_DACDITH, FB_CNVRTR1_DACDITH) + + +/* Register Values */ +#define RV_CNVRTR1_DACPOLR_INVERT \ + RV(FV_CNVRTR1_DACPOLR_INVERT, FB_CNVRTR1_DACPOLR) + +#define RV_CNVRTR1_DACPOLR_NORMAL \ + RV(FV_CNVRTR1_DACPOLR_NORMAL, FB_CNVRTR1_DACPOLR) + +#define RV_CNVRTR1_DACPOLL_INVERT \ + RV(FV_CNVRTR1_DACPOLL_INVERT, FB_CNVRTR1_DACPOLL) + +#define RV_CNVRTR1_DACPOLL_NORMAL \ + RV(FV_CNVRTR1_DACPOLL_NORMAL, FB_CNVRTR1_DACPOLL) + +#define RV_CNVRTR1_DMONOMIX_ENABLE \ + RV(FV_CNVRTR1_DMONOMIX_ENABLE, FB_CNVRTR1_DMONOMIX) + +#define RV_CNVRTR1_DMONOMIX_DISABLE \ + RV(FV_CNVRTR1_DMONOMIX_DISABLE, FB_CNVRTR1_DMONOMIX) + +#define RV_CNVRTR1_DACMU_ENABLE \ + RV(FV_CNVRTR1_DACMU_ENABLE, FB_CNVRTR1_DACMU) + +#define RV_CNVRTR1_DACMU_DISABLE \ + RV(FV_CNVRTR1_DACMU_DISABLE, FB_CNVRTR1_DACMU) + + +/**************************** + * R_DACSR (0x19) * + ****************************/ + +/* Field Offsets */ +#define FB_DACSR_DBCM 6 +#define FB_DACSR_DBR 3 +#define FB_DACSR_DBM 0 + +/* Field Masks */ +#define FM_DACSR_DBCM 0X3 +#define FM_DACSR_DBR 0X3 +#define FM_DACSR_DBM 0X7 + +/* Field Values */ +#define FV_DACSR_DBCM_AUTO 0x0 +#define FV_DACSR_DBCM_32 0x1 +#define FV_DACSR_DBCM_40 0x2 +#define FV_DACSR_DBCM_64 0x3 +#define FV_DACSR_DBR_32 0x0 +#define FV_DACSR_DBR_44_1 0x1 +#define FV_DACSR_DBR_48 0x2 +#define FV_DACSR_DBM_PT25 0x0 +#define FV_DACSR_DBM_PT5 0x1 +#define FV_DACSR_DBM_1 0x2 +#define FV_DACSR_DBM_2 0x3 + +/* Register Masks */ +#define RM_DACSR_DBCM RM(FM_DACSR_DBCM, FB_DACSR_DBCM) +#define RM_DACSR_DBR RM(FM_DACSR_DBR, FB_DACSR_DBR) +#define RM_DACSR_DBM RM(FM_DACSR_DBM, FB_DACSR_DBM) + +/* Register Values */ +#define RV_DACSR_DBCM_AUTO \ + RV(FV_DACSR_DBCM_AUTO, FB_DACSR_DBCM) + +#define RV_DACSR_DBCM_32 \ + RV(FV_DACSR_DBCM_32, FB_DACSR_DBCM) + +#define RV_DACSR_DBCM_40 \ + RV(FV_DACSR_DBCM_40, FB_DACSR_DBCM) + +#define RV_DACSR_DBCM_64 \ + RV(FV_DACSR_DBCM_64, FB_DACSR_DBCM) + +#define RV_DACSR_DBR_32 RV(FV_DACSR_DBR_32, FB_DACSR_DBR) +#define RV_DACSR_DBR_44_1 \ + RV(FV_DACSR_DBR_44_1, FB_DACSR_DBR) + +#define RV_DACSR_DBR_48 RV(FV_DACSR_DBR_48, FB_DACSR_DBR) +#define RV_DACSR_DBM_PT25 \ + RV(FV_DACSR_DBM_PT25, FB_DACSR_DBM) + +#define RV_DACSR_DBM_PT5 RV(FV_DACSR_DBM_PT5, FB_DACSR_DBM) +#define RV_DACSR_DBM_1 RV(FV_DACSR_DBM_1, FB_DACSR_DBM) +#define RV_DACSR_DBM_2 RV(FV_DACSR_DBM_2, FB_DACSR_DBM) + +/**************************** + * R_PWRM1 (0x1A) * + ****************************/ + +/* Field Offsets */ +#define FB_PWRM1_BSTL 7 +#define FB_PWRM1_BSTR 6 +#define FB_PWRM1_PGAL 5 +#define FB_PWRM1_PGAR 4 +#define FB_PWRM1_ADCL 3 +#define FB_PWRM1_ADCR 2 +#define FB_PWRM1_MICB 1 +#define FB_PWRM1_DIGENB 0 + +/* Field Masks */ +#define FM_PWRM1_BSTL 0X1 +#define FM_PWRM1_BSTR 0X1 +#define FM_PWRM1_PGAL 0X1 +#define FM_PWRM1_PGAR 0X1 +#define FM_PWRM1_ADCL 0X1 +#define FM_PWRM1_ADCR 0X1 +#define FM_PWRM1_MICB 0X1 +#define FM_PWRM1_DIGENB 0X1 + +/* Field Values */ +#define FV_PWRM1_BSTL_ENABLE 0x1 +#define FV_PWRM1_BSTL_DISABLE 0x0 +#define FV_PWRM1_BSTR_ENABLE 0x1 +#define FV_PWRM1_BSTR_DISABLE 0x0 +#define FV_PWRM1_PGAL_ENABLE 0x1 +#define FV_PWRM1_PGAL_DISABLE 0x0 +#define FV_PWRM1_PGAR_ENABLE 0x1 +#define FV_PWRM1_PGAR_DISABLE 0x0 +#define FV_PWRM1_ADCL_ENABLE 0x1 +#define FV_PWRM1_ADCL_DISABLE 0x0 +#define FV_PWRM1_ADCR_ENABLE 0x1 +#define FV_PWRM1_ADCR_DISABLE 0x0 +#define FV_PWRM1_MICB_ENABLE 0x1 +#define FV_PWRM1_MICB_DISABLE 0x0 +#define FV_PWRM1_DIGENB_DISABLE 0x1 +#define FV_PWRM1_DIGENB_ENABLE 0x0 + +/* Register Masks */ +#define RM_PWRM1_BSTL RM(FM_PWRM1_BSTL, FB_PWRM1_BSTL) +#define RM_PWRM1_BSTR RM(FM_PWRM1_BSTR, FB_PWRM1_BSTR) +#define RM_PWRM1_PGAL RM(FM_PWRM1_PGAL, FB_PWRM1_PGAL) +#define RM_PWRM1_PGAR RM(FM_PWRM1_PGAR, FB_PWRM1_PGAR) +#define RM_PWRM1_ADCL RM(FM_PWRM1_ADCL, FB_PWRM1_ADCL) +#define RM_PWRM1_ADCR RM(FM_PWRM1_ADCR, FB_PWRM1_ADCR) +#define RM_PWRM1_MICB RM(FM_PWRM1_MICB, FB_PWRM1_MICB) +#define RM_PWRM1_DIGENB \ + RM(FM_PWRM1_DIGENB, FB_PWRM1_DIGENB) + + +/* Register Values */ +#define RV_PWRM1_BSTL_ENABLE \ + RV(FV_PWRM1_BSTL_ENABLE, FB_PWRM1_BSTL) + +#define RV_PWRM1_BSTL_DISABLE \ + RV(FV_PWRM1_BSTL_DISABLE, FB_PWRM1_BSTL) + +#define RV_PWRM1_BSTR_ENABLE \ + RV(FV_PWRM1_BSTR_ENABLE, FB_PWRM1_BSTR) + +#define RV_PWRM1_BSTR_DISABLE \ + RV(FV_PWRM1_BSTR_DISABLE, FB_PWRM1_BSTR) + +#define RV_PWRM1_PGAL_ENABLE \ + RV(FV_PWRM1_PGAL_ENABLE, FB_PWRM1_PGAL) + +#define RV_PWRM1_PGAL_DISABLE \ + RV(FV_PWRM1_PGAL_DISABLE, FB_PWRM1_PGAL) + +#define RV_PWRM1_PGAR_ENABLE \ + RV(FV_PWRM1_PGAR_ENABLE, FB_PWRM1_PGAR) + +#define RV_PWRM1_PGAR_DISABLE \ + RV(FV_PWRM1_PGAR_DISABLE, FB_PWRM1_PGAR) + +#define RV_PWRM1_ADCL_ENABLE \ + RV(FV_PWRM1_ADCL_ENABLE, FB_PWRM1_ADCL) + +#define RV_PWRM1_ADCL_DISABLE \ + RV(FV_PWRM1_ADCL_DISABLE, FB_PWRM1_ADCL) + +#define RV_PWRM1_ADCR_ENABLE \ + RV(FV_PWRM1_ADCR_ENABLE, FB_PWRM1_ADCR) + +#define RV_PWRM1_ADCR_DISABLE \ + RV(FV_PWRM1_ADCR_DISABLE, FB_PWRM1_ADCR) + +#define RV_PWRM1_MICB_ENABLE \ + RV(FV_PWRM1_MICB_ENABLE, FB_PWRM1_MICB) + +#define RV_PWRM1_MICB_DISABLE \ + RV(FV_PWRM1_MICB_DISABLE, FB_PWRM1_MICB) + +#define RV_PWRM1_DIGENB_DISABLE \ + RV(FV_PWRM1_DIGENB_DISABLE, FB_PWRM1_DIGENB) + +#define RV_PWRM1_DIGENB_ENABLE \ + RV(FV_PWRM1_DIGENB_ENABLE, FB_PWRM1_DIGENB) + + +/**************************** + * R_PWRM2 (0x1B) * + ****************************/ + +/* Field Offsets */ +#define FB_PWRM2_D2S 7 +#define FB_PWRM2_HPL 6 +#define FB_PWRM2_HPR 5 +#define FB_PWRM2_SPKL 4 +#define FB_PWRM2_SPKR 3 +#define FB_PWRM2_INSELL 2 +#define FB_PWRM2_INSELR 1 +#define FB_PWRM2_VREF 0 + +/* Field Masks */ +#define FM_PWRM2_D2S 0X1 +#define FM_PWRM2_HPL 0X1 +#define FM_PWRM2_HPR 0X1 +#define FM_PWRM2_SPKL 0X1 +#define FM_PWRM2_SPKR 0X1 +#define FM_PWRM2_INSELL 0X1 +#define FM_PWRM2_INSELR 0X1 +#define FM_PWRM2_VREF 0X1 + +/* Field Values */ +#define FV_PWRM2_D2S_ENABLE 0x1 +#define FV_PWRM2_D2S_DISABLE 0x0 +#define FV_PWRM2_HPL_ENABLE 0x1 +#define FV_PWRM2_HPL_DISABLE 0x0 +#define FV_PWRM2_HPR_ENABLE 0x1 +#define FV_PWRM2_HPR_DISABLE 0x0 +#define FV_PWRM2_SPKL_ENABLE 0x1 +#define FV_PWRM2_SPKL_DISABLE 0x0 +#define FV_PWRM2_SPKR_ENABLE 0x1 +#define FV_PWRM2_SPKR_DISABLE 0x0 +#define FV_PWRM2_INSELL_ENABLE 0x1 +#define FV_PWRM2_INSELL_DISABLE 0x0 +#define FV_PWRM2_INSELR_ENABLE 0x1 +#define FV_PWRM2_INSELR_DISABLE 0x0 +#define FV_PWRM2_VREF_ENABLE 0x1 +#define FV_PWRM2_VREF_DISABLE 0x0 + +/* Register Masks */ +#define RM_PWRM2_D2S RM(FM_PWRM2_D2S, FB_PWRM2_D2S) +#define RM_PWRM2_HPL RM(FM_PWRM2_HPL, FB_PWRM2_HPL) +#define RM_PWRM2_HPR RM(FM_PWRM2_HPR, FB_PWRM2_HPR) +#define RM_PWRM2_SPKL RM(FM_PWRM2_SPKL, FB_PWRM2_SPKL) +#define RM_PWRM2_SPKR RM(FM_PWRM2_SPKR, FB_PWRM2_SPKR) +#define RM_PWRM2_INSELL \ + RM(FM_PWRM2_INSELL, FB_PWRM2_INSELL) + +#define RM_PWRM2_INSELR \ + RM(FM_PWRM2_INSELR, FB_PWRM2_INSELR) + +#define RM_PWRM2_VREF RM(FM_PWRM2_VREF, FB_PWRM2_VREF) + +/* Register Values */ +#define RV_PWRM2_D2S_ENABLE \ + RV(FV_PWRM2_D2S_ENABLE, FB_PWRM2_D2S) + +#define RV_PWRM2_D2S_DISABLE \ + RV(FV_PWRM2_D2S_DISABLE, FB_PWRM2_D2S) + +#define RV_PWRM2_HPL_ENABLE \ + RV(FV_PWRM2_HPL_ENABLE, FB_PWRM2_HPL) + +#define RV_PWRM2_HPL_DISABLE \ + RV(FV_PWRM2_HPL_DISABLE, FB_PWRM2_HPL) + +#define RV_PWRM2_HPR_ENABLE \ + RV(FV_PWRM2_HPR_ENABLE, FB_PWRM2_HPR) + +#define RV_PWRM2_HPR_DISABLE \ + RV(FV_PWRM2_HPR_DISABLE, FB_PWRM2_HPR) + +#define RV_PWRM2_SPKL_ENABLE \ + RV(FV_PWRM2_SPKL_ENABLE, FB_PWRM2_SPKL) + +#define RV_PWRM2_SPKL_DISABLE \ + RV(FV_PWRM2_SPKL_DISABLE, FB_PWRM2_SPKL) + +#define RV_PWRM2_SPKR_ENABLE \ + RV(FV_PWRM2_SPKR_ENABLE, FB_PWRM2_SPKR) + +#define RV_PWRM2_SPKR_DISABLE \ + RV(FV_PWRM2_SPKR_DISABLE, FB_PWRM2_SPKR) + +#define RV_PWRM2_INSELL_ENABLE \ + RV(FV_PWRM2_INSELL_ENABLE, FB_PWRM2_INSELL) + +#define RV_PWRM2_INSELL_DISABLE \ + RV(FV_PWRM2_INSELL_DISABLE, FB_PWRM2_INSELL) + +#define RV_PWRM2_INSELR_ENABLE \ + RV(FV_PWRM2_INSELR_ENABLE, FB_PWRM2_INSELR) + +#define RV_PWRM2_INSELR_DISABLE \ + RV(FV_PWRM2_INSELR_DISABLE, FB_PWRM2_INSELR) + +#define RV_PWRM2_VREF_ENABLE \ + RV(FV_PWRM2_VREF_ENABLE, FB_PWRM2_VREF) + +#define RV_PWRM2_VREF_DISABLE \ + RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF) + + +/****************************** + * R_CONFIG0 (0x1F) * + ******************************/ + +/* Field Offsets */ +#define FB_CONFIG0_ASDM 6 +#define FB_CONFIG0_DSDM 4 +#define FB_CONFIG0_DC_BYPASS 1 +#define FB_CONFIG0_SD_FORCE_ON 0 + +/* Field Masks */ +#define FM_CONFIG0_ASDM 0X3 +#define FM_CONFIG0_DSDM 0X3 +#define FM_CONFIG0_DC_BYPASS 0X1 +#define FM_CONFIG0_SD_FORCE_ON 0X1 + +/* Field Values */ +#define FV_CONFIG0_ASDM_HALF 0x1 +#define FV_CONFIG0_ASDM_FULL 0x2 +#define FV_CONFIG0_ASDM_AUTO 0x3 +#define FV_CONFIG0_DSDM_HALF 0x1 +#define FV_CONFIG0_DSDM_FULL 0x2 +#define FV_CONFIG0_DSDM_AUTO 0x3 +#define FV_CONFIG0_DC_BYPASS_ENABLE 0x1 +#define FV_CONFIG0_DC_BYPASS_DISABLE 0x0 +#define FV_CONFIG0_SD_FORCE_ON_ENABLE 0x1 +#define FV_CONFIG0_SD_FORCE_ON_DISABLE 0x0 + +/* Register Masks */ +#define RM_CONFIG0_ASDM \ + RM(FM_CONFIG0_ASDM, FB_CONFIG0_ASDM) + +#define RM_CONFIG0_DSDM \ + RM(FM_CONFIG0_DSDM, FB_CONFIG0_DSDM) + +#define RM_CONFIG0_DC_BYPASS \ + RM(FM_CONFIG0_DC_BYPASS, FB_CONFIG0_DC_BYPASS) + +#define RM_CONFIG0_SD_FORCE_ON \ + RM(FM_CONFIG0_SD_FORCE_ON, FB_CONFIG0_SD_FORCE_ON) + + +/* Register Values */ +#define RV_CONFIG0_ASDM_HALF \ + RV(FV_CONFIG0_ASDM_HALF, FB_CONFIG0_ASDM) + +#define RV_CONFIG0_ASDM_FULL \ + RV(FV_CONFIG0_ASDM_FULL, FB_CONFIG0_ASDM) + +#define RV_CONFIG0_ASDM_AUTO \ + RV(FV_CONFIG0_ASDM_AUTO, FB_CONFIG0_ASDM) + +#define RV_CONFIG0_DSDM_HALF \ + RV(FV_CONFIG0_DSDM_HALF, FB_CONFIG0_DSDM) + +#define RV_CONFIG0_DSDM_FULL \ + RV(FV_CONFIG0_DSDM_FULL, FB_CONFIG0_DSDM) + +#define RV_CONFIG0_DSDM_AUTO \ + RV(FV_CONFIG0_DSDM_AUTO, FB_CONFIG0_DSDM) + +#define RV_CONFIG0_DC_BYPASS_ENABLE \ + RV(FV_CONFIG0_DC_BYPASS_ENABLE, FB_CONFIG0_DC_BYPASS) + +#define RV_CONFIG0_DC_BYPASS_DISABLE \ + RV(FV_CONFIG0_DC_BYPASS_DISABLE, FB_CONFIG0_DC_BYPASS) + +#define RV_CONFIG0_SD_FORCE_ON_ENABLE \ + RV(FV_CONFIG0_SD_FORCE_ON_ENABLE, FB_CONFIG0_SD_FORCE_ON) + +#define RV_CONFIG0_SD_FORCE_ON_DISABLE \ + RV(FV_CONFIG0_SD_FORCE_ON_DISABLE, FB_CONFIG0_SD_FORCE_ON) + + +/****************************** + * R_CONFIG1 (0x20) * + ******************************/ + +/* Field Offsets */ +#define FB_CONFIG1_EQ2_EN 7 +#define FB_CONFIG1_EQ2_BE 4 +#define FB_CONFIG1_EQ1_EN 3 +#define FB_CONFIG1_EQ1_BE 0 + +/* Field Masks */ +#define FM_CONFIG1_EQ2_EN 0X1 +#define FM_CONFIG1_EQ2_BE 0X7 +#define FM_CONFIG1_EQ1_EN 0X1 +#define FM_CONFIG1_EQ1_BE 0X7 + +/* Field Values */ +#define FV_CONFIG1_EQ2_EN_ENABLE 0x1 +#define FV_CONFIG1_EQ2_EN_DISABLE 0x0 +#define FV_CONFIG1_EQ2_BE_PRE 0x0 +#define FV_CONFIG1_EQ2_BE_PRE_EQ_0 0x1 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_1 0x2 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_2 0x3 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_3 0x4 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_4 0x5 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_5 0x6 +#define FV_CONFIG1_EQ1_EN_ENABLE 0x1 +#define FV_CONFIG1_EQ1_EN_DISABLE 0x0 +#define FV_CONFIG1_EQ1_BE_PRE 0x0 +#define FV_CONFIG1_EQ1_BE_PRE_EQ_0 0x1 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_1 0x2 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_2 0x3 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_3 0x4 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_4 0x5 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_5 0x6 + +/* Register Masks */ +#define RM_CONFIG1_EQ2_EN \ + RM(FM_CONFIG1_EQ2_EN, FB_CONFIG1_EQ2_EN) + +#define RM_CONFIG1_EQ2_BE \ + RM(FM_CONFIG1_EQ2_BE, FB_CONFIG1_EQ2_BE) + +#define RM_CONFIG1_EQ1_EN \ + RM(FM_CONFIG1_EQ1_EN, FB_CONFIG1_EQ1_EN) + +#define RM_CONFIG1_EQ1_BE \ + RM(FM_CONFIG1_EQ1_BE, FB_CONFIG1_EQ1_BE) + + +/* Register Values */ +#define RV_CONFIG1_EQ2_EN_ENABLE \ + RV(FV_CONFIG1_EQ2_EN_ENABLE, FB_CONFIG1_EQ2_EN) + +#define RV_CONFIG1_EQ2_EN_DISABLE \ + RV(FV_CONFIG1_EQ2_EN_DISABLE, FB_CONFIG1_EQ2_EN) + +#define RV_CONFIG1_EQ2_BE_PRE \ + RV(FV_CONFIG1_EQ2_BE_PRE, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ_0 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ_0, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_1 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_1, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_2 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_2, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_3 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_3, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_4 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_4, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_5 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_5, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ1_EN_ENABLE \ + RV(FV_CONFIG1_EQ1_EN_ENABLE, FB_CONFIG1_EQ1_EN) + +#define RV_CONFIG1_EQ1_EN_DISABLE \ + RV(FV_CONFIG1_EQ1_EN_DISABLE, FB_CONFIG1_EQ1_EN) + +#define RV_CONFIG1_EQ1_BE_PRE \ + RV(FV_CONFIG1_EQ1_BE_PRE, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ_0 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ_0, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_1 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_1, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_2 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_2, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_3 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_3, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_4 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_4, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_5 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_5, FB_CONFIG1_EQ1_BE) + + +/****************************** + * R_DMICCTL (0x24) * + ******************************/ + +/* Field Offsets */ +#define FB_DMICCTL_DMICEN 7 +#define FB_DMICCTL_DMONO 4 +#define FB_DMICCTL_DMPHADJ 2 +#define FB_DMICCTL_DMRATE 0 + +/* Field Masks */ +#define FM_DMICCTL_DMICEN 0X1 +#define FM_DMICCTL_DMONO 0X1 +#define FM_DMICCTL_DMPHADJ 0X3 +#define FM_DMICCTL_DMRATE 0X3 + +/* Field Values */ +#define FV_DMICCTL_DMICEN_ENABLE 0x1 +#define FV_DMICCTL_DMICEN_DISABLE 0x0 +#define FV_DMICCTL_DMONO_STEREO 0x0 +#define FV_DMICCTL_DMONO_MONO 0x1 + +/* Register Masks */ +#define RM_DMICCTL_DMICEN \ + RM(FM_DMICCTL_DMICEN, FB_DMICCTL_DMICEN) + +#define RM_DMICCTL_DMONO \ + RM(FM_DMICCTL_DMONO, FB_DMICCTL_DMONO) + +#define RM_DMICCTL_DMPHADJ \ + RM(FM_DMICCTL_DMPHADJ, FB_DMICCTL_DMPHADJ) + +#define RM_DMICCTL_DMRATE \ + RM(FM_DMICCTL_DMRATE, FB_DMICCTL_DMRATE) + + +/* Register Values */ +#define RV_DMICCTL_DMICEN_ENABLE \ + RV(FV_DMICCTL_DMICEN_ENABLE, FB_DMICCTL_DMICEN) + +#define RV_DMICCTL_DMICEN_DISABLE \ + RV(FV_DMICCTL_DMICEN_DISABLE, FB_DMICCTL_DMICEN) + +#define RV_DMICCTL_DMONO_STEREO \ + RV(FV_DMICCTL_DMONO_STEREO, FB_DMICCTL_DMONO) + +#define RV_DMICCTL_DMONO_MONO \ + RV(FV_DMICCTL_DMONO_MONO, FB_DMICCTL_DMONO) + + +/***************************** + * R_CLECTL (0x25) * + *****************************/ + +/* Field Offsets */ +#define FB_CLECTL_LVL_MODE 4 +#define FB_CLECTL_WINDOWSEL 3 +#define FB_CLECTL_EXP_EN 2 +#define FB_CLECTL_LIMIT_EN 1 +#define FB_CLECTL_COMP_EN 0 + +/* Field Masks */ +#define FM_CLECTL_LVL_MODE 0X1 +#define FM_CLECTL_WINDOWSEL 0X1 +#define FM_CLECTL_EXP_EN 0X1 +#define FM_CLECTL_LIMIT_EN 0X1 +#define FM_CLECTL_COMP_EN 0X1 + +/* Field Values */ +#define FV_CLECTL_LVL_MODE_AVG 0x0 +#define FV_CLECTL_LVL_MODE_PEAK 0x1 +#define FV_CLECTL_WINDOWSEL_512 0x0 +#define FV_CLECTL_WINDOWSEL_64 0x1 +#define FV_CLECTL_EXP_EN_ENABLE 0x1 +#define FV_CLECTL_EXP_EN_DISABLE 0x0 +#define FV_CLECTL_LIMIT_EN_ENABLE 0x1 +#define FV_CLECTL_LIMIT_EN_DISABLE 0x0 +#define FV_CLECTL_COMP_EN_ENABLE 0x1 +#define FV_CLECTL_COMP_EN_DISABLE 0x0 + +/* Register Masks */ +#define RM_CLECTL_LVL_MODE \ + RM(FM_CLECTL_LVL_MODE, FB_CLECTL_LVL_MODE) + +#define RM_CLECTL_WINDOWSEL \ + RM(FM_CLECTL_WINDOWSEL, FB_CLECTL_WINDOWSEL) + +#define RM_CLECTL_EXP_EN \ + RM(FM_CLECTL_EXP_EN, FB_CLECTL_EXP_EN) + +#define RM_CLECTL_LIMIT_EN \ + RM(FM_CLECTL_LIMIT_EN, FB_CLECTL_LIMIT_EN) + +#define RM_CLECTL_COMP_EN \ + RM(FM_CLECTL_COMP_EN, FB_CLECTL_COMP_EN) + + +/* Register Values */ +#define RV_CLECTL_LVL_MODE_AVG \ + RV(FV_CLECTL_LVL_MODE_AVG, FB_CLECTL_LVL_MODE) + +#define RV_CLECTL_LVL_MODE_PEAK \ + RV(FV_CLECTL_LVL_MODE_PEAK, FB_CLECTL_LVL_MODE) + +#define RV_CLECTL_WINDOWSEL_512 \ + RV(FV_CLECTL_WINDOWSEL_512, FB_CLECTL_WINDOWSEL) + +#define RV_CLECTL_WINDOWSEL_64 \ + RV(FV_CLECTL_WINDOWSEL_64, FB_CLECTL_WINDOWSEL) + +#define RV_CLECTL_EXP_EN_ENABLE \ + RV(FV_CLECTL_EXP_EN_ENABLE, FB_CLECTL_EXP_EN) + +#define RV_CLECTL_EXP_EN_DISABLE \ + RV(FV_CLECTL_EXP_EN_DISABLE, FB_CLECTL_EXP_EN) + +#define RV_CLECTL_LIMIT_EN_ENABLE \ + RV(FV_CLECTL_LIMIT_EN_ENABLE, FB_CLECTL_LIMIT_EN) + +#define RV_CLECTL_LIMIT_EN_DISABLE \ + RV(FV_CLECTL_LIMIT_EN_DISABLE, FB_CLECTL_LIMIT_EN) + +#define RV_CLECTL_COMP_EN_ENABLE \ + RV(FV_CLECTL_COMP_EN_ENABLE, FB_CLECTL_COMP_EN) + +#define RV_CLECTL_COMP_EN_DISABLE \ + RV(FV_CLECTL_COMP_EN_DISABLE, FB_CLECTL_COMP_EN) + + +/***************************** + * R_MUGAIN (0x26) * + *****************************/ + +/* Field Offsets */ +#define FB_MUGAIN_CLEMUG 0 + +/* Field Masks */ +#define FM_MUGAIN_CLEMUG 0X1F + +/* Field Values */ +#define FV_MUGAIN_CLEMUG_46PT5DB 0x1F +#define FV_MUGAIN_CLEMUG_0DB 0x0 + +/* Register Masks */ +#define RM_MUGAIN_CLEMUG \ + RM(FM_MUGAIN_CLEMUG, FB_MUGAIN_CLEMUG) + + +/* Register Values */ +#define RV_MUGAIN_CLEMUG_46PT5DB \ + RV(FV_MUGAIN_CLEMUG_46PT5DB, FB_MUGAIN_CLEMUG) + +#define RV_MUGAIN_CLEMUG_0DB \ + RV(FV_MUGAIN_CLEMUG_0DB, FB_MUGAIN_CLEMUG) + + +/***************************** + * R_COMPTH (0x27) * + *****************************/ + +/* Field Offsets */ +#define FB_COMPTH 0 + +/* Field Masks */ +#define FM_COMPTH 0XFF + +/* Field Values */ +#define FV_COMPTH_0DB 0xFF +#define FV_COMPTH_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_COMPTH RM(FM_COMPTH, FB_COMPTH) + +/* Register Values */ +#define RV_COMPTH_0DB RV(FV_COMPTH_0DB, FB_COMPTH) +#define RV_COMPTH_N95PT625DB \ + RV(FV_COMPTH_N95PT625DB, FB_COMPTH) + + +/***************************** + * R_CMPRAT (0x28) * + *****************************/ + +/* Field Offsets */ +#define FB_CMPRAT 0 + +/* Field Masks */ +#define FM_CMPRAT 0X1F + +/* Register Masks */ +#define RM_CMPRAT RM(FM_CMPRAT, FB_CMPRAT) + +/****************************** + * R_CATKTCL (0x29) * + ******************************/ + +/* Field Offsets */ +#define FB_CATKTCL 0 + +/* Field Masks */ +#define FM_CATKTCL 0XFF + +/* Register Masks */ +#define RM_CATKTCL RM(FM_CATKTCL, FB_CATKTCL) + +/****************************** + * R_CATKTCH (0x2A) * + ******************************/ + +/* Field Offsets */ +#define FB_CATKTCH 0 + +/* Field Masks */ +#define FM_CATKTCH 0XFF + +/* Register Masks */ +#define RM_CATKTCH RM(FM_CATKTCH, FB_CATKTCH) + +/****************************** + * R_CRELTCL (0x2B) * + ******************************/ + +/* Field Offsets */ +#define FB_CRELTCL 0 + +/* Field Masks */ +#define FM_CRELTCL 0XFF + +/* Register Masks */ +#define RM_CRELTCL RM(FM_CRELTCL, FB_CRELTCL) + +/****************************** + * R_CRELTCH (0x2C) * + ******************************/ + +/* Field Offsets */ +#define FB_CRELTCH 0 + +/* Field Masks */ +#define FM_CRELTCH 0XFF + +/* Register Masks */ +#define RM_CRELTCH RM(FM_CRELTCH, FB_CRELTCH) + +/**************************** + * R_LIMTH (0x2D) * + ****************************/ + +/* Field Offsets */ +#define FB_LIMTH 0 + +/* Field Masks */ +#define FM_LIMTH 0XFF + +/* Field Values */ +#define FV_LIMTH_0DB 0xFF +#define FV_LIMTH_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_LIMTH RM(FM_LIMTH, FB_LIMTH) + +/* Register Values */ +#define RV_LIMTH_0DB RV(FV_LIMTH_0DB, FB_LIMTH) +#define RV_LIMTH_N95PT625DB RV(FV_LIMTH_N95PT625DB, FB_LIMTH) + +/***************************** + * R_LIMTGT (0x2E) * + *****************************/ + +/* Field Offsets */ +#define FB_LIMTGT 0 + +/* Field Masks */ +#define FM_LIMTGT 0XFF + +/* Field Values */ +#define FV_LIMTGT_0DB 0xFF +#define FV_LIMTGT_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_LIMTGT RM(FM_LIMTGT, FB_LIMTGT) + +/* Register Values */ +#define RV_LIMTGT_0DB RV(FV_LIMTGT_0DB, FB_LIMTGT) +#define RV_LIMTGT_N95PT625DB \ + RV(FV_LIMTGT_N95PT625DB, FB_LIMTGT) + + +/****************************** + * R_LATKTCL (0x2F) * + ******************************/ + +/* Field Offsets */ +#define FB_LATKTCL 0 + +/* Field Masks */ +#define FM_LATKTCL 0XFF + +/* Register Masks */ +#define RM_LATKTCL RM(FM_LATKTCL, FB_LATKTCL) + +/****************************** + * R_LATKTCH (0x30) * + ******************************/ + +/* Field Offsets */ +#define FB_LATKTCH 0 + +/* Field Masks */ +#define FM_LATKTCH 0XFF + +/* Register Masks */ +#define RM_LATKTCH RM(FM_LATKTCH, FB_LATKTCH) + +/****************************** + * R_LRELTCL (0x31) * + ******************************/ + +/* Field Offsets */ +#define FB_LRELTCL 0 + +/* Field Masks */ +#define FM_LRELTCL 0XFF + +/* Register Masks */ +#define RM_LRELTCL RM(FM_LRELTCL, FB_LRELTCL) + +/****************************** + * R_LRELTCH (0x32) * + ******************************/ + +/* Field Offsets */ +#define FB_LRELTCH 0 + +/* Field Masks */ +#define FM_LRELTCH 0XFF + +/* Register Masks */ +#define RM_LRELTCH RM(FM_LRELTCH, FB_LRELTCH) + +/**************************** + * R_EXPTH (0x33) * + ****************************/ + +/* Field Offsets */ +#define FB_EXPTH 0 + +/* Field Masks */ +#define FM_EXPTH 0XFF + +/* Field Values */ +#define FV_EXPTH_0DB 0xFF +#define FV_EXPTH_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_EXPTH RM(FM_EXPTH, FB_EXPTH) + +/* Register Values */ +#define RV_EXPTH_0DB RV(FV_EXPTH_0DB, FB_EXPTH) +#define RV_EXPTH_N95PT625DB RV(FV_EXPTH_N95PT625DB, FB_EXPTH) + +/***************************** + * R_EXPRAT (0x34) * + *****************************/ + +/* Field Offsets */ +#define FB_EXPRAT 0 + +/* Field Masks */ +#define FM_EXPRAT 0X7 + +/* Register Masks */ +#define RM_EXPRAT RM(FM_EXPRAT, FB_EXPRAT) + +/****************************** + * R_XATKTCL (0x35) * + ******************************/ + +/* Field Offsets */ +#define FB_XATKTCL 0 + +/* Field Masks */ +#define FM_XATKTCL 0XFF + +/* Register Masks */ +#define RM_XATKTCL RM(FM_XATKTCL, FB_XATKTCL) + +/****************************** + * R_XATKTCH (0x36) * + ******************************/ + +/* Field Offsets */ +#define FB_XATKTCH 0 + +/* Field Masks */ +#define FM_XATKTCH 0XFF + +/* Register Masks */ +#define RM_XATKTCH RM(FM_XATKTCH, FB_XATKTCH) + +/****************************** + * R_XRELTCL (0x37) * + ******************************/ + +/* Field Offsets */ +#define FB_XRELTCL 0 + +/* Field Masks */ +#define FM_XRELTCL 0XFF + +/* Register Masks */ +#define RM_XRELTCL RM(FM_XRELTCL, FB_XRELTCL) + +/****************************** + * R_XRELTCH (0x38) * + ******************************/ + +/* Field Offsets */ +#define FB_XRELTCH 0 + +/* Field Masks */ +#define FM_XRELTCH 0XFF + +/* Register Masks */ +#define RM_XRELTCH RM(FM_XRELTCH, FB_XRELTCH) + +/**************************** + * R_FXCTL (0x39) * + ****************************/ + +/* Field Offsets */ +#define FB_FXCTL_3DEN 4 +#define FB_FXCTL_TEEN 3 +#define FB_FXCTL_TNLFBYPASS 2 +#define FB_FXCTL_BEEN 1 +#define FB_FXCTL_BNLFBYPASS 0 + +/* Field Masks */ +#define FM_FXCTL_3DEN 0X1 +#define FM_FXCTL_TEEN 0X1 +#define FM_FXCTL_TNLFBYPASS 0X1 +#define FM_FXCTL_BEEN 0X1 +#define FM_FXCTL_BNLFBYPASS 0X1 + +/* Field Values */ +#define FV_FXCTL_3DEN_ENABLE 0x1 +#define FV_FXCTL_3DEN_DISABLE 0x0 +#define FV_FXCTL_TEEN_ENABLE 0x1 +#define FV_FXCTL_TEEN_DISABLE 0x0 +#define FV_FXCTL_TNLFBYPASS_ENABLE 0x1 +#define FV_FXCTL_TNLFBYPASS_DISABLE 0x0 +#define FV_FXCTL_BEEN_ENABLE 0x1 +#define FV_FXCTL_BEEN_DISABLE 0x0 +#define FV_FXCTL_BNLFBYPASS_ENABLE 0x1 +#define FV_FXCTL_BNLFBYPASS_DISABLE 0x0 + +/* Register Masks */ +#define RM_FXCTL_3DEN RM(FM_FXCTL_3DEN, FB_FXCTL_3DEN) +#define RM_FXCTL_TEEN RM(FM_FXCTL_TEEN, FB_FXCTL_TEEN) +#define RM_FXCTL_TNLFBYPASS \ + RM(FM_FXCTL_TNLFBYPASS, FB_FXCTL_TNLFBYPASS) + +#define RM_FXCTL_BEEN RM(FM_FXCTL_BEEN, FB_FXCTL_BEEN) +#define RM_FXCTL_BNLFBYPASS \ + RM(FM_FXCTL_BNLFBYPASS, FB_FXCTL_BNLFBYPASS) + + +/* Register Values */ +#define RV_FXCTL_3DEN_ENABLE \ + RV(FV_FXCTL_3DEN_ENABLE, FB_FXCTL_3DEN) + +#define RV_FXCTL_3DEN_DISABLE \ + RV(FV_FXCTL_3DEN_DISABLE, FB_FXCTL_3DEN) + +#define RV_FXCTL_TEEN_ENABLE \ + RV(FV_FXCTL_TEEN_ENABLE, FB_FXCTL_TEEN) + +#define RV_FXCTL_TEEN_DISABLE \ + RV(FV_FXCTL_TEEN_DISABLE, FB_FXCTL_TEEN) + +#define RV_FXCTL_TNLFBYPASS_ENABLE \ + RV(FV_FXCTL_TNLFBYPASS_ENABLE, FB_FXCTL_TNLFBYPASS) + +#define RV_FXCTL_TNLFBYPASS_DISABLE \ + RV(FV_FXCTL_TNLFBYPASS_DISABLE, FB_FXCTL_TNLFBYPASS) + +#define RV_FXCTL_BEEN_ENABLE \ + RV(FV_FXCTL_BEEN_ENABLE, FB_FXCTL_BEEN) + +#define RV_FXCTL_BEEN_DISABLE \ + RV(FV_FXCTL_BEEN_DISABLE, FB_FXCTL_BEEN) + +#define RV_FXCTL_BNLFBYPASS_ENABLE \ + RV(FV_FXCTL_BNLFBYPASS_ENABLE, FB_FXCTL_BNLFBYPASS) + +#define RV_FXCTL_BNLFBYPASS_DISABLE \ + RV(FV_FXCTL_BNLFBYPASS_DISABLE, FB_FXCTL_BNLFBYPASS) + + +/******************************* + * R_DACCRWRL (0x3A) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRWRL_DACCRWDL 0 + +/* Field Masks */ +#define FM_DACCRWRL_DACCRWDL 0XFF + +/* Register Masks */ +#define RM_DACCRWRL_DACCRWDL \ + RM(FM_DACCRWRL_DACCRWDL, FB_DACCRWRL_DACCRWDL) + + +/******************************* + * R_DACCRWRM (0x3B) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRWRM_DACCRWDM 0 + +/* Field Masks */ +#define FM_DACCRWRM_DACCRWDM 0XFF + +/* Register Masks */ +#define RM_DACCRWRM_DACCRWDM \ + RM(FM_DACCRWRM_DACCRWDM, FB_DACCRWRM_DACCRWDM) + + +/******************************* + * R_DACCRWRH (0x3C) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRWRH_DACCRWDH 0 + +/* Field Masks */ +#define FM_DACCRWRH_DACCRWDH 0XFF + +/* Register Masks */ +#define RM_DACCRWRH_DACCRWDH \ + RM(FM_DACCRWRH_DACCRWDH, FB_DACCRWRH_DACCRWDH) + + +/******************************* + * R_DACCRRDL (0x3D) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRRDL 0 + +/* Field Masks */ +#define FM_DACCRRDL 0XFF + +/* Register Masks */ +#define RM_DACCRRDL RM(FM_DACCRRDL, FB_DACCRRDL) + +/******************************* + * R_DACCRRDM (0x3E) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRRDM 0 + +/* Field Masks */ +#define FM_DACCRRDM 0XFF + +/* Register Masks */ +#define RM_DACCRRDM RM(FM_DACCRRDM, FB_DACCRRDM) + +/******************************* + * R_DACCRRDH (0x3F) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRRDH 0 + +/* Field Masks */ +#define FM_DACCRRDH 0XFF + +/* Register Masks */ +#define RM_DACCRRDH RM(FM_DACCRRDH, FB_DACCRRDH) + +/******************************** + * R_DACCRADDR (0x40) * + ********************************/ + +/* Field Offsets */ +#define FB_DACCRADDR_DACCRADD 0 + +/* Field Masks */ +#define FM_DACCRADDR_DACCRADD 0XFF + +/* Register Masks */ +#define RM_DACCRADDR_DACCRADD \ + RM(FM_DACCRADDR_DACCRADD, FB_DACCRADDR_DACCRADD) + + +/****************************** + * R_DCOFSEL (0x41) * + ******************************/ + +/* Field Offsets */ +#define FB_DCOFSEL_DC_COEF_SEL 0 + +/* Field Masks */ +#define FM_DCOFSEL_DC_COEF_SEL 0X7 + +/* Field Values */ +#define FV_DCOFSEL_DC_COEF_SEL_2_N8 0x0 +#define FV_DCOFSEL_DC_COEF_SEL_2_N9 0x1 +#define FV_DCOFSEL_DC_COEF_SEL_2_N10 0x2 +#define FV_DCOFSEL_DC_COEF_SEL_2_N11 0x3 +#define FV_DCOFSEL_DC_COEF_SEL_2_N12 0x4 +#define FV_DCOFSEL_DC_COEF_SEL_2_N13 0x5 +#define FV_DCOFSEL_DC_COEF_SEL_2_N14 0x6 +#define FV_DCOFSEL_DC_COEF_SEL_2_N15 0x7 + +/* Register Masks */ +#define RM_DCOFSEL_DC_COEF_SEL \ + RM(FM_DCOFSEL_DC_COEF_SEL, FB_DCOFSEL_DC_COEF_SEL) + + +/* Register Values */ +#define RV_DCOFSEL_DC_COEF_SEL_2_N8 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N8, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N9 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N9, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N10 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N10, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N11 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N11, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N12 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N12, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N13 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N13, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N14 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N14, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N15 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N15, FB_DCOFSEL_DC_COEF_SEL) + + +/****************************** + * R_PLLCTL9 (0x4E) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTL9_REFDIV_PLL1 0 + +/* Field Masks */ +#define FM_PLLCTL9_REFDIV_PLL1 0XFF + +/* Register Masks */ +#define RM_PLLCTL9_REFDIV_PLL1 \ + RM(FM_PLLCTL9_REFDIV_PLL1, FB_PLLCTL9_REFDIV_PLL1) + + +/****************************** + * R_PLLCTLA (0x4F) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLA_OUTDIV_PLL1 0 + +/* Field Masks */ +#define FM_PLLCTLA_OUTDIV_PLL1 0XFF + +/* Register Masks */ +#define RM_PLLCTLA_OUTDIV_PLL1 \ + RM(FM_PLLCTLA_OUTDIV_PLL1, FB_PLLCTLA_OUTDIV_PLL1) + + +/****************************** + * R_PLLCTLB (0x50) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLB_FBDIV_PLL1L 0 + +/* Field Masks */ +#define FM_PLLCTLB_FBDIV_PLL1L 0XFF + +/* Register Masks */ +#define RM_PLLCTLB_FBDIV_PLL1L \ + RM(FM_PLLCTLB_FBDIV_PLL1L, FB_PLLCTLB_FBDIV_PLL1L) + + +/****************************** + * R_PLLCTLC (0x51) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLC_FBDIV_PLL1H 0 + +/* Field Masks */ +#define FM_PLLCTLC_FBDIV_PLL1H 0X7 + +/* Register Masks */ +#define RM_PLLCTLC_FBDIV_PLL1H \ + RM(FM_PLLCTLC_FBDIV_PLL1H, FB_PLLCTLC_FBDIV_PLL1H) + + +/****************************** + * R_PLLCTLD (0x52) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLD_RZ_PLL1 3 +#define FB_PLLCTLD_CP_PLL1 0 + +/* Field Masks */ +#define FM_PLLCTLD_RZ_PLL1 0X7 +#define FM_PLLCTLD_CP_PLL1 0X7 + +/* Register Masks */ +#define RM_PLLCTLD_RZ_PLL1 \ + RM(FM_PLLCTLD_RZ_PLL1, FB_PLLCTLD_RZ_PLL1) + +#define RM_PLLCTLD_CP_PLL1 \ + RM(FM_PLLCTLD_CP_PLL1, FB_PLLCTLD_CP_PLL1) + + +/****************************** + * R_PLLCTLE (0x53) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLE_REFDIV_PLL2 0 + +/* Field Masks */ +#define FM_PLLCTLE_REFDIV_PLL2 0XFF + +/* Register Masks */ +#define RM_PLLCTLE_REFDIV_PLL2 \ + RM(FM_PLLCTLE_REFDIV_PLL2, FB_PLLCTLE_REFDIV_PLL2) + + +/****************************** + * R_PLLCTLF (0x54) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLF_OUTDIV_PLL2 0 + +/* Field Masks */ +#define FM_PLLCTLF_OUTDIV_PLL2 0XFF + +/* Register Masks */ +#define RM_PLLCTLF_OUTDIV_PLL2 \ + RM(FM_PLLCTLF_OUTDIV_PLL2, FB_PLLCTLF_OUTDIV_PLL2) + + +/******************************* + * R_PLLCTL10 (0x55) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL10_FBDIV_PLL2L 0 + +/* Field Masks */ +#define FM_PLLCTL10_FBDIV_PLL2L 0XFF + +/* Register Masks */ +#define RM_PLLCTL10_FBDIV_PLL2L \ + RM(FM_PLLCTL10_FBDIV_PLL2L, FB_PLLCTL10_FBDIV_PLL2L) + + +/******************************* + * R_PLLCTL11 (0x56) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL11_FBDIV_PLL2H 0 + +/* Field Masks */ +#define FM_PLLCTL11_FBDIV_PLL2H 0X7 + +/* Register Masks */ +#define RM_PLLCTL11_FBDIV_PLL2H \ + RM(FM_PLLCTL11_FBDIV_PLL2H, FB_PLLCTL11_FBDIV_PLL2H) + + +/******************************* + * R_PLLCTL12 (0x57) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL12_RZ_PLL2 3 +#define FB_PLLCTL12_CP_PLL2 0 + +/* Field Masks */ +#define FM_PLLCTL12_RZ_PLL2 0X7 +#define FM_PLLCTL12_CP_PLL2 0X7 + +/* Register Masks */ +#define RM_PLLCTL12_RZ_PLL2 \ + RM(FM_PLLCTL12_RZ_PLL2, FB_PLLCTL12_RZ_PLL2) + +#define RM_PLLCTL12_CP_PLL2 \ + RM(FM_PLLCTL12_CP_PLL2, FB_PLLCTL12_CP_PLL2) + + +/******************************* + * R_PLLCTL1B (0x60) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL1B_VCOI_PLL2 4 +#define FB_PLLCTL1B_VCOI_PLL1 2 + +/* Field Masks */ +#define FM_PLLCTL1B_VCOI_PLL2 0X3 +#define FM_PLLCTL1B_VCOI_PLL1 0X3 + +/* Register Masks */ +#define RM_PLLCTL1B_VCOI_PLL2 \ + RM(FM_PLLCTL1B_VCOI_PLL2, FB_PLLCTL1B_VCOI_PLL2) + +#define RM_PLLCTL1B_VCOI_PLL1 \ + RM(FM_PLLCTL1B_VCOI_PLL1, FB_PLLCTL1B_VCOI_PLL1) + + +/******************************* + * R_PLLCTL1C (0x61) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL1C_PDB_PLL2 2 +#define FB_PLLCTL1C_PDB_PLL1 1 + +/* Field Masks */ +#define FM_PLLCTL1C_PDB_PLL2 0X1 +#define FM_PLLCTL1C_PDB_PLL1 0X1 + +/* Field Values */ +#define FV_PLLCTL1C_PDB_PLL2_ENABLE 0x1 +#define FV_PLLCTL1C_PDB_PLL2_DISABLE 0x0 +#define FV_PLLCTL1C_PDB_PLL1_ENABLE 0x1 +#define FV_PLLCTL1C_PDB_PLL1_DISABLE 0x0 + +/* Register Masks */ +#define RM_PLLCTL1C_PDB_PLL2 \ + RM(FM_PLLCTL1C_PDB_PLL2, FB_PLLCTL1C_PDB_PLL2) + +#define RM_PLLCTL1C_PDB_PLL1 \ + RM(FM_PLLCTL1C_PDB_PLL1, FB_PLLCTL1C_PDB_PLL1) + + +/* Register Values */ +#define RV_PLLCTL1C_PDB_PLL2_ENABLE \ + RV(FV_PLLCTL1C_PDB_PLL2_ENABLE, FB_PLLCTL1C_PDB_PLL2) + +#define RV_PLLCTL1C_PDB_PLL2_DISABLE \ + RV(FV_PLLCTL1C_PDB_PLL2_DISABLE, FB_PLLCTL1C_PDB_PLL2) + +#define RV_PLLCTL1C_PDB_PLL1_ENABLE \ + RV(FV_PLLCTL1C_PDB_PLL1_ENABLE, FB_PLLCTL1C_PDB_PLL1) + +#define RV_PLLCTL1C_PDB_PLL1_DISABLE \ + RV(FV_PLLCTL1C_PDB_PLL1_DISABLE, FB_PLLCTL1C_PDB_PLL1) + + +/******************************* + * R_TIMEBASE (0x77) * + *******************************/ + +/* Field Offsets */ +#define FB_TIMEBASE_DIVIDER 0 + +/* Field Masks */ +#define FM_TIMEBASE_DIVIDER 0XFF + +/* Register Masks */ +#define RM_TIMEBASE_DIVIDER \ + RM(FM_TIMEBASE_DIVIDER, FB_TIMEBASE_DIVIDER) + + +/***************************** + * R_DEVIDL (0x7D) * + *****************************/ + +/* Field Offsets */ +#define FB_DEVIDL_DIDL 0 + +/* Field Masks */ +#define FM_DEVIDL_DIDL 0XFF + +/* Register Masks */ +#define RM_DEVIDL_DIDL RM(FM_DEVIDL_DIDL, FB_DEVIDL_DIDL) + +/***************************** + * R_DEVIDH (0x7E) * + *****************************/ + +/* Field Offsets */ +#define FB_DEVIDH_DIDH 0 + +/* Field Masks */ +#define FM_DEVIDH_DIDH 0XFF + +/* Register Masks */ +#define RM_DEVIDH_DIDH RM(FM_DEVIDH_DIDH, FB_DEVIDH_DIDH) + +/**************************** + * R_RESET (0x80) * + ****************************/ + +/* Field Offsets */ +#define FB_RESET 0 + +/* Field Masks */ +#define FM_RESET 0XFF + +/* Field Values */ +#define FV_RESET_ENABLE 0x85 + +/* Register Masks */ +#define RM_RESET RM(FM_RESET, FB_RESET) + +/* Register Values */ +#define RV_RESET_ENABLE RV(FV_RESET_ENABLE, FB_RESET) + +/******************************** + * R_DACCRSTAT (0x8A) * + ********************************/ + +/* Field Offsets */ +#define FB_DACCRSTAT_DACCR_BUSY 7 + +/* Field Masks */ +#define FM_DACCRSTAT_DACCR_BUSY 0X1 + +/* Register Masks */ +#define RM_DACCRSTAT_DACCR_BUSY \ + RM(FM_DACCRSTAT_DACCR_BUSY, FB_DACCRSTAT_DACCR_BUSY) + + +/****************************** + * R_PLLCTL0 (0x8E) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTL0_PLL2_LOCK 1 +#define FB_PLLCTL0_PLL1_LOCK 0 + +/* Field Masks */ +#define FM_PLLCTL0_PLL2_LOCK 0X1 +#define FM_PLLCTL0_PLL1_LOCK 0X1 + +/* Register Masks */ +#define RM_PLLCTL0_PLL2_LOCK \ + RM(FM_PLLCTL0_PLL2_LOCK, FB_PLLCTL0_PLL2_LOCK) + +#define RM_PLLCTL0_PLL1_LOCK \ + RM(FM_PLLCTL0_PLL1_LOCK, FB_PLLCTL0_PLL1_LOCK) + + +/******************************** + * R_PLLREFSEL (0x8F) * + ********************************/ + +/* Field Offsets */ +#define FB_PLLREFSEL_PLL2_REF_SEL 4 +#define FB_PLLREFSEL_PLL1_REF_SEL 0 + +/* Field Masks */ +#define FM_PLLREFSEL_PLL2_REF_SEL 0X7 +#define FM_PLLREFSEL_PLL1_REF_SEL 0X7 + +/* Field Values */ +#define FV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1 0x0 +#define FV_PLLREFSEL_PLL2_REF_SEL_MCLK2 0x1 +#define FV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1 0x0 +#define FV_PLLREFSEL_PLL1_REF_SEL_MCLK2 0x1 + +/* Register Masks */ +#define RM_PLLREFSEL_PLL2_REF_SEL \ + RM(FM_PLLREFSEL_PLL2_REF_SEL, FB_PLLREFSEL_PLL2_REF_SEL) + +#define RM_PLLREFSEL_PLL1_REF_SEL \ + RM(FM_PLLREFSEL_PLL1_REF_SEL, FB_PLLREFSEL_PLL1_REF_SEL) + + +/* Register Values */ +#define RV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1 \ + RV(FV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1, FB_PLLREFSEL_PLL2_REF_SEL) + +#define RV_PLLREFSEL_PLL2_REF_SEL_MCLK2 \ + RV(FV_PLLREFSEL_PLL2_REF_SEL_MCLK2, FB_PLLREFSEL_PLL2_REF_SEL) + +#define RV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1 \ + RV(FV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1, FB_PLLREFSEL_PLL1_REF_SEL) + +#define RV_PLLREFSEL_PLL1_REF_SEL_MCLK2 \ + RV(FV_PLLREFSEL_PLL1_REF_SEL_MCLK2, FB_PLLREFSEL_PLL1_REF_SEL) + + +/******************************* + * R_DACMBCEN (0xC7) * + *******************************/ + +/* Field Offsets */ +#define FB_DACMBCEN_MBCEN3 2 +#define FB_DACMBCEN_MBCEN2 1 +#define FB_DACMBCEN_MBCEN1 0 + +/* Field Masks */ +#define FM_DACMBCEN_MBCEN3 0X1 +#define FM_DACMBCEN_MBCEN2 0X1 +#define FM_DACMBCEN_MBCEN1 0X1 + +/* Register Masks */ +#define RM_DACMBCEN_MBCEN3 \ + RM(FM_DACMBCEN_MBCEN3, FB_DACMBCEN_MBCEN3) + +#define RM_DACMBCEN_MBCEN2 \ + RM(FM_DACMBCEN_MBCEN2, FB_DACMBCEN_MBCEN2) + +#define RM_DACMBCEN_MBCEN1 \ + RM(FM_DACMBCEN_MBCEN1, FB_DACMBCEN_MBCEN1) + + +/******************************** + * R_DACMBCCTL (0xC8) * + ********************************/ + +/* Field Offsets */ +#define FB_DACMBCCTL_LVLMODE3 5 +#define FB_DACMBCCTL_WINSEL3 4 +#define FB_DACMBCCTL_LVLMODE2 3 +#define FB_DACMBCCTL_WINSEL2 2 +#define FB_DACMBCCTL_LVLMODE1 1 +#define FB_DACMBCCTL_WINSEL1 0 + +/* Field Masks */ +#define FM_DACMBCCTL_LVLMODE3 0X1 +#define FM_DACMBCCTL_WINSEL3 0X1 +#define FM_DACMBCCTL_LVLMODE2 0X1 +#define FM_DACMBCCTL_WINSEL2 0X1 +#define FM_DACMBCCTL_LVLMODE1 0X1 +#define FM_DACMBCCTL_WINSEL1 0X1 + +/* Register Masks */ +#define RM_DACMBCCTL_LVLMODE3 \ + RM(FM_DACMBCCTL_LVLMODE3, FB_DACMBCCTL_LVLMODE3) + +#define RM_DACMBCCTL_WINSEL3 \ + RM(FM_DACMBCCTL_WINSEL3, FB_DACMBCCTL_WINSEL3) + +#define RM_DACMBCCTL_LVLMODE2 \ + RM(FM_DACMBCCTL_LVLMODE2, FB_DACMBCCTL_LVLMODE2) + +#define RM_DACMBCCTL_WINSEL2 \ + RM(FM_DACMBCCTL_WINSEL2, FB_DACMBCCTL_WINSEL2) + +#define RM_DACMBCCTL_LVLMODE1 \ + RM(FM_DACMBCCTL_LVLMODE1, FB_DACMBCCTL_LVLMODE1) + +#define RM_DACMBCCTL_WINSEL1 \ + RM(FM_DACMBCCTL_WINSEL1, FB_DACMBCCTL_WINSEL1) + + +/********************************* + * R_DACMBCMUG1 (0xC9) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCMUG1_PHASE 5 +#define FB_DACMBCMUG1_MUGAIN 0 + +/* Field Masks */ +#define FM_DACMBCMUG1_PHASE 0X1 +#define FM_DACMBCMUG1_MUGAIN 0X1F + +/* Register Masks */ +#define RM_DACMBCMUG1_PHASE \ + RM(FM_DACMBCMUG1_PHASE, FB_DACMBCMUG1_PHASE) + +#define RM_DACMBCMUG1_MUGAIN \ + RM(FM_DACMBCMUG1_MUGAIN, FB_DACMBCMUG1_MUGAIN) + + +/********************************* + * R_DACMBCTHR1 (0xCA) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCTHR1_THRESH 0 + +/* Field Masks */ +#define FM_DACMBCTHR1_THRESH 0XFF + +/* Register Masks */ +#define RM_DACMBCTHR1_THRESH \ + RM(FM_DACMBCTHR1_THRESH, FB_DACMBCTHR1_THRESH) + + +/********************************* + * R_DACMBCRAT1 (0xCB) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCRAT1_RATIO 0 + +/* Field Masks */ +#define FM_DACMBCRAT1_RATIO 0X1F + +/* Register Masks */ +#define RM_DACMBCRAT1_RATIO \ + RM(FM_DACMBCRAT1_RATIO, FB_DACMBCRAT1_RATIO) + + +/********************************** + * R_DACMBCATK1L (0xCC) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK1L_TCATKL 0 + +/* Field Masks */ +#define FM_DACMBCATK1L_TCATKL 0XFF + +/* Register Masks */ +#define RM_DACMBCATK1L_TCATKL \ + RM(FM_DACMBCATK1L_TCATKL, FB_DACMBCATK1L_TCATKL) + + +/********************************** + * R_DACMBCATK1H (0xCD) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK1H_TCATKH 0 + +/* Field Masks */ +#define FM_DACMBCATK1H_TCATKH 0XFF + +/* Register Masks */ +#define RM_DACMBCATK1H_TCATKH \ + RM(FM_DACMBCATK1H_TCATKH, FB_DACMBCATK1H_TCATKH) + + +/********************************** + * R_DACMBCREL1L (0xCE) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL1L_TCRELL 0 + +/* Field Masks */ +#define FM_DACMBCREL1L_TCRELL 0XFF + +/* Register Masks */ +#define RM_DACMBCREL1L_TCRELL \ + RM(FM_DACMBCREL1L_TCRELL, FB_DACMBCREL1L_TCRELL) + + +/********************************** + * R_DACMBCREL1H (0xCF) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL1H_TCRELH 0 + +/* Field Masks */ +#define FM_DACMBCREL1H_TCRELH 0XFF + +/* Register Masks */ +#define RM_DACMBCREL1H_TCRELH \ + RM(FM_DACMBCREL1H_TCRELH, FB_DACMBCREL1H_TCRELH) + + +/********************************* + * R_DACMBCMUG2 (0xD0) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCMUG2_PHASE 5 +#define FB_DACMBCMUG2_MUGAIN 0 + +/* Field Masks */ +#define FM_DACMBCMUG2_PHASE 0X1 +#define FM_DACMBCMUG2_MUGAIN 0X1F + +/* Register Masks */ +#define RM_DACMBCMUG2_PHASE \ + RM(FM_DACMBCMUG2_PHASE, FB_DACMBCMUG2_PHASE) + +#define RM_DACMBCMUG2_MUGAIN \ + RM(FM_DACMBCMUG2_MUGAIN, FB_DACMBCMUG2_MUGAIN) + + +/********************************* + * R_DACMBCTHR2 (0xD1) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCTHR2_THRESH 0 + +/* Field Masks */ +#define FM_DACMBCTHR2_THRESH 0XFF + +/* Register Masks */ +#define RM_DACMBCTHR2_THRESH \ + RM(FM_DACMBCTHR2_THRESH, FB_DACMBCTHR2_THRESH) + + +/********************************* + * R_DACMBCRAT2 (0xD2) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCRAT2_RATIO 0 + +/* Field Masks */ +#define FM_DACMBCRAT2_RATIO 0X1F + +/* Register Masks */ +#define RM_DACMBCRAT2_RATIO \ + RM(FM_DACMBCRAT2_RATIO, FB_DACMBCRAT2_RATIO) + + +/********************************** + * R_DACMBCATK2L (0xD3) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK2L_TCATKL 0 + +/* Field Masks */ +#define FM_DACMBCATK2L_TCATKL 0XFF + +/* Register Masks */ +#define RM_DACMBCATK2L_TCATKL \ + RM(FM_DACMBCATK2L_TCATKL, FB_DACMBCATK2L_TCATKL) + + +/********************************** + * R_DACMBCATK2H (0xD4) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK2H_TCATKH 0 + +/* Field Masks */ +#define FM_DACMBCATK2H_TCATKH 0XFF + +/* Register Masks */ +#define RM_DACMBCATK2H_TCATKH \ + RM(FM_DACMBCATK2H_TCATKH, FB_DACMBCATK2H_TCATKH) + + +/********************************** + * R_DACMBCREL2L (0xD5) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL2L_TCRELL 0 + +/* Field Masks */ +#define FM_DACMBCREL2L_TCRELL 0XFF + +/* Register Masks */ +#define RM_DACMBCREL2L_TCRELL \ + RM(FM_DACMBCREL2L_TCRELL, FB_DACMBCREL2L_TCRELL) + + +/********************************** + * R_DACMBCREL2H (0xD6) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL2H_TCRELH 0 + +/* Field Masks */ +#define FM_DACMBCREL2H_TCRELH 0XFF + +/* Register Masks */ +#define RM_DACMBCREL2H_TCRELH \ + RM(FM_DACMBCREL2H_TCRELH, FB_DACMBCREL2H_TCRELH) + + +/********************************* + * R_DACMBCMUG3 (0xD7) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCMUG3_PHASE 5 +#define FB_DACMBCMUG3_MUGAIN 0 + +/* Field Masks */ +#define FM_DACMBCMUG3_PHASE 0X1 +#define FM_DACMBCMUG3_MUGAIN 0X1F + +/* Register Masks */ +#define RM_DACMBCMUG3_PHASE \ + RM(FM_DACMBCMUG3_PHASE, FB_DACMBCMUG3_PHASE) + +#define RM_DACMBCMUG3_MUGAIN \ + RM(FM_DACMBCMUG3_MUGAIN, FB_DACMBCMUG3_MUGAIN) + + +/********************************* + * R_DACMBCTHR3 (0xD8) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCTHR3_THRESH 0 + +/* Field Masks */ +#define FM_DACMBCTHR3_THRESH 0XFF + +/* Register Masks */ +#define RM_DACMBCTHR3_THRESH \ + RM(FM_DACMBCTHR3_THRESH, FB_DACMBCTHR3_THRESH) + + +/********************************* + * R_DACMBCRAT3 (0xD9) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCRAT3_RATIO 0 + +/* Field Masks */ +#define FM_DACMBCRAT3_RATIO 0X1F + +/* Register Masks */ +#define RM_DACMBCRAT3_RATIO \ + RM(FM_DACMBCRAT3_RATIO, FB_DACMBCRAT3_RATIO) + + +/********************************** + * R_DACMBCATK3L (0xDA) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK3L_TCATKL 0 + +/* Field Masks */ +#define FM_DACMBCATK3L_TCATKL 0XFF + +/* Register Masks */ +#define RM_DACMBCATK3L_TCATKL \ + RM(FM_DACMBCATK3L_TCATKL, FB_DACMBCATK3L_TCATKL) + + +/********************************** + * R_DACMBCATK3H (0xDB) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK3H_TCATKH 0 + +/* Field Masks */ +#define FM_DACMBCATK3H_TCATKH 0XFF + +/* Register Masks */ +#define RM_DACMBCATK3H_TCATKH \ + RM(FM_DACMBCATK3H_TCATKH, FB_DACMBCATK3H_TCATKH) + + +/********************************** + * R_DACMBCREL3L (0xDC) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL3L_TCRELL 0 + +/* Field Masks */ +#define FM_DACMBCREL3L_TCRELL 0XFF + +/* Register Masks */ +#define RM_DACMBCREL3L_TCRELL \ + RM(FM_DACMBCREL3L_TCRELL, FB_DACMBCREL3L_TCRELL) + + +/********************************** + * R_DACMBCREL3H (0xDD) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL3H_TCRELH 0 + +/* Field Masks */ +#define FM_DACMBCREL3H_TCRELH 0XFF + +/* Register Masks */ +#define RM_DACMBCREL3H_TCRELH \ + RM(FM_DACMBCREL3H_TCRELH, FB_DACMBCREL3H_TCRELH) + + +#endif /* __WOOKIE_H__ */ -- 2.11.0

2 1

[alsa-devel] [PATCH 0/3] ALSA: es1968: Fine-tuning for seven function implementations
by SF Markus Elfring 15 Nov '17

15 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 15 Nov 2017 10:23:45 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 16 function calls together with a variable assignment Use common error handling code in two functions Use common error handling code in snd_es1968_capture_open() sound/pci/es1968.c | 163 ++++++++++++++++++++++++++++++----------------------- 1 file changed, 92 insertions(+), 71 deletions(-) -- 2.15.0

1 3

[alsa-devel] [PATCH] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 15 Nov '17

15 Nov '17

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 250 +++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..29066f0b744f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -262,4 +262,17 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + depends on SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a5c5bc5732a2..915d2ab9cdd6 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -18,6 +18,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..db271ecd3370 --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,250 @@ +/* + * cnl_rt274.c - ASOC Machine driver for CNL + * + * Copyright (C) 2016-17 Intel Corp + * Author: Guneshwor Singh <guneshwor.o.singh(a)intel.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; 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. + * + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 19200000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_jack cnl_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_codec_set_jack(codec, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = "cnl-audio", + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int snd_cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver snd_cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cnl_rt274_probe, +}; + +module_platform_driver(snd_cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.15.0

2 1

[alsa-devel] [GIT PULL] ASoC updates for v4.15
by Mark Brown 15 Nov '17

15 Nov '17

The following changes since commit 39dae59d66acd86d1de24294bd2f343fd5e7a625: Linux 4.14-rc8 (2017-11-05 13:05:14 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.15 for you to fetch changes up to df6a3e245541ac61cc99f2887437e0a43dd08f2e: Merge remote-tracking branches 'asoc/topic/tfa9879', 'asoc/topic/ts3a277e', 'asoc/topic/wm8741', 'asoc/topic/wm97xx' and 'asoc/topic/zte' into asoc-next (2017-11-10 21:31:33 +0000) ---------------------------------------------------------------- ASoC: Updates for v4.15 The biggest thing this release has been the conversion of the AC98 bus to the driver model, that's been a long time coming so thanks to Robert Jarzmik for his dedication there. Due to there being some AC97 MFD there's a few fairly large changes in input and the MFD layer, mainly to the wm97xx driver. There's also some drivers/drm changes to support the new AMD Stoney platform, these are shared with the DRM subsystem and should be being merged via both. Within the subsystem the overwhelming bulk of the changes is in the Intel drivers which continue to need lots of cleanups and fixes, this release they've also gained support for their open source firmware. There's also some large changs in the core as Morimoto-san continues to mirror operations into the component level in preparation for conversion of drivers to that. - The AC97 bus has finally caught up with the driver model thanks to some dedicated and persistent work from Robert Jarzmik. - Continued work from Morimoto-san on moving us towards being able to use components for everything. - Lots of cleanups for the Intel platform code, including support for their open source audio firmware. - Support for scaling MCLK with sample rate in simple-card. - Support for AMD Stoney platform. ---------------------------------------------------------------- Akshu Agrawal (4): drm/amdgpu Moving amdgpu asic types to a separate file ASoC: AMD: Add machine driver for cz rt5650 ASoC: amd: Make the driver name consistent across files ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2 Andrea Bondavalli (1): ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+ Andrew F. Davis (1): ASoC: cs42l56: Fix reset GPIO name in example DT binding Arnd Bergmann (8): ASoC: dapm: add initialization for w_param_text pointer ASoC: dapm: mark 'snd_soc_dapm_free_kcontrol' as static ASoC: Intel: fix Kconfig dependencies ASoC: intel: include linux/module.h as needed ASoC: Intel: improve DMADEVICES dependency ASoC: Intel: improve SND_SOC_INTEL_MACH dependencies ASoC: rt5514: mark PM functions as __maybe_unused ASoC: rt5514: work around link error Arvind Yadav (3): ASoC: davinci-mcasp: Handle return value of devm_kasprintf ASoC: omap-hdmi-audio: Handle return value of devm_kasprintf ASoC: fsl-asoc-card: Handle return value of devm_kasprintf Axel Lin (1): ASoC: max98925: Return proper error if revision mismatch Bard Liao (4): ASoC: rt5670: add set_bclk_ratio in dai ops ASoC: rt5659: move set_sysclk to codec level ASoC: rt5659: move set_pll to codec level ASoC: rt5645: remove unexisting route on new rt5645 Bhumika Goyal (2): ASoC: Intel: atom: make sst_platform_compr_ops const ASoC: kirkwood: make kirkwood_soc_platform const Carlo Caione (8): SoC: intel: byt: Introduce new custom IN2 map ASoC: rt5651: Convert rt5651 micbias1 to a supply widget ASoC: rt5651: Enable jack detection on JD* pins ASoC: rt5651: Enable jack detection on JD* pins ASoC: rt5651: Rework quirk logic ASoC: rt5651: Enable JD1_1 quirk for KIANO laptopt ASoC: intel: byt: Add headset jack ASoC: intel: byt: Enable IN2 map quirk for a KIANO laptop Charles Keepax (9): ASoC: arizona: Add new common Arizona init function ASoC: arizona: Add handling for audio related device tree entries ASoC: arizona: Add audio device tree bindings mfd: arizona: Remove audio related device tree code mfd: arizona: Remove audio bindings from MFD binding document ASoC: arizona: Add support for setting the output volume limits ASoC: arizona: Add device tree binding doc for volume limits ASoC: wm8998: Correct handling of input muxes ASoC: wm8741: Use snd_soc_update_bits rather than hard coding Chintan Patel (2): ASoC: Intel: Skylake: Fix jack name format substitution ASoC: Intel: Skylake: Decrease loglevel for topology loading Christophe Jaillet (2): ASoC: samsung: i2s: Fix error handling path in i2s_set_sysclk() ASoC: davinci-mcasp: Fix an error handling path in 'davinci_mcasp_probe()' Christos Gkekas (1): ASoC: cygnus: Remove unnecessary active_slots check Colin Ian King (7): ASoC: fsl_spdif: make const arrays rate static ASoC: zte: spdif: remove duplicate initialization of dma_data ASoC: qcom: remove duplicate initializations of dma_ch and v ASoC: hdmi-codec: use sizeof_field rather than declaring hcp ASoC: rl6231: make arrays div and pd static const, reduces object code size ASoC: Intel: Skylake: fix swapped order of function arguments dir and pin_index ASoC: Intel: sst: remove redundant variable dma_dev_name Corentin LABBE (1): ASoC: Intel: Atom: Remove unneeded linux/miscdevice.h include Damien Riegel (2): ASoC: codecs: msm8916-wcd-analog: use btn0 released detection ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup Douglas Anderson (1): ASoC: rockchip: Allocate enough memory so we don't overflow routes Ed Blake (12): ASoC: img-i2s-out: Rename suspend / resume funcs ASoC: img-parallel-out: Add pm_runtime_get/put to set_fmt callback ASoC: img-spdif-out: Rename suspend / resume funcs ASoC: img-parallel-out: Check pm_runtime_get_sync return code ASoC: img-i2s-in: Add suspend / resume handling ASoC: img-i2s-in: Add runtime PM ASoC: img-i2s-out: Add suspend / resume handling ASoC: img-i2s-out: Add control of sys clock to runtime PM ASoC: img-spdif-in: Add suspend / resume handling ASoC: img-spdif-in: Add runtime PM ASoC: img-spdif-out: Add suspend / resume handling ASoC: img-spdif-out: Add control of sys clock to runtime PM Fabio Estevam (1): ASoC: tfa9879: Add device tree bindings Fang, Yang A (1): ASoC: ts3a227e: add acpi table Geert Uytterhoeven (1): ASoC: fsi: Use of_device_get_match_data() helper Guenter Roeck (1): ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds Guneshwor Singh (5): ASoC: Intel: Skylake: Add flag to check to register FE dais from topology ASoC: Intel: Skylake: Add dai load ops for dais from topology ASoC: Intel: Skylake: Fix missing sentinel in sst_acpi_mach ASoC: Intel: Skylake: Fix updown mixer module format ASoC: Intel: Skylake: Add channel map in updown mixer module IPC Gustavo A. R. Silva (7): ASoC: msm8916-wcd-analog: mark expected switch fall-through ASoC: tlv320aic23: mark expected switch fall-through ASoC: tlv320dac31xx: mark expected switch fall-through ASoC: tpa6130a2: mark expected switch fall-through ASoC: wm8753: mark expected switch fall-throughs ASoC: wm8993: mark expected switch fall-throughs ASoC: wm8994: mark expected switch fall-throughs Harsha Priya N (3): ASoC: Intel: Kbl: Add Playback DAI for fixup ASoC: Intel: Enable tdm slots for max98927 ASoC: Intel: Fix setting of SSP parameters in Kabylake machine driver Jaechul Lee (3): ASoC: samsung: i2s: Use specific name for i2s dais ASoC: samsung: Use 'samsung-i2s' cpu_dai for dai_links ASoC: samsung: Fix invalid argument when devm_gpiod_get is called Jaikrishna Nemallapudi (1): ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments Jean Delvare (1): ASoC: rt5645: Make a few struct const Jean-François Têtu (2): ASoC: msm8916-wcd-digital: fix RX2 MIX1 and RX3 MIX1 ASoC: codecs: msm8916-wcd-analog: fix micbias level Jeffy Chen (5): ASoC: rt5514: Add devicetree binding support for rt5514-spi ASoC: rockchip: Add dapm route for DMic ASoC: rockchip: Add dapm route for HDMI ASoC: rockchip: Init dapm routes dynamically ASoC: rockchip: Use bus_type to distinguish rt5514 dsp from rt5514 codec John Keeping (1): ASoC: rockchip: i2s: fix unbalanced clk_disable Kuninori Morimoto (40): ASoC: rsnd: fix ADG flags ASoC: add missing snd_soc_component_set_jack ASoC: rsnd: add rsnd_dma_alloc() ASoC: soc-core: remove rtd NULL check on soc_free_pcm_runtime() ASoC: remove unneeded dai->driver check ASoC: remove unneeded dai->driver->ops check ASoC: add null_snd_soc_ops and reduce NULL ops check ASoC: add Component level pcm_new/pcm_free v2 ASoC: add Component level set_bias_level ASoC: rcar: skip disabled-SSI nodes ASoC: rsnd: add generic rsnd_flags_xxx() macro ASoC: rsnd: use generic rsnd_flags_xxx() macro on ADG ASoC: rsnd: DVC kctrl sets once ASoC: rsnd: CTU kctrl sets once ASoC: rsnd: makes volume ramp rate list generic ASoC: rsnd: add MIX Volume Ramp support ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() ASoC: soc-core: remove unnecessary message from snd_soc_register_component() ASoC: rsnd: add rsnd_kctrl_xxx() macro ASoC: rsnd: more clear ADG clock debug info ASoC: audio-graph-scu-card: add missing Capture routing on Example ASoC: audio-graph-scu-card: remove unnecessary route patch from Example 1 ASoC: snd_soc_component_driver has snd_pcm_ops ASoC: snd_soc_component_driver has snd_compr_ops ASoC: snd_soc_component_driver has pmdown_time ASoC: snd_soc_component_driver has endianness ASoC: snd_soc_component_driver has non_legacy_dai_naming ASoC: rsnd: don't use io->mod[] directly ASoC: rsnd: tidyup rsnd_mod_next() for loop method ASoC: rsnd: NULL check is not needed for clk_unprepare() ASoC: rsnd: use snd_pcm_running() in rsnd_io_is_working() ASoC: rsnd: Don't check SSISR::DIRQ when Capture ASoC: rsnd: rsnd_ssi_run_mods() needs to care ssi_parent_mod ASoC: rsnd: remove NULL check from rsnd_mod_name()/rsnd_mod_id() ASoC: rsnd: return -EIO if rsnd_dmaen_request_channel() failed ASoC: add snd_soc_component_read32 ASoC: add snd_soc_component_xxx_bias_level() ASoC: add snd_soc_component_cache_sync() ASoC: add snd_soc_dapm_kcontrol_component() Li Xu (1): ASoC: cs43130: Add break keyword to switch case Lori Hikichi (3): ASoC: cygnus: Add EXPORT_SYMBOL for helper function ASoC: cygnus: Remove set_fmt from SPDIF dai ops ASoC: cygnus: Remove support for 8 bit audio and for mono Marco Franchi (1): ASoC: sgtl5000: Remove leading zero from '@0a' notation Marek Szyprowski (1): ASoC: samsung: i2s: disable secondary DAI until it gets fixed Mark Brown (38): Merge tag 'v4.14-rc1' into asoc-rockchip ASoC: wm9712: Add missing brace Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'fix/pcm' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'topic/helpers' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'linus-4.14-rc4-acp-prereq' of git://people.freedesktop.org/~agd5f/linux into asoc-amd Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/msm8916' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/rockchip' into asoc-linus Merge remote-tracking branches 'asoc/fix/adau17x1', 'asoc/fix/davinci' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5514', 'asoc/fix/rt5616', 'asoc/fix/rt5659', 'asoc/fix/rt5663', 'asoc/fix/samsung' and 'asoc/fix/stm32' into asoc-linus Merge remote-tracking branches 'asoc/fix/topology' and 'asoc/fix/wm8998' into asoc-linus Revert "ASoC: rt5651: Enable jack detection on JD* pins" ASoC: pcm512x: Scrub my work address from the driver Merge tag 'asoc-fix-v4.14-rc6' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/sunxi' into asoc-linus Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branch 'asoc/topic/qcom' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ac97-mfd', 'asoc/topic/amd' and 'asoc/topic/arizona-mfd' into asoc-next Merge remote-tracking branches 'asoc/topic/bcm2835', 'asoc/topic/cs42l56' and 'asoc/topic/cs43130' into asoc-next Merge remote-tracking branches 'asoc/topic/cygnus', 'asoc/topic/da7213', 'asoc/topic/davinci' and 'asoc/topic/doc' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/fallthrough', 'asoc/topic/fsi', 'asoc/topic/fsl' and 'asoc/topic/graph' into asoc-next Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/img' and 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branches 'asoc/topic/max98925', 'asoc/topic/max98927', 'asoc/topic/msm8916' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/rl6231' and 'asoc/topic/rt5514' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5651', 'asoc/topic/rt5659', 'asoc/topic/rt5663' and 'asoc/topic/rt5670' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/stm32', 'asoc/topic/sunxi' and 'asoc/topic/tas571x' into asoc-next Merge remote-tracking branches 'asoc/topic/tfa9879', 'asoc/topic/ts3a277e', 'asoc/topic/wm8741', 'asoc/topic/wm97xx' and 'asoc/topic/zte' into asoc-next Matthias Reichl (4): ASoC: bcm2835: Add support for TDM modes ASoC: bcm2835: Support additional samplerates up to 384kHz ASoC: bcm2835: Enforce full symmetry ASoC: bcm2835: Support left/right justified and DSP modes Maxime Ripard (4): ASoC: sun8i-codec: Invert Master / Slave condition ASoC: sun8i-codec: Fix left and right channels inversion ASoC: sun8i-codec: Set the BCLK divider ASoC: sun8i-codec: Add a comment on the LRCK inversion Naveen M (1): ASoC: Intel: Headset button support in kabylake machine driver Nicolas Dechesne (1): ASoC: codecs: msm8916-wcd-analog: fix module autoload Nicolin Chen (2): ASoC: fsl_ssi: Caculate bit clock rate using slot number and width ASoC: fsl-asoc-card: Don't error out if ENOTSUPP Nik Nyby (1): ASoC: dwc: fix typos in Kconfig Oder Chiou (3): ASoC: rt5663: Add the function of impedance sensing ASoC: rt5514-spi: Let the buf_size to align with period_bytes ASoC: rt5663: Delay and retry reading rt5663 ID register Olivier Moysan (9): ASoC: stm32: sai: fix stop management in isr ASoC: stm32: sai: Fix DMA burst size ASoC: stm32: sai: Fix get reset controller ASoC: stm32: Add synchronization to SAI bindings ASoC: stm32: sai: Move static settings to DAI init ASoC: stm32: sai: Remove spurious IRQs on stop ASoC: stm32: sai: Add synchronization support ASoC: add mclk-fs to audio graph card binding ASoC: add mclk-fs support to audio graph card Pankaj Bharadiya (3): ASoC: Intel: Skylake: Fix potential NULL pointer dereference ASoC: Intel: Skylake: Fix uuid_module memory leak in failure case ASoC: Intel: Skylake: Check for NHLT ACPI header signature Pierre-Louis Bossart (31): ASoC: Intel: boards: use devm_clk_get() unconditionally ASoC: Intel: bytcr-rt5651: fix capture routes ASoC: Intel: bytcr_rt5640: simplify MCLK quirk tests ASoC: max98090: reduce verbosity on PLL unlock ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code ASoC: Intel: cht_bsw_max98090: add support for Baytrail ASoC: Intel: atom: use cht_bsw_max98090 for Baytrail Chromebooks ASoC: Intel: cht_bsw_max98090: add gpio-based jack detection ASoC: rt5670: refactor DMI quirks and fix Dell Venue settings ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name ASoC: Intel: cht_bsw_max98090: remove useless code, align with ChromeOS driver ASoC: Add helper to find codec_dai from dai_name ASoC: Intel: bytcr_rt5651: add MCLK, quirks and cleanups ASoC: Intel: bytcr_rt5640: cosmetic fixes ASoC: Intel: cht_bsw_rt5645: cosmetic fixes ASoC: Intel: bytcht_da7213: cosmetic fixes ASoC: Intel: bytcht_es8316: remove useless code ASoC: Intel: boards: use helper to get codec_dai ASoC: Intel: boards: fix off-by-one dailink id ASoC: Intel: boards: remove hard-coded compressed dailinks ASoC: move ACPI common code out of Intel/sst tree ASoC: Intel: common: use c99 syntax for ACPI/machine tables ASoC: ACPI: add new fields for SOF support ASoC: Intel: move all ACPI match tables to common module ASoC: Intel: add SOF firmare/topology file information ASoC: Intel: move machine drivers to dedicated KConfig ASoC: Intel: reorder boards Kconfig by chronological order ASoC: Intel: clarify Kconfig dependencies ASoC: Intel: Add depends on X86 ASoC: da7213: add support for DSP modes Randy Dunlap (1): ASoC: fix build warning in soc-core.c Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: wm97xx: fix compilation corner case Ryan Lee (2): ASoC: max98927: Added support for DSP_A and DSP_B format ASoC: max98927: Added max98927_dai_tdm_slot function Sergej Sawazki (1): ASoC: wm8741: Fix setting BCLK and LRCLK polarity Sriram Periyasamy (2): ASoC: Intel: Skylake: Optimize UUID handling to fill pin info ASoC: Intel: Skylake: Add dynamic module id support Stuart Henderson (1): ASoC: dapm: Make snd_soc_dapm_add/del_routes use runtime mutex subclass Subhransu S. Prusty (3): ASoC: hdac_hdmi: Fix static checker warning for sprintf usage ASoC: hdac_hdmi: Fix possible NULL pointer dereference ASoC: hdac_hdmi: Fix possible memory leak on parse and map nid failure Takashi Iwai (4): ASoC: tas571x: Kill BUG_ON() usage ASoC: davinci: Kill BUG_ON() usage ASoC: intel: Kill BUG_ON() usage ASoC: pcm: Sync delayed work before releasing resources Thierry Escande (1): ASoC: Intel: cht_bsw_max98090_ti: Fix jack initialization Vijendar Mukunda (7): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: Added asic_type as ACP DMA driver platform data ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney ASoC: amd: Report accurate hw_ptr during dma ASoC: amd: Modified DMA transfer Mechanism for Playback Vinod Koul (1): ASoC: Intel: kbl: fix jack name anish kumar (3): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params jiada wang (1): ASoC: soc-pcm: check symmetry after hw_params oder_chiou(a)realtek.com (4): ASoC: rt5514: The ACPI also should use the function rt5514_parse_dp() ASoC: rt5514: Voice wakeup support. ASoC: rt5514-spi: check irq status to schedule data copy in resume function ASoC: rt5663: Check the JD status in the button pushing olivier moysan (3): ASoC: stm32: sai: fix warning in stm32_sai_set_config() ASoC: stm32: spdifrx: fix 16 bits capture ASoC: stm32: spdifrx: fix control DMA error management Documentation/devicetree/bindings/mfd/arizona.txt | 40 +- .../devicetree/bindings/sound/audio-graph-card.txt | 1 + .../bindings/sound/audio-graph-scu-card.txt | 5 +- .../devicetree/bindings/sound/cs42l56.txt | 2 +- Documentation/devicetree/bindings/sound/rt5514.txt | 13 +- Documentation/devicetree/bindings/sound/rt5663.txt | 16 + .../devicetree/bindings/sound/sgtl5000.txt | 2 +- .../devicetree/bindings/sound/st,stm32-sai.txt | 14 +- .../devicetree/bindings/sound/tfa9879.txt | 23 + .../devicetree/bindings/sound/wlf,arizona.txt | 53 ++ MAINTAINERS | 1 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 + drivers/gpu/drm/amd/include/amd_shared.h | 29 +- drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++---- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/arizona-core.c | 132 +---- drivers/mfd/wm97xx-core.c | 366 ++++++++++++++ include/drm/amd_asic_type.h | 52 ++ include/linux/mfd/arizona/pdata.h | 3 + include/linux/mfd/wm97xx.h | 25 + include/sound/ac97/codec.h | 118 +++++ include/sound/ac97/compat.h | 20 + include/sound/ac97/controller.h | 85 ++++ include/sound/ac97/regs.h | 262 ++++++++++ include/sound/ac97_codec.h | 239 +-------- include/sound/pxa2xx-lib.h | 15 +- include/sound/rt5651.h | 8 + include/sound/rt5663.h | 3 + include/sound/soc-acpi-intel-match.h | 32 ++ include/sound/soc-acpi.h | 111 +++++ include/sound/soc.h | 115 +++++ sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 + sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 16 + sound/ac97/bus.c | 539 +++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 108 +++++ sound/arm/pxa2xx-ac97-lib.c | 37 +- sound/arm/pxa2xx-ac97.c | 35 +- sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 + sound/soc/amd/Kconfig | 7 + sound/soc/amd/Makefile | 6 +- sound/soc/amd/acp-pcm-dma.c | 356 ++++++++++---- sound/soc/amd/acp-rt5645.c | 199 ++++++++ sound/soc/amd/acp.h | 19 + sound/soc/bcm/bcm2835-i2s.c | 391 +++++++++++---- sound/soc/bcm/cygnus-ssp.c | 34 +- sound/soc/codecs/Kconfig | 13 +- sound/soc/codecs/Makefile | 1 + sound/soc/codecs/arizona.c | 166 ++++++- sound/soc/codecs/arizona.h | 6 +- sound/soc/codecs/cs43130.c | 16 +- sound/soc/codecs/cs47l24.c | 14 +- sound/soc/codecs/da7213.c | 58 ++- sound/soc/codecs/da7213.h | 1 + sound/soc/codecs/hdac_hdmi.c | 51 +- sound/soc/codecs/hdmi-codec.c | 5 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98925.c | 23 +- sound/soc/codecs/max98927.c | 155 ++++-- sound/soc/codecs/max98927.h | 7 +- sound/soc/codecs/msm8916-wcd-analog.c | 120 ++--- sound/soc/codecs/msm8916-wcd-digital.c | 4 +- sound/soc/codecs/pcm512x-i2c.c | 4 +- sound/soc/codecs/pcm512x-spi.c | 2 +- sound/soc/codecs/pcm512x.c | 4 +- sound/soc/codecs/pcm512x.h | 2 +- sound/soc/codecs/rl6231.c | 5 +- sound/soc/codecs/rt5514-spi.c | 46 ++ sound/soc/codecs/rt5514.c | 6 +- sound/soc/codecs/rt5645.c | 32 +- sound/soc/codecs/rt5651.c | 219 ++++++++- sound/soc/codecs/rt5651.h | 4 + sound/soc/codecs/rt5659.c | 26 +- sound/soc/codecs/rt5663.c | 272 ++++++++++- sound/soc/codecs/rt5670.c | 143 +++++- sound/soc/codecs/rt5670.h | 4 + sound/soc/codecs/tas571x.c | 3 +- sound/soc/codecs/tfa9879.c | 6 + sound/soc/codecs/tlv320aic23.c | 1 + sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tpa6130a2.c | 1 + sound/soc/codecs/ts3a227e.c | 10 + sound/soc/codecs/wm5102.c | 14 +- sound/soc/codecs/wm5110.c | 14 +- sound/soc/codecs/wm8741.c | 39 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8993.c | 2 + sound/soc/codecs/wm8994.c | 2 + sound/soc/codecs/wm8997.c | 15 +- sound/soc/codecs/wm8998.c | 95 ++-- sound/soc/codecs/wm9705.c | 68 ++- sound/soc/codecs/wm9712.c | 48 +- sound/soc/codecs/wm9713.c | 39 +- sound/soc/davinci/davinci-mcasp.c | 21 +- sound/soc/dwc/Kconfig | 4 +- sound/soc/fsl/fsl-asoc-card.c | 14 +- sound/soc/fsl/fsl_spdif.c | 4 +- sound/soc/fsl/fsl_ssi.c | 46 +- sound/soc/generic/audio-graph-card.c | 47 +- sound/soc/img/img-i2s-in.c | 130 ++++- sound/soc/img/img-i2s-out.c | 120 +++-- sound/soc/img/img-parallel-out.c | 6 + sound/soc/img/img-spdif-in.c | 110 ++++- sound/soc/img/img-spdif-out.c | 87 +++- sound/soc/intel/Kconfig | 302 +----------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst-mfld-platform-compress.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 312 +----------- sound/soc/intel/atom/sst/sst_loader.c | 1 - sound/soc/intel/atom/sst/sst_stream.c | 1 - sound/soc/intel/boards/Kconfig | 265 ++++++++++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 16 +- sound/soc/intel/boards/bytcht_da7213.c | 23 +- sound/soc/intel/boards/bytcht_es8316.c | 27 +- sound/soc/intel/boards/bytcht_nocodec.c | 10 +- sound/soc/intel/boards/bytcr_rt5640.c | 118 ++--- sound/soc/intel/boards/bytcr_rt5651.c | 297 ++++++++++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 185 +++++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 128 ++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 68 +-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 76 ++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 + sound/soc/intel/boards/skl_nau88l25_max98357a.c | 16 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +- sound/soc/intel/common/Makefile | 4 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 196 ++++++++ sound/soc/intel/common/soc-acpi-intel-cht-match.c | 194 ++++++++ .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 64 +++ sound/soc/intel/common/sst-acpi.c | 36 +- sound/soc/intel/common/sst-acpi.h | 82 ---- sound/soc/intel/common/sst-firmware.c | 3 +- sound/soc/intel/skylake/skl-messages.c | 32 +- sound/soc/intel/skylake/skl-nhlt.c | 9 + sound/soc/intel/skylake/skl-pcm.c | 49 +- sound/soc/intel/skylake/skl-sst-utils.c | 15 +- sound/soc/intel/skylake/skl-topology.c | 70 ++- sound/soc/intel/skylake/skl-topology.h | 10 +- sound/soc/intel/skylake/skl.c | 50 +- sound/soc/intel/skylake/skl.h | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/kirkwood/kirkwood.h | 2 +- sound/soc/omap/omap-hdmi-audio.c | 3 + sound/soc/pxa/pxa2xx-ac97.c | 32 +- sound/soc/qcom/lpass-platform.c | 2 - sound/soc/rockchip/rk3399_gru_sound.c | 159 +++++- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/samsung/i2s.c | 10 +- sound/soc/samsung/i2s.h | 3 + sound/soc/samsung/tm2_wm5110.c | 7 +- sound/soc/sh/fsi.c | 11 +- sound/soc/sh/rcar/adg.c | 72 ++- sound/soc/sh/rcar/core.c | 51 +- sound/soc/sh/rcar/ctu.c | 88 ++-- sound/soc/sh/rcar/dma.c | 84 ++-- sound/soc/sh/rcar/dvc.c | 60 +-- sound/soc/sh/rcar/mix.c | 158 +++++- sound/soc/sh/rcar/rsnd.h | 22 +- sound/soc/sh/rcar/ssi.c | 58 ++- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 +-- sound/soc/soc-compress.c | 461 ++++++++++++++++-- sound/soc/soc-core.c | 222 ++++++--- sound/soc/soc-dapm.c | 158 +++--- sound/soc/soc-io.c | 14 + sound/soc/soc-pcm.c | 462 ++++++++++++++++-- sound/soc/stm/stm32_sai.c | 162 ++++++- sound/soc/stm/stm32_sai.h | 22 +- sound/soc/stm/stm32_sai_sub.c | 162 +++++-- sound/soc/stm/stm32_spdifrx.c | 23 +- sound/soc/sunxi/sun4i-codec.c | 29 +- sound/soc/sunxi/sun8i-codec.c | 72 ++- sound/soc/zte/zx-spdif.c | 4 +- 178 files changed, 8694 insertions(+), 2886 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tfa9879.txt create mode 100644 Documentation/devicetree/bindings/sound/wlf,arizona.txt create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/drm/amd_asic_type.h create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 include/sound/soc-acpi-intel-match.h create mode 100644 include/sound/soc-acpi.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c create mode 100644 sound/soc/amd/acp-rt5645.c create mode 100644 sound/soc/intel/boards/Kconfig create mode 100644 sound/soc/intel/common/soc-acpi-intel-byt-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cht-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c delete mode 100644 sound/soc/intel/common/sst-acpi.h rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%)

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[alsa-devel] [PATCH] ALSA: hda/realtek - Enable jack detection for Intel CFL + ALC700
by Guneshwor Singh 15 Nov '17

15 Nov '17

From: PeiSen Hou <pshou(a)realtek.com> ALC700 has jack detection disabled by default in Intel Coffeelake Reference board. To enable it, this fixup is required. Signed-off-by: PeiSen Hou <pshou(a)realtek.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/pci/hda/patch_realtek.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ce71111b4e3..61ef08173b1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5264,6 +5264,7 @@ enum { ALC233_FIXUP_LENOVO_MULTI_CODECS, ALC294_FIXUP_LENOVO_MIC_LOCATION, ALC700_FIXUP_INTEL_REFERENCE, + ALC700_FIXUP_INTEL_CFL, }; static const struct hda_fixup alc269_fixups[] = { @@ -6074,6 +6075,21 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC700_FIXUP_INTEL_CFL] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Enables internal speaker */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x45}, + {0x20, AC_VERB_SET_PROC_COEF, 0x5289}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x4a}, + {0x20, AC_VERB_SET_PROC_COEF, 0x201b}, + {0x58, AC_VERB_SET_COEF_INDEX, 0x00}, + {0x58, AC_VERB_SET_PROC_COEF, 0x3888}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x6f}, + {0x20, AC_VERB_SET_PROC_COEF, 0x2c0b}, + {} + } + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6277,6 +6293,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x8086, 0x7270, "Intel Coffeelake", ALC700_FIXUP_INTEL_CFL), #if 0 /* Below is a quirk table taken from the old code. -- 2.15.0

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[alsa-devel] [PATCH v3] ASoC: Replace snd_soc_acpi_check_hid with acpi_dev_present
by Jeremy Cline 14 Nov '17

14 Nov '17

Replace snd_soc_acpi_check_hid() with the generic acpi_dev_present() and remove the now unused snd_soc_acpi_check_hid function. This should have no functional change. Signed-off-by: Jeremy Cline <jeremy(a)jcline.org> --- Changes in v3: - Remove the unnecessary "== true" and "!= true" include/sound/soc-acpi.h | 3 --- sound/soc/soc-acpi.c | 32 ++------------------------------ 2 files changed, 2 insertions(+), 33 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index a7d8d335b043..057805489af3 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -49,9 +49,6 @@ snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines); -/* acpi check hid */ -bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); - /** * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are * related to the hardware, except for the firmware and topology file names. diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index f21df28bc28e..3543a7749490 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -49,41 +49,13 @@ const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) } EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid); -static acpi_status snd_soc_acpi_mach_match(acpi_handle handle, u32 level, - void *context, void **ret) -{ - unsigned long long sta; - acpi_status status; - - *(bool *)context = true; - status = acpi_evaluate_integer(handle, "_STA", NULL, &sta); - if (ACPI_FAILURE(status) || !(sta & ACPI_STA_DEVICE_PRESENT)) - *(bool *)context = false; - - return AE_OK; -} - -bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) -{ - acpi_status status; - bool found = false; - - status = acpi_get_devices(hid, snd_soc_acpi_mach_match, &found, NULL); - - if (ACPI_FAILURE(status)) - return false; - - return found; -} -EXPORT_SYMBOL_GPL(snd_soc_acpi_check_hid); - struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { struct snd_soc_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (snd_soc_acpi_check_hid(mach->id) == true) { + if (acpi_dev_present(mach->id, NULL, -1)) { if (mach->machine_quirk == NULL) return mach; @@ -163,7 +135,7 @@ struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (snd_soc_acpi_check_hid(codec_list->codecs[i]) != true) + if (!acpi_dev_present(codec_list->codecs[i], NULL, -1)) return NULL; } -- 2.14.3

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[alsa-devel] [PATCH 0/3] ALSA: es1938: Fine-tuning for seven function implementations
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 14 Nov 2017 21:30:12 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 14 function calls together with a variable assignment Use common error handling code in snd_es1938_probe() Use common error handling code in snd_es1938_create() sound/pci/es1938.c | 117 ++++++++++++++++++++++++++++++----------------------- 1 file changed, 67 insertions(+), 50 deletions(-) -- 2.15.0

1 3

[alsa-devel] [PATCH] Add support for Tempo Semiconductor's TSCS42xx audio CODEC
by Steven Eckhoff 14 Nov '17

14 Nov '17

Currently there is no support for the TSCS42xx audio CODEC. Add support for it. Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> Cc: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- .../ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt | 185 ++ .../devicetree/bindings/sound/tscs42xx.txt | 24 + .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 5 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tscs42xx.c | 2006 +++++++++++++++ sound/soc/codecs/tscs42xx.h | 2693 ++++++++++++++++++++ 8 files changed, 4924 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt create mode 100644 Documentation/devicetree/bindings/sound/tscs42xx.txt create mode 100644 sound/soc/codecs/tscs42xx.c create mode 100644 sound/soc/codecs/tscs42xx.h diff --git a/Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt b/Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt new file mode 100644 index 000000000000..39cd8912c020 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-bus-i2c-devices-tscs42xx.txt @@ -0,0 +1,185 @@ +What: /sys/bus/i2c/devices/.../dsp/eq[n]_ch[n]_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the equalizer biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/eq[n]_ch[n]_prescale +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the prescaler attenuation of the equalizer + biquad cascade + +What: /sys/bus/i2c/devices/.../dsp/bass_ext[n]_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the bass extraction biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/bass_lmt_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the bass non linear function coefficient + +What: /sys/bus/i2c/devices/.../dsp/bass_cto_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the bass cut-off biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_ext[n]_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the treb extraction biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_lmt_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the treb non linear function coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_cto_[ab][n] +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the treb cut-off biquad coefficient + +What: /sys/bus/i2c/devices/.../dsp/3d_coef +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the 3d coefficient + +What: /sys/bus/i2c/devices/.../dsp/3d_mix +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets 3d mix coefficient + +What: /sys/bus/i2c/devices/.../dsp/bass_mix +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets bass mix coefficient + +What: /sys/bus/i2c/devices/.../dsp/treb_mix +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets treb mix coefficient + +What: /sys/bus/i2c/devices/.../controls/eq1_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one + +What: /sys/bus/i2c/devices/.../controls/eq1_bands_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one bands: + 0h = Pre-scale only + 1h = Pre-scale + EQ band 0 + 2h = Pre-scale + EQ bands 0 to 1 + ... + 6h = Pre-scale + EQ bands 0 to 5 + +What: /sys/bus/i2c/devices/.../controls/eq2_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one + +What: /sys/bus/i2c/devices/.../controls/eq2_bands_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables equalizer one bands: + 0h = Pre-scale only + 1h = Pre-scale + EQ band 0 + 2h = Pre-scale + EQ bands 0 to 1 + ... + 6h = Pre-scale + EQ bands 0 to 5 + +What: /sys/bus/i2c/devices/.../controls/cle_level_mode +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the compressor, limiter and expander level detection mode: + 0h = Average + 1h = Peak + +What: /sys/bus/i2c/devices/.../controls/cle_level_detect_window +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Sets the compressor, limiter and expander level detection mode + window: + 0h = Equivalent of 512 samples + at the selected Base Rate (~10-16ms) + 1h = Equivalent of 64 samples + at the selected Base Rate (~1.3-2ms) + +What: /sys/bus/i2c/devices/.../controls/exp_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the expander + +What: /sys/bus/i2c/devices/.../controls/limit_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the limiter + +What: /sys/bus/i2c/devices/.../controls/comp_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the compressor + +What: /sys/bus/i2c/devices/.../controls/3d_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the 3D enhancer + +What: /sys/bus/i2c/devices/.../controls/treb_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the treble enhancer + +What: /sys/bus/i2c/devices/.../controls/treb_nlf_bypass +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the treble non-linear function bypass + +What: /sys/bus/i2c/devices/.../controls/bass_en +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the bass enhancer + +What: /sys/bus/i2c/devices/.../controls/bass_nlf_bypass +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Enables the bass non-linear function bypass + +What: /sys/bus/i2c/devices/.../control_regs/export +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Adds the specified control register to the sysfs interface + +What: /sys/bus/i2c/devices/.../control_regs/unexport +Date: November 2017 +KernelVersion: 4.14 +Contact: Steven Eckhoff <steve.eckhoff.opensource(a)gmail.com> +Description: Removes the specified control register to the sysfs interface diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt new file mode 100644 index 000000000000..2f9c8affde48 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -0,0 +1,24 @@ +TSCS42XX Audio CODEC + +Required Properties: + + - compatible : "tscs:tscs42xx" + + - reg : The I2C address of the device + + - mclk-src : The source of the master clock. This can be "mclk" or "xtal". + + - mclk-src-freq : The frequency of the master clock + + - clocks : The phandle of the master clock + +Example: + +codec: tscs42xx@69 { + compatible = "tscs,tscs42xx"; + reg = <0x69>; + mclk-src = "mclk"; + mclk-src-freq = <25000000>; + clocks = <&cprman BCM2835_CLOCK_GP1>; + status = "okay"; +}; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index b1eeca851d6f..24036f503aa0 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -349,6 +349,7 @@ tpo TPO tronfy Tronfy tronsmart Tronsmart truly Truly Semiconductors Limited +tscs Tempo Semiconductor tsd Theobroma Systems Design und Consulting GmbH tyan Tyan Computer Corporation ucrobotics uCRobotics diff --git a/MAINTAINERS b/MAINTAINERS index 650aa0e0e9d8..63d1d892842a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13674,6 +13674,11 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial.git S: Maintained K: ^Subject:.*(?i)trivial +TEMPO SEMICONDUCTOR DRIVERS +M: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +S: Maintained +F: sound/soc/codecs/tscs42xx.* + TTY LAYER M: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> M: Jiri Slaby <jslaby(a)suse.com> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c367d11079bc..34e911c1e082 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -217,6 +217,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_TSCS42XX if I2C help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -1142,6 +1143,13 @@ config SND_SOC_ZX_AUD96P22 depends on I2C select REGMAP_I2C +config SND_SOC_TSCS42XX + tristate "Tempo Semiconductor TSCS42xx CODEC" + depends on I2C + select REGMAP_I2C + help + Add support for Tempo Semiconductor's TSCS42xx audio CODEC. + # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 05018b7ca72b..a20e643b4067 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -230,6 +230,7 @@ snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o +snd-soc-tscs42xx-objs := tscs42xx.o # Amp snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o @@ -468,6 +469,7 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o +obj-$(CONFIG_SND_SOC_TSCS42XX) += snd-soc-tscs42xx.o # Amp obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c new file mode 100644 index 000000000000..dad874a19c58 --- /dev/null +++ b/sound/soc/codecs/tscs42xx.c @@ -0,0 +1,2006 @@ +/* + * tscs42xx.c -- TSCS42xx ALSA SoC Audio driver + * + * Copyright 2017 Tempo Semiconductor, Inc. + * + * Author: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/moduleparam.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/mutex.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/firmware.h> +#include <linux/sysfs.h> + +#include "tscs42xx.h" + +enum { + PLL_SRC_CLK_XTAL, + PLL_SRC_CLK_MCLK2, +}; + +/* + * Functions that receive a pointer to data + * should assume that the data is locked. + */ +struct tscs42xx_priv { + struct regmap *regmap; + struct clk *mclk; + int mclk_src_freq; + int pll_src_clk; + int bclk_ratio; + int samplerate; + int pll_users; + struct snd_soc_codec *codec; + struct mutex lock; +}; + +static bool tscs42xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case R_DACCRWRL: + case R_DACCRWRM: + case R_DACCRWRH: + case R_DACCRRDL: + case R_DACCRRDM: + case R_DACCRRDH: + case R_DACCRSTAT: + case R_DACCRADDR: + case R_PLLCTL0: + return true; + default: + return false; + }; +} + +static bool tscs42xx_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case R_DACCRWRL: + case R_DACCRWRM: + case R_DACCRWRH: + case R_DACCRRDL: + case R_DACCRRDM: + case R_DACCRRDH: + return true; + default: + return false; + }; +} + +static const struct regmap_config tscs42xx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .volatile_reg = tscs42xx_volatile, + .precious_reg = tscs42xx_precious, + .max_register = R_DACMBCREL3H, + + .cache_type = REGCACHE_RBTREE, +}; + +static struct reg_default r_inits[] = { + { .reg = R_ADCSR, .def = RV_ADCSR_ABCM_64 }, + { .reg = R_DACSR, .def = RV_DACSR_DBCM_64 }, + { .reg = R_AIC2, .def = RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, +}; + +#define NUM_DACCR_BYTES 3 +static int load_dac_coefficient_ram(struct snd_soc_codec *codec) +{ + const struct firmware *fw = NULL; + int ret; + int i; + int j; + int addr; + + ret = request_firmware_direct(&fw, "tscs42xx_daccram.dfw", codec->dev); + if (ret) { + dev_info(codec->dev, + "No tscs42xx_daccram.dfw file found (%d)\n", ret); + return 0; + } + + if (fw->size % NUM_DACCR_BYTES != 0) { + ret = -EINVAL; + dev_err(codec->dev, "Malformed daccram file (%d)\n", ret); + return ret; + } + + for (i = 0, addr = 0; i < fw->size; i += NUM_DACCR_BYTES, addr++) { + + do { + ret = snd_soc_read(codec, R_DACCRSTAT); + } while (ret > 0); + + if (ret < 0) { + dev_err(codec->dev, + "Failed to read daccrstat (%d)\n", ret); + return ret; + } + + /* Explicit address update */ + ret = snd_soc_write(codec, R_DACCRADDR, addr); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write DACCRADDR (%d)\n", ret); + return ret; + } + + for (j = 0; j < NUM_DACCR_BYTES; j++) { /* FW in big endian */ + + do { + ret = snd_soc_read(codec, R_DACCRSTAT); + } while (ret > 0); + + if (ret < 0) { + dev_err(codec->dev, + "Failed to read daccrstat (%d)\n", ret); + return ret; + } + + /* Write LSB first due to auto increment on MSB */ + ret = snd_soc_write(codec, R_DACCRWRL + j, + fw->data[i + NUM_DACCR_BYTES - 1 - j]); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write coefficient (%d)\n", + ret); + return ret; + } + } + } + + dev_info(codec->dev, "Loaded tscs42xx_daccram.dfw\n"); + + return 0; +} + +#define NUM_CONTROL_BYTES 2 +static int load_control_regs(struct snd_soc_codec *codec) +{ + const struct firmware *fw = NULL; + int ret; + int i; + + ret = request_firmware_direct(&fw, "tscs42xx_controls.dfw", codec->dev); + if (ret) { + dev_info(codec->dev, + "No tscs42xx_controls.dfw file found (%d)\n", ret); + return 0; + } + + if (fw->size % NUM_CONTROL_BYTES != 0) { + ret = -EINVAL; + dev_err(codec->dev, "Malformed controls file (%d)\n", ret); + return ret; + } + + for (i = 0; i < fw->size; i += NUM_CONTROL_BYTES) { + ret = snd_soc_write(codec, fw->data[i], fw->data[i + 1]); + if (ret < 0) { + dev_err(codec->dev, "Failed to write control (%d)\n", + ret); + return ret; + } + } + + dev_info(codec->dev, "Loaded tscs42xx_controls.dfw\n"); + + return 0; +} + +#define PLL_LOCK_TIME_MAX 10 +static bool plls_locked(struct snd_soc_codec *codec) +{ + int reg; + int count; + bool ret = false; + + for (count = 0; count < PLL_LOCK_TIME_MAX; count++) { + reg = snd_soc_read(codec, R_PLLCTL0); + if (reg < 0) { + dev_err(codec->dev, + "Failed to read PLL lock status (%d)\n", ret); + break; + } else if (reg > 0) { + ret = true; + break; + } + mdelay(1); + } + + return ret; +} + +/* D2S Input Select */ +static const char * const d2s_input_select_text[] = { + "Line 1", "Line 2" +}; + +static const struct soc_enum d2s_input_select_enum = +SOC_ENUM_SINGLE(R_INMODE, FB_INMODE_DS, ARRAY_SIZE(d2s_input_select_text), + d2s_input_select_text); + +static const struct snd_kcontrol_new d2s_input_mux = +SOC_DAPM_ENUM("D2S_IN_MUX", d2s_input_select_enum); + +/* Input L Capture Route */ +static const char * const input_select_text[] = { + "Line 1", "Line 2", "Line 3", "D2S" +}; + +static const struct soc_enum left_input_select_enum = +SOC_ENUM_SINGLE(R_INSELL, FB_INSELL, ARRAY_SIZE(input_select_text), + input_select_text); + +static const struct snd_kcontrol_new left_input_select = +SOC_DAPM_ENUM("LEFT_INPUT_SELECT_ENUM", left_input_select_enum); + +/* Input R Capture Route */ +static const struct soc_enum right_input_select_enum = +SOC_ENUM_SINGLE(R_INSELR, FB_INSELR, ARRAY_SIZE(input_select_text), + input_select_text); + +static const struct snd_kcontrol_new right_input_select = +SOC_DAPM_ENUM("RIGHT_INPUT_SELECT_ENUM", right_input_select_enum); + +/* Input Channel Mapping */ +static const char * const ch_map_select_text[] = { + "Normal", "Left to Right", "Right to Left", "Swap" +}; + +static const struct soc_enum ch_map_select_enum = +SOC_ENUM_SINGLE(R_AIC2, FB_AIC2_ADCDSEL, ARRAY_SIZE(ch_map_select_text), + ch_map_select_text); + +static int dapm_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + mdelay(5); + return 0; +} + +static int dapm_micb_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + mdelay(5); + return 0; +} + +static const struct snd_soc_dapm_widget tscs42xx_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("Vref", 1, R_PWRM2, FB_PWRM2_VREF, 0, + dapm_vref_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + + /* Headphone */ + SND_SOC_DAPM_DAC("DAC L", "HiFi Playback", R_PWRM2, FB_PWRM2_HPL, 0), + SND_SOC_DAPM_DAC("DAC R", "HiFi Playback", R_PWRM2, FB_PWRM2_HPR, 0), + SND_SOC_DAPM_OUTPUT("Headphone L"), + SND_SOC_DAPM_OUTPUT("Headphone R"), + + /* Speaker */ + SND_SOC_DAPM_DAC("ClassD L", "HiFi Playback", + R_PWRM2, FB_PWRM2_SPKL, 0), + SND_SOC_DAPM_DAC("ClassD R", "HiFi Playback", + R_PWRM2, FB_PWRM2_SPKR, 0), + SND_SOC_DAPM_OUTPUT("Speaker L"), + SND_SOC_DAPM_OUTPUT("Speaker R"), + + /* Capture */ + SND_SOC_DAPM_PGA("Analog In PGA L", R_PWRM1, FB_PWRM1_PGAL, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog In PGA R", R_PWRM1, FB_PWRM1_PGAR, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog Boost L", R_PWRM1, FB_PWRM1_BSTL, 0, NULL, 0), + SND_SOC_DAPM_PGA("Analog Boost R", R_PWRM1, FB_PWRM1_BSTR, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC Mute", R_CNVRTR0, FB_CNVRTR0_HPOR, true, NULL, 0), + SND_SOC_DAPM_ADC("ADC L", "HiFi Capture", R_PWRM1, FB_PWRM1_ADCL, 0), + SND_SOC_DAPM_ADC("ADC R", "HiFi Capture", R_PWRM1, FB_PWRM1_ADCR, 0), + + /* Capture Input */ + SND_SOC_DAPM_MUX("Input L Capture Route", R_PWRM2, + FB_PWRM2_INSELL, 0, &left_input_select), + SND_SOC_DAPM_MUX("Input R Capture Route", R_PWRM2, + FB_PWRM2_INSELR, 0, &right_input_select), + + /* Digital Mic */ + SND_SOC_DAPM_SUPPLY_S("Digital Mic Enable", 2, R_DMICCTL, + FB_DMICCTL_DMICEN, 0, NULL, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_INPUT("Digital Mic L"), + SND_SOC_DAPM_INPUT("Digital Mic R"), + + /* Analog Mic */ + SND_SOC_DAPM_SUPPLY_S("Mic Bias", 2, R_PWRM1, FB_PWRM1_MICB, + 0, dapm_micb_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + + /* Line In */ + SND_SOC_DAPM_INPUT("Line In 1 L"), + SND_SOC_DAPM_INPUT("Line In 1 R"), + SND_SOC_DAPM_INPUT("Line In 2 L"), + SND_SOC_DAPM_INPUT("Line In 2 R"), + SND_SOC_DAPM_INPUT("Line In 3 L"), + SND_SOC_DAPM_INPUT("Line In 3 R"), +}; + +static const struct snd_soc_dapm_route tscs42xx_intercon[] = { + {"DAC L", NULL, "Vref"}, + {"DAC R", NULL, "Vref"}, + {"Headphone L", NULL, "DAC L"}, + {"Headphone R", NULL, "DAC R"}, + + {"ClassD L", NULL, "Vref"}, + {"ClassD R", NULL, "Vref"}, + {"Speaker L", NULL, "ClassD L"}, + {"Speaker R", NULL, "ClassD R"}, + + {"Input L Capture Route", NULL, "Vref"}, + {"Input R Capture Route", NULL, "Vref"}, + + {"Mic Bias", NULL, "Vref"}, + + {"Input L Capture Route", "Line 1", "Line In 1 L"}, + {"Input R Capture Route", "Line 1", "Line In 1 R"}, + {"Input L Capture Route", "Line 2", "Line In 2 L"}, + {"Input R Capture Route", "Line 2", "Line In 2 R"}, + {"Input L Capture Route", "Line 3", "Line In 3 L"}, + {"Input R Capture Route", "Line 3", "Line In 3 R"}, + + {"Analog In PGA L", NULL, "Input L Capture Route"}, + {"Analog In PGA R", NULL, "Input R Capture Route"}, + {"Analog Boost L", NULL, "Analog In PGA L"}, + {"Analog Boost R", NULL, "Analog In PGA R"}, + {"ADC Mute", NULL, "Analog Boost L"}, + {"ADC Mute", NULL, "Analog Boost R"}, + {"ADC L", NULL, "ADC Mute"}, + {"ADC R", NULL, "ADC Mute"}, +}; + +/*********** + * Volumes * + ***********/ + +static DECLARE_TLV_DB_SCALE(hpvol, -8850, 75, 0); +static DECLARE_TLV_DB_SCALE(spkvol, -7725, 75, 0); +static DECLARE_TLV_DB_SCALE(dacvol, -9563, 38, 0); +static DECLARE_TLV_DB_SCALE(adcvol, -7125, 38, 0); +static DECLARE_TLV_DB_SCALE(invol, -1725, 75, 0); + +/********* + * INSEL * + *********/ + +static DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 1000, 0); + +/************ + * CONTROLS * + ************/ +static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { + /* Volumes */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR, + FB_HPVOLL, 0x7F, 0, hpvol), + SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR, + FB_SPKVOLL, 0x7F, 0, spkvol), + SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR, + FB_DACVOLL, 0xFF, 0, dacvol), + SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR, + FB_ADCVOLL, 0xFF, 0, adcvol), + SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR, + FB_INVOLL, 0x3F, 0, invol), + + /* INSEL */ + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR, + FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB, + 0, mic_boost_tlv), + + /* Input Channel Map */ + SOC_ENUM("Input Channel Map Switch", ch_map_select_enum), +}; + +#define TSCS42XX_RATES SNDRV_PCM_RATE_8000_96000 + +#define TSCS42XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int setup_sample_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + unsigned int width; + int ret; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + width = RV_AIC1_WL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + width = RV_AIC1_WL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + width = RV_AIC1_WL_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + width = RV_AIC1_WL_32; + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unsupported format width (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_AIC1, RM_AIC1_WL, width); + if (ret < 0) { + dev_err(codec->dev, "Failed to set sample width (%d)\n", ret); + return ret; + } + + return 0; +} + +static int setup_sample_rate(struct snd_soc_codec *codec, + unsigned int rate, + struct tscs42xx_priv *tscs42xx) +{ + unsigned int br, bm; + int ret; + + switch (rate) { + case 8000: + br = RV_DACSR_DBR_32; + bm = RV_DACSR_DBM_PT25; + break; + case 16000: + br = RV_DACSR_DBR_32; + bm = RV_DACSR_DBM_PT5; + break; + case 24000: + br = RV_DACSR_DBR_48; + bm = RV_DACSR_DBM_PT5; + break; + case 32000: + br = RV_DACSR_DBR_32; + bm = RV_DACSR_DBM_1; + break; + case 48000: + br = RV_DACSR_DBR_48; + bm = RV_DACSR_DBM_1; + break; + case 96000: + br = RV_DACSR_DBR_48; + bm = RV_DACSR_DBM_2; + break; + case 11025: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_PT25; + break; + case 22050: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_PT5; + break; + case 44100: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_1; + break; + case 88200: + br = RV_DACSR_DBR_44_1; + bm = RV_DACSR_DBM_2; + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", rate); + return -EINVAL; + } + + /* DAC and ADC share bit and frame clock */ + ret = snd_soc_update_bits(codec, R_DACSR, RM_DACSR_DBR, br); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_DACSR, RM_DACSR_DBM, bm); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_ADCSR, RM_DACSR_DBR, br); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_ADCSR, RM_DACSR_DBM, bm); + if (ret < 0) { + dev_err(codec->dev, "Failed to update register (%d)\n", ret); + return ret; + } + + tscs42xx->samplerate = rate; + + return 0; +} + +struct reg_setting { + unsigned int addr; + unsigned int val; + unsigned int mask; +}; + +#define PLL_REG_SETTINGS_COUNT 13 +struct pll_ctl { + int input_freq; + struct reg_setting settings[PLL_REG_SETTINGS_COUNT]; +}; + +#define PLL_CTL(f, rt, rd, r1b_l, r9, ra, rb, \ + rc, r12, r1b_h, re, rf, r10, r11) \ + { \ + .input_freq = f, \ + .settings = { \ + {R_TIMEBASE, rt, 0xFF}, \ + {R_PLLCTLD, rd, 0xFF}, \ + {R_PLLCTL1B, r1b_l, 0x0F}, \ + {R_PLLCTL9, r9, 0xFF}, \ + {R_PLLCTLA, ra, 0xFF}, \ + {R_PLLCTLB, rb, 0xFF}, \ + {R_PLLCTLC, rc, 0xFF}, \ + {R_PLLCTL12, r12, 0xFF}, \ + {R_PLLCTL1B, r1b_h, 0xF0}, \ + {R_PLLCTLE, re, 0xFF}, \ + {R_PLLCTLF, rf, 0xFF}, \ + {R_PLLCTL10, r10, 0xFF}, \ + {R_PLLCTL11, r11, 0xFF}, \ + }, \ + } + +static const struct pll_ctl pll_ctls[] = { + PLL_CTL(1411200, 0x05, + 0x39, 0x04, 0x07, 0x02, 0xC3, 0x04, + 0x1B, 0x10, 0x03, 0x03, 0xD0, 0x02), + PLL_CTL(1536000, 0x05, + 0x1A, 0x04, 0x02, 0x03, 0xE0, 0x01, + 0x1A, 0x10, 0x02, 0x03, 0xB9, 0x01), + PLL_CTL(2822400, 0x0A, + 0x23, 0x04, 0x07, 0x04, 0xC3, 0x04, + 0x22, 0x10, 0x05, 0x03, 0x58, 0x02), + PLL_CTL(3072000, 0x0B, + 0x22, 0x04, 0x07, 0x03, 0x48, 0x03, + 0x1A, 0x10, 0x04, 0x03, 0xB9, 0x01), + PLL_CTL(5644800, 0x15, + 0x23, 0x04, 0x0E, 0x04, 0xC3, 0x04, + 0x1A, 0x10, 0x08, 0x03, 0xE0, 0x01), + PLL_CTL(6144000, 0x17, + 0x1A, 0x04, 0x08, 0x03, 0xE0, 0x01, + 0x1A, 0x10, 0x08, 0x03, 0xB9, 0x01), + PLL_CTL(12000000, 0x2E, + 0x1B, 0x04, 0x19, 0x03, 0x00, 0x03, + 0x2A, 0x10, 0x19, 0x05, 0x98, 0x04), + PLL_CTL(19200000, 0x4A, + 0x13, 0x04, 0x14, 0x03, 0x80, 0x01, + 0x1A, 0x10, 0x19, 0x03, 0xB9, 0x01), + PLL_CTL(22000000, 0x55, + 0x2A, 0x04, 0x37, 0x05, 0x00, 0x06, + 0x22, 0x10, 0x26, 0x03, 0x49, 0x02), + PLL_CTL(22579200, 0x57, + 0x22, 0x04, 0x31, 0x03, 0x20, 0x03, + 0x1A, 0x10, 0x1D, 0x03, 0xB3, 0x01), + PLL_CTL(24000000, 0x5D, + 0x13, 0x04, 0x19, 0x03, 0x80, 0x01, + 0x1B, 0x10, 0x19, 0x05, 0x4C, 0x02), + PLL_CTL(24576000, 0x5F, + 0x13, 0x04, 0x1D, 0x03, 0xB3, 0x01, + 0x22, 0x10, 0x40, 0x03, 0x72, 0x03), + PLL_CTL(27000000, 0x68, + 0x22, 0x04, 0x4B, 0x03, 0x00, 0x04, + 0x2A, 0x10, 0x7D, 0x03, 0x20, 0x06), + PLL_CTL(36000000, 0x8C, + 0x1B, 0x04, 0x4B, 0x03, 0x00, 0x03, + 0x2A, 0x10, 0x7D, 0x03, 0x98, 0x04), + PLL_CTL(25000000, 0x61, + 0x1B, 0x04, 0x37, 0x03, 0x2B, 0x03, + 0x1A, 0x10, 0x2A, 0x03, 0x39, 0x02), + PLL_CTL(26000000, 0x65, + 0x23, 0x04, 0x41, 0x05, 0x00, 0x06, + 0x1A, 0x10, 0x26, 0x03, 0xEF, 0x01), + PLL_CTL(12288000, 0x2F, + 0x1A, 0x04, 0x12, 0x03, 0x1C, 0x02, + 0x22, 0x10, 0x20, 0x03, 0x72, 0x03), + PLL_CTL(40000000, 0x9B, + 0x22, 0x08, 0x7D, 0x03, 0x80, 0x04, + 0x23, 0x10, 0x7D, 0x05, 0xE4, 0x06), + PLL_CTL(512000, 0x01, + 0x22, 0x04, 0x01, 0x03, 0xD0, 0x02, + 0x1B, 0x10, 0x01, 0x04, 0x72, 0x03), + PLL_CTL(705600, 0x02, + 0x22, 0x04, 0x02, 0x03, 0x15, 0x04, + 0x22, 0x10, 0x01, 0x04, 0x80, 0x02), + PLL_CTL(1024000, 0x03, + 0x22, 0x04, 0x02, 0x03, 0xD0, 0x02, + 0x1B, 0x10, 0x02, 0x04, 0x72, 0x03), + PLL_CTL(2048000, 0x07, + 0x22, 0x04, 0x04, 0x03, 0xD0, 0x02, + 0x1B, 0x10, 0x04, 0x04, 0x72, 0x03), + PLL_CTL(2400000, 0x08, + 0x22, 0x04, 0x05, 0x03, 0x00, 0x03, + 0x23, 0x10, 0x05, 0x05, 0x98, 0x04), +}; + +static const struct pll_ctl *get_pll_ctl(int input_freq) +{ + int i; + const struct pll_ctl *pll_ctl = NULL; + + for (i = 0; i < ARRAY_SIZE(pll_ctls); ++i) + if (input_freq == pll_ctls[i].input_freq) { + pll_ctl = &pll_ctls[i]; + break; + } + + return pll_ctl; +} + +static int sample_rate_to_pll_freq_out(int sample_rate) +{ + switch (sample_rate) { + case 11025: + case 22050: + case 44100: + case 88200: + return 112896000; + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + return 122880000; + default: + return -EINVAL; + } +} + + +static int set_pll_ctl_from_input_freq(struct snd_soc_codec *codec, + const int input_freq) +{ + int ret; + int i; + const struct pll_ctl *pll_ctl; + + pll_ctl = get_pll_ctl(input_freq); + if (!pll_ctl) { + ret = -EINVAL; + dev_err(codec->dev, "No PLL input entry for %d (%d)\n", + input_freq, ret); + return ret; + } + + for (i = 0; i < PLL_REG_SETTINGS_COUNT; ++i) { + ret = snd_soc_update_bits(codec, + pll_ctl->settings[i].addr, + pll_ctl->settings[i].mask, + pll_ctl->settings[i].val); + if (ret < 0) { + dev_err(codec->dev, "Failed to set pll ctl (%d)\n", + ret); + return ret; + } + } + + return 0; +} + +static int configure_clocks(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = set_pll_ctl_from_input_freq(codec, tscs42xx->mclk_src_freq); + if (ret < 0) { + dev_err(codec->dev, "Failed to setup PLL input (%d)\n", ret); + return ret; + } + + switch (tscs42xx->pll_src_clk) { + case PLL_SRC_CLK_XTAL: + ret = snd_soc_write(codec, R_PLLREFSEL, + RV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1 | + RV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set pll reference input (%d)\n", + ret); + return ret; + } + break; + case PLL_SRC_CLK_MCLK2: + ret = clk_set_rate(tscs42xx->mclk, tscs42xx->mclk_src_freq); + if (ret < 0) { + dev_err(codec->dev, + "Could not set mclk rate %d (%d)\n", + tscs42xx->mclk_src_freq, ret); + return ret; + } + + ret = clk_prepare_enable(tscs42xx->mclk); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable mclk: (%d)\n", + ret); + return ret; + } + + ret = snd_soc_write(codec, R_PLLREFSEL, + RV_PLLREFSEL_PLL1_REF_SEL_MCLK2 | + RV_PLLREFSEL_PLL2_REF_SEL_MCLK2); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set PLL reference (%d)\n", ret); + return ret; + } + break; + } + + return 0; +} + +static int power_down_audio_plls(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + tscs42xx->pll_users--; + if (tscs42xx->pll_users > 0) + return 0; + + ret = snd_soc_update_bits(codec, R_PLLCTL1C, + RM_PLLCTL1C_PDB_PLL1, + RV_PLLCTL1C_PDB_PLL1_DISABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to turn PLL off (%d)\n", ret); + return ret; + } + ret = snd_soc_update_bits(codec, R_PLLCTL1C, + RM_PLLCTL1C_PDB_PLL2, + RV_PLLCTL1C_PDB_PLL2_DISABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to turn PLL off (%d)\n", ret); + return ret; + } + + return 0; +} + +static int power_up_audio_plls(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int freq_out; + int ret; + unsigned int mask; + unsigned int val; + + freq_out = sample_rate_to_pll_freq_out(tscs42xx->samplerate); + switch (freq_out) { + case 122880000: /* 48k */ + mask = RM_PLLCTL1C_PDB_PLL1; + val = RV_PLLCTL1C_PDB_PLL1_ENABLE; + break; + case 112896000: /* 44.1k */ + mask = RM_PLLCTL1C_PDB_PLL2; + val = RV_PLLCTL1C_PDB_PLL2_ENABLE; + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unrecognized PLL output freq (%d)\n", ret); + return ret; + } + + ret = snd_soc_update_bits(codec, R_PLLCTL1C, mask, val); + if (ret < 0) { + dev_err(codec->dev, "Failed to turn PLL on (%d)\n", ret); + return ret; + } + + if (!plls_locked(codec)) { + dev_err(codec->dev, "Failed to lock plls\n"); + return -ENOMSG; + } + + tscs42xx->pll_users++; + + return 0; +} + +static int tscs42xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = setup_sample_format(codec, params_format(params)); + if (ret < 0) { + dev_err(codec->dev, "Failed to setup sample format (%d)\n", + ret); + goto exit; + } + + ret = setup_sample_rate(codec, params_rate(params), tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to setup sample rate (%d)\n", ret); + goto exit; + } + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int dac_mute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = snd_soc_update_bits(codec, R_CNVRTR1, RM_CNVRTR1_DACMU, + RV_CNVRTR1_DACMU_ENABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to mute DAC (%d)\n", + ret); + return ret; + } + + ret = power_down_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power down plls (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int dac_unmute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = power_up_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power up plls (%d)\n", + ret); + return ret; + } + + ret = snd_soc_update_bits(codec, R_CNVRTR1, RM_CNVRTR1_DACMU, + RV_CNVRTR1_DACMU_DISABLE); + if (ret < 0) { + power_down_audio_plls(codec, tscs42xx); + dev_err(codec->dev, "Failed to unmute DAC (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int adc_mute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = snd_soc_update_bits(codec, R_CNVRTR0, RM_CNVRTR0_ADCMU, + RV_CNVRTR0_ADCMU_ENABLE); + if (ret < 0) { + dev_err(codec->dev, "Failed to mute ADC (%d)\n", + ret); + return ret; + } + + ret = power_down_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power down plls (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int adc_unmute(struct snd_soc_codec *codec, + struct tscs42xx_priv *tscs42xx) +{ + int ret; + + ret = power_up_audio_plls(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to power up plls (%d)\n", + ret); + return ret; + } + + ret = snd_soc_update_bits(codec, R_CNVRTR0, RM_CNVRTR0_ADCMU, + RV_CNVRTR0_ADCMU_DISABLE); + if (ret < 0) { + power_down_audio_plls(codec, tscs42xx); + dev_err(codec->dev, "Failed to unmute ADC (%d)\n", + ret); + return ret; + } + + return 0; +} + +static int tscs42xx_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&tscs42xx->lock); + + if (mute) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = dac_mute(codec, tscs42xx); + else + ret = adc_mute(codec, tscs42xx); + else + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = dac_unmute(codec, tscs42xx); + else + ret = adc_unmute(codec, tscs42xx); + + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ret = snd_soc_update_bits(codec, R_AIC1, RM_AIC1_MS, + RV_AIC1_MS_MASTER); + if (ret < 0) + dev_err(codec->dev, + "Failed to set codec DAI master (%d)\n", ret); + else + ret = 0; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ret = -EINVAL; + dev_err(codec->dev, "tscs42xx slave mode not supported (%d)\n", + ret); + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unsupported format (%d)\n", ret); + break; + } + + return ret; +} + +static int tscs42xx_set_bclk_ratio(struct snd_soc_dai *codec_dai, + unsigned int ratio) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int ret = 0; + + mutex_lock(&tscs42xx->lock); + + switch (ratio) { + case 32: + value = RV_DACSR_DBCM_32; + break; + case 40: + value = RV_DACSR_DBCM_40; + break; + case 64: + value = RV_DACSR_DBCM_64; + break; + default: + ret = -EINVAL; + dev_err(codec->dev, "Unsupported bclk ratio (%d)\n", ret); + goto exit; + } + + ret = snd_soc_update_bits(codec, R_DACSR, RM_DACSR_DBCM, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set DAC BCLK ratio (%d)\n", ret); + goto exit; + } + ret = snd_soc_update_bits(codec, R_ADCSR, RM_ADCSR_ABCM, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set ADC BCLK ratio (%d)\n", ret); + goto exit; + } + + tscs42xx->bclk_ratio = ratio; + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver tscs42xx_dai = { + .name = "tscs42xx-HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TSCS42XX_RATES, + .formats = TSCS42XX_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = TSCS42XX_RATES, + .formats = TSCS42XX_FORMATS,}, + .ops = &tscs42xx_dai_ops, + .symmetric_rates = 1, +}; + +static int tscs42xx_i2c_read(struct i2c_client *i2c, u8 reg, u8 *val) +{ + int ret; + + ret = i2c_smbus_write_byte(i2c, reg); + if (ret < 0) { + dev_err(&i2c->dev, "I2C write failed (%d)\n", ret); + return ret; + } + + ret = i2c_smbus_read_byte(i2c); + if (ret < 0) { + dev_err(&i2c->dev, "I2C read failed (%d)\n", ret); + return ret; + } + + *val = (u8)ret; + + return 0; +} + +static int part_is_valid(struct i2c_client *i2c) +{ + int val; + int ret; + u8 reg; + + ret = tscs42xx_i2c_read(i2c, R_DEVIDH, &reg); + if (ret < 0) + return ret; + + val = reg << 8; + ret = tscs42xx_i2c_read(i2c, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + ret = true; + break; + default: + ret = false; + break; + }; + + if (ret) + dev_info(&i2c->dev, "Found part 0x%04x\n", val); + else + dev_err(&i2c->dev, "0x%04x is not a valid part\n", val); + + return ret; +} + +static int set_data_from_of(struct i2c_client *i2c, + struct tscs42xx_priv *tscs42xx) +{ + struct device_node *np = i2c->dev.of_node; + const char *mclk_src = NULL; + int ret; + + ret = of_property_read_string(np, "mclk-src", &mclk_src); + if (ret) { + dev_err(&i2c->dev, "mclk-src is needed (%d)\n", ret); + return ret; + } + + if (!strncmp(mclk_src, "mclk", 4)) { + tscs42xx->mclk = devm_clk_get(&i2c->dev, NULL); + if (IS_ERR(tscs42xx->mclk)) { + dev_info(&i2c->dev, "mclk not present trying again\n"); + return -EPROBE_DEFER; + } + tscs42xx->pll_src_clk = PLL_SRC_CLK_MCLK2; + } else if (!strncmp(mclk_src, "xtal", 4)) { + tscs42xx->pll_src_clk = PLL_SRC_CLK_XTAL; + } else { + dev_err(&i2c->dev, "mclk-src %s is unsupported\n", mclk_src); + return -EINVAL; + } + + ret = of_property_read_u32(np, "mclk-src-freq", + &tscs42xx->mclk_src_freq); + if (ret) { + dev_err(&i2c->dev, "mclk-src-freq not provided (%d)\n", ret); + return ret; + } + if (!get_pll_ctl(tscs42xx->mclk_src_freq)) { + dev_err(&i2c->dev, "mclk frequency unsupported\n"); + return ret; + } + + return 0; +} + +/******************* + * SYSFS Interface * + *******************/ +#define FMODE 0664 + +/* Control Reg Interface */ + +struct tempo_control_reg { + struct device *dev; + struct kobject *dir_kobj; + struct kobj_attribute val_kobj_attr; + struct kobj_attribute addr_kobj_attr; + const char *name; + const uint8_t addr; +}; + +static ssize_t ctrl_reg_val_show(struct kobject *kobj, + struct kobj_attribute *attr, char *buf) +{ + struct tempo_control_reg *control_reg = + container_of(attr, struct tempo_control_reg, val_kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control_reg->dev); + unsigned int show; + int ret = 0; + + ret = regmap_read(tscs42xx->regmap, + control_reg->addr, + &show); + if (ret < 0) + return ret; + + ret = sprintf(buf, "0x%02x\n", show); + + return ret; +} + +static ssize_t ctrl_reg_val_store(struct kobject *kobj, + struct kobj_attribute *attr, + const char *buf, size_t count) +{ + struct tempo_control_reg *control_reg = + container_of(attr, struct tempo_control_reg, val_kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control_reg->dev); + unsigned int store; + int ret; + + ret = kstrtoint(buf, 0, &store); + if (ret < 0) + return ret; + + ret = regmap_write(tscs42xx->regmap, control_reg->addr, store); + if (ret < 0) + return ret; + + return count; +} + +static ssize_t ctrl_reg_addr_show(struct kobject *kobj, + struct kobj_attribute *attr, char *buf) +{ + struct tempo_control_reg *control_reg = + container_of(attr, struct tempo_control_reg, addr_kobj_attr); + + return sprintf(buf, "0x%02x\n", control_reg->addr); +} + +#define TEMPO_CONTROL_REG(_name, _addr) \ + { \ + NULL, \ + NULL, \ + __ATTR(value, FMODE, ctrl_reg_val_show, \ + ctrl_reg_val_store), \ + __ATTR(address, FMODE, ctrl_reg_addr_show, \ + NULL), \ + .name = __stringify(_name), \ + .addr = _addr, \ + } + +static struct tempo_control_reg control_regs[] = { + TEMPO_CONTROL_REG(config0, R_CONFIG0), /* 0x1F */ + TEMPO_CONTROL_REG(config1, R_CONFIG1), /* 0x20 */ + TEMPO_CONTROL_REG(clectl, R_CLECTL), /* 0x25 */ + TEMPO_CONTROL_REG(mugain, R_MUGAIN), /* 0x26 */ + TEMPO_CONTROL_REG(compth, R_COMPTH), /* 0x27 */ + TEMPO_CONTROL_REG(cmprat, R_CMPRAT), /* 0x28 */ + TEMPO_CONTROL_REG(catktcl, R_CATKTCL), /* 0x29 */ + TEMPO_CONTROL_REG(catktch, R_CATKTCH), /* 0x2A */ + TEMPO_CONTROL_REG(creltcl, R_CRELTCL), /* 0x2B */ + TEMPO_CONTROL_REG(creltch, R_CRELTCH), /* 0x2C */ + TEMPO_CONTROL_REG(limth, R_LIMTH), /* 0x2D */ + TEMPO_CONTROL_REG(limtgt, R_LIMTGT), /* 0x2E */ + TEMPO_CONTROL_REG(latktcl, R_LATKTCL), /* 0x2F */ + TEMPO_CONTROL_REG(latktch, R_LATKTCH), /* 0x30 */ + TEMPO_CONTROL_REG(lreltcl, R_LRELTCL), /* 0x31 */ + TEMPO_CONTROL_REG(lreltch, R_LRELTCH), /* 0x32 */ + TEMPO_CONTROL_REG(expth, R_EXPTH), /* 0x33 */ + TEMPO_CONTROL_REG(exprat, R_EXPRAT), /* 0x34 */ + TEMPO_CONTROL_REG(xatktcl, R_XATKTCL), /* 0x35 */ + TEMPO_CONTROL_REG(xatktch, R_XATKTCH), /* 0x36 */ + TEMPO_CONTROL_REG(xreltcl, R_XRELTCL), /* 0x37 */ + TEMPO_CONTROL_REG(xreltch, R_XRELTCH), /* 0x38 */ + TEMPO_CONTROL_REG(fxctl, R_FXCTL), /* 0x39 */ + TEMPO_CONTROL_REG(daccrwrl, R_DACCRWRL), /* 0x3A */ + TEMPO_CONTROL_REG(daccrwrm, R_DACCRWRM), /* 0x3B */ + TEMPO_CONTROL_REG(daccrwrh, R_DACCRWRH), /* 0x3C */ + TEMPO_CONTROL_REG(daccrrdl, R_DACCRRDL), /* 0x3D */ + TEMPO_CONTROL_REG(daccrrdm, R_DACCRRDM), /* 0x3E */ + TEMPO_CONTROL_REG(daccrrdh, R_DACCRRDH), /* 0x3F */ + TEMPO_CONTROL_REG(daccraddr, R_DACCRADDR), /* 0x40 */ + TEMPO_CONTROL_REG(dcofsel, R_DCOFSEL), /* 0x41 */ + TEMPO_CONTROL_REG(daccrstat, R_DACCRSTAT), /* 0x8A */ + TEMPO_CONTROL_REG(dacmbcen, R_DACMBCEN), /* 0xC7 */ + TEMPO_CONTROL_REG(dacmbcctl, R_DACMBCCTL), /* 0xC8 */ + TEMPO_CONTROL_REG(dacmbcmug1, R_DACMBCMUG1), /* 0xC9 */ + TEMPO_CONTROL_REG(dacmbcthr1, R_DACMBCTHR1), /* 0xCA */ + TEMPO_CONTROL_REG(dacmbcrat1, R_DACMBCRAT1), /* 0xCB */ + TEMPO_CONTROL_REG(dacmbcatk1l, R_DACMBCATK1L), /* 0xCC */ + TEMPO_CONTROL_REG(dacmbcatk1h, R_DACMBCATK1H), /* 0xCD */ + TEMPO_CONTROL_REG(dacmbcrel1l, R_DACMBCREL1L), /* 0xCE */ + TEMPO_CONTROL_REG(dacmbcrel1h, R_DACMBCREL1H), /* 0xCF */ + TEMPO_CONTROL_REG(dacmbcmug2, R_DACMBCMUG2), /* 0xD0 */ + TEMPO_CONTROL_REG(dacmbcthr2, R_DACMBCTHR2), /* 0xD1 */ + TEMPO_CONTROL_REG(dacmbcrat2, R_DACMBCRAT2), /* 0xD2 */ + TEMPO_CONTROL_REG(dacmbcatk2l, R_DACMBCATK2L), /* 0xD3 */ + TEMPO_CONTROL_REG(dacmbcatk2h, R_DACMBCATK2H), /* 0xD4 */ + TEMPO_CONTROL_REG(dacmbcrel2l, R_DACMBCREL2L), /* 0xD5 */ + TEMPO_CONTROL_REG(dacmbcrel2h, R_DACMBCREL2H), /* 0xD6 */ + TEMPO_CONTROL_REG(dacmbcmug3, R_DACMBCMUG3), /* 0xD7 */ + TEMPO_CONTROL_REG(dacmbcthr3, R_DACMBCTHR3), /* 0xD8 */ + TEMPO_CONTROL_REG(dacmbcrat3, R_DACMBCRAT3), /* 0xD9 */ + TEMPO_CONTROL_REG(dacmbcatk3l, R_DACMBCATK3L), /* 0xDA */ + TEMPO_CONTROL_REG(dacmbcatk3h, R_DACMBCATK3H), /* 0xDB */ + TEMPO_CONTROL_REG(dacmbcrel3l, R_DACMBCREL3L), /* 0xDC */ + TEMPO_CONTROL_REG(dacmbcrel3h, R_DACMBCREL3H), /* 0xDD */ +}; + +static ssize_t control_reg_export_store(struct kobject *kobj, + struct kobj_attribute *attr, + const char *buf, size_t count) +{ + int i; + int ret; + char *nl; + + /* Strip the newline */ + nl = strchr(buf, '\n'); + if (nl != NULL) + *nl = '\0'; + + for (i = 0; i < ARRAY_SIZE(control_regs); i++) { + if (strcmp(buf, control_regs[i].name) == 0) { + + if (control_regs[i].dir_kobj) + break; + + control_regs[i].dev = kobj_to_dev(kobj->parent); + if (!control_regs[i].dev) + return -ENODEV; + control_regs[i].dir_kobj = kobject_create_and_add( + control_regs[i].name, kobj); + if (!control_regs[i].dir_kobj) + return -ENOMEM; + ret = sysfs_create_file(control_regs[i].dir_kobj, + &control_regs[i].val_kobj_attr.attr); + if (ret < 0) + return ret; + ret = sysfs_create_file(control_regs[i].dir_kobj, + &control_regs[i].addr_kobj_attr.attr); + if (ret < 0) + return ret; + + return count; + } + } + + return count; +} + +static ssize_t control_reg_unexport_store(struct kobject *kobj, + struct kobj_attribute *attr, + const char *buf, size_t count) +{ + int i; + char *nl; + + /* Strip the newline */ + nl = strchr(buf, '\n'); + if (nl != NULL) + *nl = '\0'; + + for (i = 0; i < ARRAY_SIZE(control_regs); i++) { + if (strcmp(buf, control_regs[i].name) == 0) { + if (control_regs[i].dir_kobj) { + sysfs_remove_file(control_regs[i].dir_kobj, + &control_regs[i].val_kobj_attr.attr); + sysfs_remove_file(control_regs[i].dir_kobj, + &control_regs[i].addr_kobj_attr.attr); + kobject_put(control_regs[i].dir_kobj); + control_regs[i].dir_kobj = NULL; + return count; + } + } + } + + return count; +} + +static struct kobj_attribute control_reg_export = + __ATTR(export, 0664, NULL, control_reg_export_store); +static struct kobj_attribute control_reg_unexport = + __ATTR(unexport, 0664, NULL, control_reg_unexport_store); + +/* Control Interface */ + +struct tempo_control { + struct device *dev; + struct kobj_attribute kobj_attr; + uint8_t addr; + uint8_t mask; + uint8_t shift; +}; + +static ssize_t ctrl_show(struct kobject *kobj, struct kobj_attribute *attr, + char *buf) +{ + struct tempo_control *control = + container_of(attr, struct tempo_control, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control->dev); + unsigned int val, show; + int ret; + + ret = regmap_read(tscs42xx->regmap, + control->addr, + &val); + if (ret < 0) + return ret; + show = (val & control->mask) >> control->shift; + + return sprintf(buf, "0x%02x\n", show); +} + +static ssize_t ctrl_store(struct kobject *kobj, struct kobj_attribute *attr, + const char *buf, size_t count) +{ + struct tempo_control *control = + container_of(attr, struct tempo_control, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(control->dev); + unsigned int store; + int ret; + + ret = kstrtoint(buf, 0, &store); + if (ret < 0) + return ret; + + store = store << control->shift; + ret = regmap_update_bits(tscs42xx->regmap, + control->addr, control->mask, store); + if (ret < 0) + return ret; + + return count; +} + +#define TEMPO_CONTROL(_name, _addr, _mask, _shift) \ + { \ + NULL, \ + __ATTR(_name, FMODE, ctrl_show, ctrl_store), \ + .addr = _addr, \ + .mask = _mask, \ + .shift = _shift, \ + } + +/* Coefficient Interface */ + +struct tempo_coefficient { + struct device *dev; + struct kobj_attribute kobj_attr; + uint8_t addr; +}; + +static int enable_daccram_access(struct tscs42xx_priv *tscs42xx) +{ + struct snd_soc_codec *codec = tscs42xx->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + /* DAC needs to be powered */ + ret = snd_soc_dapm_force_enable_pin(dapm, "DAC L"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable DAC for DACCRAM access (%d)\n", ret); + return ret; + } + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync dapm context (%d)\n", ret); + return ret; + } + + /* If no one is using the PLL make sure there is a valid rate */ + if (tscs42xx->pll_users == 0) + tscs42xx->samplerate = 48000; + + return power_up_audio_plls(tscs42xx->codec, tscs42xx); +} + +static int disable_daccram_access(struct tscs42xx_priv *tscs42xx) +{ + struct snd_soc_codec *codec = tscs42xx->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + /* DAC needs to be powered */ + ret = snd_soc_dapm_disable_pin(dapm, "DAC L"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to disable DAC after DACCRAM access (%d)\n", + ret); + return ret; + } + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync dapm context (%d)\n", ret); + return ret; + } + + return power_down_audio_plls(tscs42xx->codec, tscs42xx); +} + +static ssize_t cff_show(struct kobject *kobj, struct kobj_attribute *attr, + char *buf) +{ + struct tempo_coefficient *coefficient = + container_of(attr, struct tempo_coefficient, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(coefficient->dev); + unsigned int val, show; + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = enable_daccram_access(tscs42xx); + if (ret < 0) + goto early_exit; + + ret = regmap_write(tscs42xx->regmap, + R_DACCRADDR, + coefficient->addr); + if (ret < 0) + goto exit; + ret = regmap_read(tscs42xx->regmap, + R_DACCRRDL, + &val); + if (ret < 0) + goto exit; + show = val; + ret = regmap_read(tscs42xx->regmap, + R_DACCRRDM, + &val); + if (ret < 0) + goto exit; + show |= (val << 8); + ret = regmap_read(tscs42xx->regmap, + R_DACCRRDH, + &val); + if (ret < 0) + goto exit; + show |= (val << 16); + + ret = sprintf(buf, "0x%06x\n", show); +exit: + disable_daccram_access(tscs42xx); + +early_exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static ssize_t cff_store(struct kobject *kobj, struct kobj_attribute *attr, + const char *buf, size_t count) +{ + struct tempo_coefficient *coefficient = + container_of(attr, struct tempo_coefficient, kobj_attr); + struct tscs42xx_priv *tscs42xx = dev_get_drvdata(coefficient->dev); + unsigned int val, store; + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = enable_daccram_access(tscs42xx); + if (ret < 0) + goto early_exit; + + ret = kstrtoint(buf, 0, &store); + if (ret < 0) + goto exit; + + ret = regmap_write(tscs42xx->regmap, + R_DACCRADDR, + coefficient->addr); + if (ret < 0) + goto exit; + val = store & 0xff; + ret = regmap_write(tscs42xx->regmap, + R_DACCRWRL, + val); + if (ret < 0) + goto exit; + val = (store >> 8) & 0xff; + ret = regmap_write(tscs42xx->regmap, + R_DACCRWRM, + val); + if (ret < 0) + goto exit; + val = (store >> 16) & 0xff; + ret = regmap_write(tscs42xx->regmap, + R_DACCRWRH, + val); + if (ret < 0) + goto exit; + + ret = count; +exit: + disable_daccram_access(tscs42xx); + +early_exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +#define TEMPO_COEFFICIENT(_name, _addr) \ + { \ + NULL, \ + __ATTR(_name, FMODE, cff_show, cff_store), \ + .addr = _addr, \ + } + +#define BQC_COUNT 5 + +struct tempo_biquad { + struct tempo_coefficient coefficients[BQC_COUNT]; +}; + +#define TEMPO_BIQUAD(name, addr) \ + { \ + .coefficients = { \ + TEMPO_COEFFICIENT(name ## _b0, ((addr) + 0)), \ + TEMPO_COEFFICIENT(name ## _b1, ((addr) + 1)), \ + TEMPO_COEFFICIENT(name ## _b2, ((addr) + 2)), \ + TEMPO_COEFFICIENT(name ## _a1, ((addr) + 3)), \ + TEMPO_COEFFICIENT(name ## _a2, ((addr) + 4)) \ + }, \ + } + +static struct tempo_biquad dsp_biquads[] = { + /* EQ1 */ + TEMPO_BIQUAD(eq1_ch0_band1, 0x00), + TEMPO_BIQUAD(eq1_ch0_band2, 0x05), + TEMPO_BIQUAD(eq1_ch0_band3, 0x0a), + TEMPO_BIQUAD(eq1_ch0_band4, 0x0f), + TEMPO_BIQUAD(eq1_ch0_band5, 0x14), + TEMPO_BIQUAD(eq1_ch0_band6, 0x19), + TEMPO_BIQUAD(eq1_ch1_band1, 0x20), + TEMPO_BIQUAD(eq1_ch1_band2, 0x25), + TEMPO_BIQUAD(eq1_ch1_band3, 0x2a), + TEMPO_BIQUAD(eq1_ch1_band4, 0x2f), + TEMPO_BIQUAD(eq1_ch1_band5, 0x34), + TEMPO_BIQUAD(eq1_ch1_band6, 0x39), + /* EQ2 */ + TEMPO_BIQUAD(eq2_ch0_band1, 0x40), + TEMPO_BIQUAD(eq2_ch0_band2, 0x45), + TEMPO_BIQUAD(eq2_ch0_band3, 0x4a), + TEMPO_BIQUAD(eq2_ch0_band4, 0x4f), + TEMPO_BIQUAD(eq2_ch0_band5, 0x54), + TEMPO_BIQUAD(eq2_ch0_band6, 0x59), + TEMPO_BIQUAD(eq2_ch1_band1, 0x60), + TEMPO_BIQUAD(eq2_ch1_band2, 0x65), + TEMPO_BIQUAD(eq2_ch1_band3, 0x6a), + TEMPO_BIQUAD(eq2_ch1_band4, 0x6f), + TEMPO_BIQUAD(eq2_ch1_band5, 0x74), + TEMPO_BIQUAD(eq2_ch1_band6, 0x79), + /* Bass */ + TEMPO_BIQUAD(bass_ext1, 0x80), + TEMPO_BIQUAD(bass_ext2, 0x85), + TEMPO_BIQUAD(bass_lmt, 0x8c), + TEMPO_BIQUAD(bass_cto, 0x91), + /* Treble */ + TEMPO_BIQUAD(treb_ext1, 0x97), + TEMPO_BIQUAD(treb_ext2, 0x9c), + TEMPO_BIQUAD(treb_lmt, 0xa3), + TEMPO_BIQUAD(treb_cto, 0xa8), + /* Multi Band Compressor */ + TEMPO_BIQUAD(mbc_1_bq1, 0xb0), + TEMPO_BIQUAD(mbc_1_bq2, 0xb5), + TEMPO_BIQUAD(mbc_2_bq1, 0xba), + TEMPO_BIQUAD(mbc_2_bq2, 0xbf), + TEMPO_BIQUAD(mbc_3_bq1, 0xc4), + TEMPO_BIQUAD(mbc_3_bq2, 0xc9), +}; + +static struct tempo_coefficient gen_coefficients[] = { + /* 3D */ + TEMPO_COEFFICIENT(3d_coef, 0xae), + TEMPO_COEFFICIENT(3d_mix, 0xaf), + /* EQ1 */ + TEMPO_COEFFICIENT(eq1_ch0_prescale, 0x1f), + TEMPO_COEFFICIENT(eq1_ch1_prescale, 0x3f), + /* EQ2 */ + TEMPO_COEFFICIENT(eq2_ch0_prescale, 0x5f), + TEMPO_COEFFICIENT(eq2_ch1_prescale, 0x7f), + /* Bass */ + TEMPO_COEFFICIENT(bass_nlf_m1, 0x8a), + TEMPO_COEFFICIENT(bass_nlf_m2, 0x8b), + TEMPO_COEFFICIENT(bass_mix, 0x96), + /* Treble */ + TEMPO_COEFFICIENT(treb_nlf_m1, 0xa1), + TEMPO_COEFFICIENT(treb_nlf_m2, 0xa2), + TEMPO_COEFFICIENT(treb_mix, 0xad), +}; + +static struct tempo_control controls[] = { + TEMPO_CONTROL(eq1_en, R_CONFIG1, + RM_CONFIG1_EQ1_EN, FB_CONFIG1_EQ1_EN), + + TEMPO_CONTROL(eq1_bands_en, R_CONFIG1, + RM_CONFIG1_EQ1_BE, FB_CONFIG1_EQ1_BE), + + TEMPO_CONTROL(eq2_en, R_CONFIG1, + RM_CONFIG1_EQ2_EN, FB_CONFIG1_EQ2_EN), + + TEMPO_CONTROL(eq2_bands_en, R_CONFIG1, + RM_CONFIG1_EQ2_BE, FB_CONFIG1_EQ2_BE), + + TEMPO_CONTROL(cle_level_mode, R_CLECTL, + RM_CLECTL_LVL_MODE, FB_CLECTL_LVL_MODE), + + TEMPO_CONTROL(cle_level_detect_window, R_CLECTL, + RM_CLECTL_WINDOWSEL, FB_CLECTL_WINDOWSEL), + + TEMPO_CONTROL(exp_en, R_CLECTL, RM_CLECTL_EXP_EN, FB_CLECTL_EXP_EN), + + TEMPO_CONTROL(limit_en, R_CLECTL, RM_CLECTL_LIMIT_EN, + FB_CLECTL_LIMIT_EN), + + TEMPO_CONTROL(comp_en, R_CLECTL, RM_CLECTL_COMP_EN, FB_CLECTL_COMP_EN), + + TEMPO_CONTROL(3d_en, R_FXCTL, RM_FXCTL_3DEN, FB_FXCTL_3DEN), + + TEMPO_CONTROL(treb_en, R_FXCTL, RM_FXCTL_TEEN, FB_FXCTL_TEEN), + + TEMPO_CONTROL(treb_nlf_bypass, R_FXCTL, RM_FXCTL_TNLFBYPASS, + FB_FXCTL_TNLFBYPASS), + + TEMPO_CONTROL(bass_en, R_FXCTL, RM_FXCTL_BEEN, FB_FXCTL_BEEN), + + TEMPO_CONTROL(bass_nlf_bypass, R_FXCTL, RM_FXCTL_BNLFBYPASS, + FB_FXCTL_BNLFBYPASS), +}; + +static int create_sysfs_interface(struct kobject *parent_kobj) +{ + struct kobject *dsp_kobj; + struct kobject *controls_kobj; + struct kobject *control_regs_dir_kobj; + int i, j; + int ret; + + dsp_kobj = kobject_create_and_add("dsp", parent_kobj); + if (!dsp_kobj) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(dsp_biquads); i++) { + for (j = 0; j < BQC_COUNT; j++) { + dsp_biquads[i].coefficients[j].dev = + kobj_to_dev(parent_kobj); + if (!dsp_biquads[i].coefficients[j].dev) + return -ENODEV; + ret = sysfs_create_file(dsp_kobj, + &dsp_biquads[i].coefficients[j].kobj_attr.attr); + if (ret < 0) + return ret; + } + } + + for (i = 0; i < ARRAY_SIZE(gen_coefficients); i++) { + gen_coefficients[i].dev = kobj_to_dev(parent_kobj); + if (!gen_coefficients[i].dev) + return -ENODEV; + ret = sysfs_create_file(dsp_kobj, + &gen_coefficients[i].kobj_attr.attr); + if (ret < 0) + return ret; + } + + controls_kobj = kobject_create_and_add("controls", + parent_kobj); + if (!controls_kobj) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(controls); i++) { + controls[i].dev = kobj_to_dev(parent_kobj); + if (!controls[i].dev) + return -ENODEV; + ret = sysfs_create_file(controls_kobj, + &controls[i].kobj_attr.attr); + if (ret < 0) + return ret; + } + + control_regs_dir_kobj = kobject_create_and_add("control_regs", + parent_kobj); + + ret = sysfs_create_file(control_regs_dir_kobj, + &control_reg_export.attr); + if (ret < 0) + return ret; + + ret = sysfs_create_file(control_regs_dir_kobj, + &control_reg_unexport.attr); + if (ret < 0) + return ret; + + return 0; +} + +static int tscs42xx_probe(struct snd_soc_codec *codec) +{ + struct tscs42xx_priv *tscs42xx = snd_soc_codec_get_drvdata(codec); + int i; + int ret; + + mutex_lock(&tscs42xx->lock); + + ret = create_sysfs_interface(&codec->dev->kobj); + if (ret < 0) { + dev_info(codec->dev, "Failed to create dsp interface (%d)\n", + ret); + goto exit; + } + + tscs42xx->codec = codec; + + ret = configure_clocks(codec, tscs42xx); + if (ret < 0) { + dev_err(codec->dev, "Failed to configure clocks (%d)\n", ret); + goto exit; + } + + for (i = 0; i < ARRAY_SIZE(r_inits); ++i) { + ret = snd_soc_write(codec, r_inits[i].reg, r_inits[i].def); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write codec defaults (%d)\n", ret); + goto exit; + } + } + + /* Power up an interface so the daccram can be accessed */ + ret = snd_soc_update_bits(codec, R_PWRM2, RM_PWRM2_HPL, + RV_PWRM2_HPL_ENABLE); + if (ret < 0) { + dev_err(codec->dev, + "Failed to power up interface (%d)\n", ret); + goto exit; + } + + /* PLLs also needed to be powered */ + tscs42xx->samplerate = 48000; /* No valid rate exist yet */ + ret = power_up_audio_plls(codec, tscs42xx); + if (ret < 0) { + goto exit; + snd_soc_update_bits(codec, R_PWRM2, + RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + } + + ret = load_dac_coefficient_ram(codec); + if (ret < 0) { + dev_info(codec->dev, "Failed to load DAC Coefficients (%d)\n", + ret); + power_down_audio_plls(codec, tscs42xx); + snd_soc_update_bits(codec, R_PWRM2, + RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + goto exit; + } + + ret = load_control_regs(codec); + if (ret < 0) { + dev_info(codec->dev, "Failed to load controls (%d)\n", + ret); + power_down_audio_plls(codec, tscs42xx); + snd_soc_update_bits(codec, R_PWRM2, + RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + goto exit; + } + + power_down_audio_plls(codec, tscs42xx); + snd_soc_update_bits(codec, R_PWRM2, RM_PWRM2_HPL, RV_PWRM2_HPL_DISABLE); + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int tscs42xx_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, + .remove = tscs42xx_remove, + .component_driver = { + .dapm_widgets = tscs42xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), + .dapm_routes = tscs42xx_intercon, + .num_dapm_routes = ARRAY_SIZE(tscs42xx_intercon), + .controls = tscs42xx_snd_controls, + .num_controls = ARRAY_SIZE(tscs42xx_snd_controls), + }, +}; + +static int tscs42xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tscs42xx_priv *tscs42xx; + int ret = 0; + + tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); + if (!tscs42xx) + return -ENOMEM; + + mutex_init(&tscs42xx->lock); + mutex_lock(&tscs42xx->lock); + + tscs42xx->pll_users = 0; + + ret = set_data_from_of(i2c, tscs42xx); + if (ret < 0) { + dev_err(&i2c->dev, "Error parsing device tree info (%d)", ret); + goto exit; + } + + ret = part_is_valid(i2c); + if (ret <= 0) { + dev_err(&i2c->dev, "No valid part (%d)\n", ret); + ret = -ENODEV; + goto exit; + } + + /* Reset device */ + ret = i2c_smbus_write_byte_data(i2c, R_RESET, + RV_RESET_ENABLE); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to reset device (%d)\n", ret); + goto exit; + } + + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); + if (IS_ERR(tscs42xx->regmap)) { + ret = PTR_ERR(tscs42xx->regmap); + dev_err(&i2c->dev, "Failed to allocat regmap (%d)\n", ret); + goto exit; + } + + i2c_set_clientdata(i2c, tscs42xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tscs42xx, + &tscs42xx_dai, 1); + if (ret) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + goto exit; + } + + ret = 0; +exit: + mutex_unlock(&tscs42xx->lock); + + return ret; +} + +static int tscs42xx_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id tscs42xx_i2c_id[] = { + { "tscs42xx", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tscs42xx_i2c_id); + +static const struct of_device_id tscs42xx_of_match[] = { + { .compatible = "tscs,tscs42xx", }, + { } +}; +MODULE_DEVICE_TABLE(of, tscs42xx_of_match); + +static struct i2c_driver tscs42xx_i2c_driver = { + .driver = { + .name = "tscs42xx", + .owner = THIS_MODULE, + .of_match_table = tscs42xx_of_match, + }, + .probe = tscs42xx_i2c_probe, + .remove = tscs42xx_i2c_remove, + .id_table = tscs42xx_i2c_id, +}; + +static int __init tscs42xx_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&tscs42xx_i2c_driver); + if (ret < 0) + pr_err("Failed to register TSCS42xx I2C driver (%d)\n", ret); + + return ret; +} +module_init(tscs42xx_modinit); + +static void __exit tscs42xx_exit(void) +{ + i2c_del_driver(&tscs42xx_i2c_driver); +} +module_exit(tscs42xx_exit); + +MODULE_AUTHOR("Tempo Semiconductor <steven.eckhoff.opensource(a)gmail.com"); +MODULE_DESCRIPTION("ASoC TSCS42xx driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h new file mode 100644 index 000000000000..e38ff8c7b6c9 --- /dev/null +++ b/sound/soc/codecs/tscs42xx.h @@ -0,0 +1,2693 @@ +/* + * tscs42xx.c -- TSCS42xx ALSA SoC Audio driver + * + * Copyright 2017 Tempo Semiconductor, Inc. + * + * Author: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __WOOKIE_H__ +#define __WOOKIE_H__ + +#define R_HPVOLL 0x0 +#define R_HPVOLR 0x1 +#define R_SPKVOLL 0x2 +#define R_SPKVOLR 0x3 +#define R_DACVOLL 0x4 +#define R_DACVOLR 0x5 +#define R_ADCVOLL 0x6 +#define R_ADCVOLR 0x7 +#define R_INVOLL 0x8 +#define R_INVOLR 0x9 +#define R_INMODE 0x0B +#define R_INSELL 0x0C +#define R_INSELR 0x0D +#define R_AIC1 0x13 +#define R_AIC2 0x14 +#define R_CNVRTR0 0x16 +#define R_ADCSR 0x17 +#define R_CNVRTR1 0x18 +#define R_DACSR 0x19 +#define R_PWRM1 0x1A +#define R_PWRM2 0x1B +#define R_CONFIG0 0x1F +#define R_CONFIG1 0x20 +#define R_DMICCTL 0x24 +#define R_CLECTL 0x25 +#define R_MUGAIN 0x26 +#define R_COMPTH 0x27 +#define R_CMPRAT 0x28 +#define R_CATKTCL 0x29 +#define R_CATKTCH 0x2A +#define R_CRELTCL 0x2B +#define R_CRELTCH 0x2C +#define R_LIMTH 0x2D +#define R_LIMTGT 0x2E +#define R_LATKTCL 0x2F +#define R_LATKTCH 0x30 +#define R_LRELTCL 0x31 +#define R_LRELTCH 0x32 +#define R_EXPTH 0x33 +#define R_EXPRAT 0x34 +#define R_XATKTCL 0x35 +#define R_XATKTCH 0x36 +#define R_XRELTCL 0x37 +#define R_XRELTCH 0x38 +#define R_FXCTL 0x39 +#define R_DACCRWRL 0x3A +#define R_DACCRWRM 0x3B +#define R_DACCRWRH 0x3C +#define R_DACCRRDL 0x3D +#define R_DACCRRDM 0x3E +#define R_DACCRRDH 0x3F +#define R_DACCRADDR 0x40 +#define R_DCOFSEL 0x41 +#define R_PLLCTL9 0x4E +#define R_PLLCTLA 0x4F +#define R_PLLCTLB 0x50 +#define R_PLLCTLC 0x51 +#define R_PLLCTLD 0x52 +#define R_PLLCTLE 0x53 +#define R_PLLCTLF 0x54 +#define R_PLLCTL10 0x55 +#define R_PLLCTL11 0x56 +#define R_PLLCTL12 0x57 +#define R_PLLCTL1B 0x60 +#define R_PLLCTL1C 0x61 +#define R_TIMEBASE 0x77 +#define R_DEVIDL 0x7D +#define R_DEVIDH 0x7E +#define R_RESET 0x80 +#define R_DACCRSTAT 0x8A +#define R_PLLCTL0 0x8E +#define R_PLLREFSEL 0x8F +#define R_DACMBCEN 0xC7 +#define R_DACMBCCTL 0xC8 +#define R_DACMBCMUG1 0xC9 +#define R_DACMBCTHR1 0xCA +#define R_DACMBCRAT1 0xCB +#define R_DACMBCATK1L 0xCC +#define R_DACMBCATK1H 0xCD +#define R_DACMBCREL1L 0xCE +#define R_DACMBCREL1H 0xCF +#define R_DACMBCMUG2 0xD0 +#define R_DACMBCTHR2 0xD1 +#define R_DACMBCRAT2 0xD2 +#define R_DACMBCATK2L 0xD3 +#define R_DACMBCATK2H 0xD4 +#define R_DACMBCREL2L 0xD5 +#define R_DACMBCREL2H 0xD6 +#define R_DACMBCMUG3 0xD7 +#define R_DACMBCTHR3 0xD8 +#define R_DACMBCRAT3 0xD9 +#define R_DACMBCATK3L 0xDA +#define R_DACMBCATK3H 0xDB +#define R_DACMBCREL3L 0xDC +#define R_DACMBCREL3H 0xDD + +/* Helpers */ +#define RM(m, b) ((m)<<(b)) +#define RV(v, b) ((v)<<(b)) + +/**************************** + * R_HPVOLL (0x0) * + ****************************/ + +/* Field Offsets */ +#define FB_HPVOLL 0 + +/* Field Masks */ +#define FM_HPVOLL 0X7F + +/* Field Values */ +#define FV_HPVOLL_P6DB 0x7F +#define FV_HPVOLL_N88PT5DB 0x1 +#define FV_HPVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_HPVOLL RM(FM_HPVOLL, FB_HPVOLL) + +/* Register Values */ +#define RV_HPVOLL_P6DB RV(FV_HPVOLL_P6DB, FB_HPVOLL) +#define RV_HPVOLL_N88PT5DB RV(FV_HPVOLL_N88PT5DB, FB_HPVOLL) +#define RV_HPVOLL_MUTE RV(FV_HPVOLL_MUTE, FB_HPVOLL) + +/**************************** + * R_HPVOLR (0x1) * + ****************************/ + +/* Field Offsets */ +#define FB_HPVOLR 0 + +/* Field Masks */ +#define FM_HPVOLR 0X7F + +/* Field Values */ +#define FV_HPVOLR_P6DB 0x7F +#define FV_HPVOLR_N88PT5DB 0x1 +#define FV_HPVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_HPVOLR RM(FM_HPVOLR, FB_HPVOLR) + +/* Register Values */ +#define RV_HPVOLR_P6DB RV(FV_HPVOLR_P6DB, FB_HPVOLR) +#define RV_HPVOLR_N88PT5DB RV(FV_HPVOLR_N88PT5DB, FB_HPVOLR) +#define RV_HPVOLR_MUTE RV(FV_HPVOLR_MUTE, FB_HPVOLR) + +/***************************** + * R_SPKVOLL (0x2) * + *****************************/ + +/* Field Offsets */ +#define FB_SPKVOLL 0 + +/* Field Masks */ +#define FM_SPKVOLL 0X7F + +/* Field Values */ +#define FV_SPKVOLL_P12DB 0x7F +#define FV_SPKVOLL_N77PT25DB 0x8 +#define FV_SPKVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_SPKVOLL RM(FM_SPKVOLL, FB_SPKVOLL) + +/* Register Values */ +#define RV_SPKVOLL_P12DB RV(FV_SPKVOLL_P12DB, FB_SPKVOLL) +#define RV_SPKVOLL_N77PT25DB \ + RV(FV_SPKVOLL_N77PT25DB, FB_SPKVOLL) + +#define RV_SPKVOLL_MUTE RV(FV_SPKVOLL_MUTE, FB_SPKVOLL) + +/***************************** + * R_SPKVOLR (0x3) * + *****************************/ + +/* Field Offsets */ +#define FB_SPKVOLR 0 + +/* Field Masks */ +#define FM_SPKVOLR 0X7F + +/* Field Values */ +#define FV_SPKVOLR_P12DB 0x7F +#define FV_SPKVOLR_N77PT25DB 0x8 +#define FV_SPKVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_SPKVOLR RM(FM_SPKVOLR, FB_SPKVOLR) + +/* Register Values */ +#define RV_SPKVOLR_P12DB RV(FV_SPKVOLR_P12DB, FB_SPKVOLR) +#define RV_SPKVOLR_N77PT25DB \ + RV(FV_SPKVOLR_N77PT25DB, FB_SPKVOLR) + +#define RV_SPKVOLR_MUTE RV(FV_SPKVOLR_MUTE, FB_SPKVOLR) + +/***************************** + * R_DACVOLL (0x4) * + *****************************/ + +/* Field Offsets */ +#define FB_DACVOLL 0 + +/* Field Masks */ +#define FM_DACVOLL 0XFF + +/* Field Values */ +#define FV_DACVOLL_0DB 0xFF +#define FV_DACVOLL_N95PT625DB 0x1 +#define FV_DACVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_DACVOLL RM(FM_DACVOLL, FB_DACVOLL) + +/* Register Values */ +#define RV_DACVOLL_0DB RV(FV_DACVOLL_0DB, FB_DACVOLL) +#define RV_DACVOLL_N95PT625DB \ + RV(FV_DACVOLL_N95PT625DB, FB_DACVOLL) + +#define RV_DACVOLL_MUTE RV(FV_DACVOLL_MUTE, FB_DACVOLL) + +/***************************** + * R_DACVOLR (0x5) * + *****************************/ + +/* Field Offsets */ +#define FB_DACVOLR 0 + +/* Field Masks */ +#define FM_DACVOLR 0XFF + +/* Field Values */ +#define FV_DACVOLR_0DB 0xFF +#define FV_DACVOLR_N95PT625DB 0x1 +#define FV_DACVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_DACVOLR RM(FM_DACVOLR, FB_DACVOLR) + +/* Register Values */ +#define RV_DACVOLR_0DB RV(FV_DACVOLR_0DB, FB_DACVOLR) +#define RV_DACVOLR_N95PT625DB \ + RV(FV_DACVOLR_N95PT625DB, FB_DACVOLR) + +#define RV_DACVOLR_MUTE RV(FV_DACVOLR_MUTE, FB_DACVOLR) + +/***************************** + * R_ADCVOLL (0x6) * + *****************************/ + +/* Field Offsets */ +#define FB_ADCVOLL 0 + +/* Field Masks */ +#define FM_ADCVOLL 0XFF + +/* Field Values */ +#define FV_ADCVOLL_P24DB 0xFF +#define FV_ADCVOLL_N71PT25DB 0x1 +#define FV_ADCVOLL_MUTE 0x0 + +/* Register Masks */ +#define RM_ADCVOLL RM(FM_ADCVOLL, FB_ADCVOLL) + +/* Register Values */ +#define RV_ADCVOLL_P24DB RV(FV_ADCVOLL_P24DB, FB_ADCVOLL) +#define RV_ADCVOLL_N71PT25DB \ + RV(FV_ADCVOLL_N71PT25DB, FB_ADCVOLL) + +#define RV_ADCVOLL_MUTE RV(FV_ADCVOLL_MUTE, FB_ADCVOLL) + +/***************************** + * R_ADCVOLR (0x7) * + *****************************/ + +/* Field Offsets */ +#define FB_ADCVOLR 0 + +/* Field Masks */ +#define FM_ADCVOLR 0XFF + +/* Field Values */ +#define FV_ADCVOLR_P24DB 0xFF +#define FV_ADCVOLR_N71PT25DB 0x1 +#define FV_ADCVOLR_MUTE 0x0 + +/* Register Masks */ +#define RM_ADCVOLR RM(FM_ADCVOLR, FB_ADCVOLR) + +/* Register Values */ +#define RV_ADCVOLR_P24DB RV(FV_ADCVOLR_P24DB, FB_ADCVOLR) +#define RV_ADCVOLR_N71PT25DB \ + RV(FV_ADCVOLR_N71PT25DB, FB_ADCVOLR) + +#define RV_ADCVOLR_MUTE RV(FV_ADCVOLR_MUTE, FB_ADCVOLR) + +/**************************** + * R_INVOLL (0x8) * + ****************************/ + +/* Field Offsets */ +#define FB_INVOLL_INMUTEL 7 +#define FB_INVOLL_IZCL 6 +#define FB_INVOLL 0 + +/* Field Masks */ +#define FM_INVOLL_INMUTEL 0X1 +#define FM_INVOLL_IZCL 0X1 +#define FM_INVOLL 0X3F + +/* Field Values */ +#define FV_INVOLL_INMUTEL_ENABLE 0x1 +#define FV_INVOLL_INMUTEL_DISABLE 0x0 +#define FV_INVOLL_IZCL_ENABLE 0x1 +#define FV_INVOLL_IZCL_DISABLE 0x0 +#define FV_INVOLL_P30DB 0x3F +#define FV_INVOLL_N17PT25DB 0x0 + +/* Register Masks */ +#define RM_INVOLL_INMUTEL \ + RM(FM_INVOLL_INMUTEL, FB_INVOLL_INMUTEL) + +#define RM_INVOLL_IZCL RM(FM_INVOLL_IZCL, FB_INVOLL_IZCL) +#define RM_INVOLL RM(FM_INVOLL, FB_INVOLL) + +/* Register Values */ +#define RV_INVOLL_INMUTEL_ENABLE \ + RV(FV_INVOLL_INMUTEL_ENABLE, FB_INVOLL_INMUTEL) + +#define RV_INVOLL_INMUTEL_DISABLE \ + RV(FV_INVOLL_INMUTEL_DISABLE, FB_INVOLL_INMUTEL) + +#define RV_INVOLL_IZCL_ENABLE \ + RV(FV_INVOLL_IZCL_ENABLE, FB_INVOLL_IZCL) + +#define RV_INVOLL_IZCL_DISABLE \ + RV(FV_INVOLL_IZCL_DISABLE, FB_INVOLL_IZCL) + +#define RV_INVOLL_P30DB RV(FV_INVOLL_P30DB, FB_INVOLL) +#define RV_INVOLL_N17PT25DB RV(FV_INVOLL_N17PT25DB, FB_INVOLL) + +/**************************** + * R_INVOLR (0x9) * + ****************************/ + +/* Field Offsets */ +#define FB_INVOLR_INMUTER 7 +#define FB_INVOLR_IZCR 6 +#define FB_INVOLR 0 + +/* Field Masks */ +#define FM_INVOLR_INMUTER 0X1 +#define FM_INVOLR_IZCR 0X1 +#define FM_INVOLR 0X3F + +/* Field Values */ +#define FV_INVOLR_INMUTER_ENABLE 0x1 +#define FV_INVOLR_INMUTER_DISABLE 0x0 +#define FV_INVOLR_IZCR_ENABLE 0x1 +#define FV_INVOLR_IZCR_DISABLE 0x0 +#define FV_INVOLR_P30DB 0x3F +#define FV_INVOLR_N17PT25DB 0x0 + +/* Register Masks */ +#define RM_INVOLR_INMUTER \ + RM(FM_INVOLR_INMUTER, FB_INVOLR_INMUTER) + +#define RM_INVOLR_IZCR RM(FM_INVOLR_IZCR, FB_INVOLR_IZCR) +#define RM_INVOLR RM(FM_INVOLR, FB_INVOLR) + +/* Register Values */ +#define RV_INVOLR_INMUTER_ENABLE \ + RV(FV_INVOLR_INMUTER_ENABLE, FB_INVOLR_INMUTER) + +#define RV_INVOLR_INMUTER_DISABLE \ + RV(FV_INVOLR_INMUTER_DISABLE, FB_INVOLR_INMUTER) + +#define RV_INVOLR_IZCR_ENABLE \ + RV(FV_INVOLR_IZCR_ENABLE, FB_INVOLR_IZCR) + +#define RV_INVOLR_IZCR_DISABLE \ + RV(FV_INVOLR_IZCR_DISABLE, FB_INVOLR_IZCR) + +#define RV_INVOLR_P30DB RV(FV_INVOLR_P30DB, FB_INVOLR) +#define RV_INVOLR_N17PT25DB RV(FV_INVOLR_N17PT25DB, FB_INVOLR) + +/***************************** + * R_INMODE (0x0B) * + *****************************/ + +/* Field Offsets */ +#define FB_INMODE_DS 0 + +/* Field Masks */ +#define FM_INMODE_DS 0X1 + +/* Field Values */ +#define FV_INMODE_DS_LRIN1 0x0 +#define FV_INMODE_DS_LRIN2 0x1 + +/* Register Masks */ +#define RM_INMODE_DS RM(FM_INMODE_DS, FB_INMODE_DS) + +/* Register Values */ +#define RV_INMODE_DS_LRIN1 \ + RV(FV_INMODE_DS_LRIN1, FB_INMODE_DS) + +#define RV_INMODE_DS_LRIN2 \ + RV(FV_INMODE_DS_LRIN2, FB_INMODE_DS) + + +/***************************** + * R_INSELL (0x0C) * + *****************************/ + +/* Field Offsets */ +#define FB_INSELL 6 +#define FB_INSELL_MICBSTL 4 + +/* Field Masks */ +#define FM_INSELL 0X3 +#define FM_INSELL_MICBSTL 0X3 + +/* Field Values */ +#define FV_INSELL_IN1 0x0 +#define FV_INSELL_IN2 0x1 +#define FV_INSELL_IN3 0x2 +#define FV_INSELL_D2S 0x3 +#define FV_INSELL_MICBSTL_OFF 0x0 +#define FV_INSELL_MICBSTL_10DB 0x1 +#define FV_INSELL_MICBSTL_20DB 0x2 +#define FV_INSELL_MICBSTL_30DB 0x3 + +/* Register Masks */ +#define RM_INSELL RM(FM_INSELL, FB_INSELL) +#define RM_INSELL_MICBSTL \ + RM(FM_INSELL_MICBSTL, FB_INSELL_MICBSTL) + + +/* Register Values */ +#define RV_INSELL_IN1 RV(FV_INSELL_IN1, FB_INSELL) +#define RV_INSELL_IN2 RV(FV_INSELL_IN2, FB_INSELL) +#define RV_INSELL_IN3 RV(FV_INSELL_IN3, FB_INSELL) +#define RV_INSELL_D2S RV(FV_INSELL_D2S, FB_INSELL) +#define RV_INSELL_MICBSTL_OFF \ + RV(FV_INSELL_MICBSTL_OFF, FB_INSELL_MICBSTL) + +#define RV_INSELL_MICBSTL_10DB \ + RV(FV_INSELL_MICBSTL_10DB, FB_INSELL_MICBSTL) + +#define RV_INSELL_MICBSTL_20DB \ + RV(FV_INSELL_MICBSTL_20DB, FB_INSELL_MICBSTL) + +#define RV_INSELL_MICBSTL_30DB \ + RV(FV_INSELL_MICBSTL_30DB, FB_INSELL_MICBSTL) + + +/***************************** + * R_INSELR (0x0D) * + *****************************/ + +/* Field Offsets */ +#define FB_INSELR 6 +#define FB_INSELR_MICBSTR 4 + +/* Field Masks */ +#define FM_INSELR 0X3 +#define FM_INSELR_MICBSTR 0X3 + +/* Field Values */ +#define FV_INSELR_IN1 0x0 +#define FV_INSELR_IN2 0x1 +#define FV_INSELR_IN3 0x2 +#define FV_INSELR_D2S 0x3 +#define FV_INSELR_MICBSTR_OFF 0x0 +#define FV_INSELR_MICBSTR_10DB 0x1 +#define FV_INSELR_MICBSTR_20DB 0x2 +#define FV_INSELR_MICBSTR_30DB 0x3 + +/* Register Masks */ +#define RM_INSELR RM(FM_INSELR, FB_INSELR) +#define RM_INSELR_MICBSTR \ + RM(FM_INSELR_MICBSTR, FB_INSELR_MICBSTR) + + +/* Register Values */ +#define RV_INSELR_IN1 RV(FV_INSELR_IN1, FB_INSELR) +#define RV_INSELR_IN2 RV(FV_INSELR_IN2, FB_INSELR) +#define RV_INSELR_IN3 RV(FV_INSELR_IN3, FB_INSELR) +#define RV_INSELR_D2S RV(FV_INSELR_D2S, FB_INSELR) +#define RV_INSELR_MICBSTR_OFF \ + RV(FV_INSELR_MICBSTR_OFF, FB_INSELR_MICBSTR) + +#define RV_INSELR_MICBSTR_10DB \ + RV(FV_INSELR_MICBSTR_10DB, FB_INSELR_MICBSTR) + +#define RV_INSELR_MICBSTR_20DB \ + RV(FV_INSELR_MICBSTR_20DB, FB_INSELR_MICBSTR) + +#define RV_INSELR_MICBSTR_30DB \ + RV(FV_INSELR_MICBSTR_30DB, FB_INSELR_MICBSTR) + + +/*************************** + * R_AIC1 (0x13) * + ***************************/ + +/* Field Offsets */ +#define FB_AIC1_BCLKINV 6 +#define FB_AIC1_MS 5 +#define FB_AIC1_LRP 4 +#define FB_AIC1_WL 2 +#define FB_AIC1_FORMAT 0 + +/* Field Masks */ +#define FM_AIC1_BCLKINV 0X1 +#define FM_AIC1_MS 0X1 +#define FM_AIC1_LRP 0X1 +#define FM_AIC1_WL 0X3 +#define FM_AIC1_FORMAT 0X3 + +/* Field Values */ +#define FV_AIC1_BCLKINV_ENABLE 0x1 +#define FV_AIC1_BCLKINV_DISABLE 0x0 +#define FV_AIC1_MS_MASTER 0x1 +#define FV_AIC1_MS_SLAVE 0x0 +#define FV_AIC1_LRP_INVERT 0x1 +#define FV_AIC1_LRP_NORMAL 0x0 +#define FV_AIC1_WL_16 0x0 +#define FV_AIC1_WL_20 0x1 +#define FV_AIC1_WL_24 0x2 +#define FV_AIC1_WL_32 0x3 +#define FV_AIC1_FORMAT_RIGHT 0x0 +#define FV_AIC1_FORMAT_LEFT 0x1 +#define FV_AIC1_FORMAT_I2S 0x2 + +/* Register Masks */ +#define RM_AIC1_BCLKINV \ + RM(FM_AIC1_BCLKINV, FB_AIC1_BCLKINV) + +#define RM_AIC1_MS RM(FM_AIC1_MS, FB_AIC1_MS) +#define RM_AIC1_LRP RM(FM_AIC1_LRP, FB_AIC1_LRP) +#define RM_AIC1_WL RM(FM_AIC1_WL, FB_AIC1_WL) +#define RM_AIC1_FORMAT RM(FM_AIC1_FORMAT, FB_AIC1_FORMAT) + +/* Register Values */ +#define RV_AIC1_BCLKINV_ENABLE \ + RV(FV_AIC1_BCLKINV_ENABLE, FB_AIC1_BCLKINV) + +#define RV_AIC1_BCLKINV_DISABLE \ + RV(FV_AIC1_BCLKINV_DISABLE, FB_AIC1_BCLKINV) + +#define RV_AIC1_MS_MASTER RV(FV_AIC1_MS_MASTER, FB_AIC1_MS) +#define RV_AIC1_MS_SLAVE RV(FV_AIC1_MS_SLAVE, FB_AIC1_MS) +#define RV_AIC1_LRP_INVERT \ + RV(FV_AIC1_LRP_INVERT, FB_AIC1_LRP) + +#define RV_AIC1_LRP_NORMAL \ + RV(FV_AIC1_LRP_NORMAL, FB_AIC1_LRP) + +#define RV_AIC1_WL_16 RV(FV_AIC1_WL_16, FB_AIC1_WL) +#define RV_AIC1_WL_20 RV(FV_AIC1_WL_20, FB_AIC1_WL) +#define RV_AIC1_WL_24 RV(FV_AIC1_WL_24, FB_AIC1_WL) +#define RV_AIC1_WL_32 RV(FV_AIC1_WL_32, FB_AIC1_WL) +#define RV_AIC1_FORMAT_RIGHT \ + RV(FV_AIC1_FORMAT_RIGHT, FB_AIC1_FORMAT) + +#define RV_AIC1_FORMAT_LEFT \ + RV(FV_AIC1_FORMAT_LEFT, FB_AIC1_FORMAT) + +#define RV_AIC1_FORMAT_I2S \ + RV(FV_AIC1_FORMAT_I2S, FB_AIC1_FORMAT) + + +/*************************** + * R_AIC2 (0x14) * + ***************************/ + +/* Field Offsets */ +#define FB_AIC2_DACDSEL 6 +#define FB_AIC2_ADCDSEL 4 +#define FB_AIC2_TRI 3 +#define FB_AIC2_BLRCM 0 + +/* Field Masks */ +#define FM_AIC2_DACDSEL 0X3 +#define FM_AIC2_ADCDSEL 0X3 +#define FM_AIC2_TRI 0X1 +#define FM_AIC2_BLRCM 0X7 + +/* Field Values */ +#define FV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED 0x3 + +/* Register Masks */ +#define RM_AIC2_DACDSEL \ + RM(FM_AIC2_DACDSEL, FB_AIC2_DACDSEL) + +#define RM_AIC2_ADCDSEL \ + RM(FM_AIC2_ADCDSEL, FB_AIC2_ADCDSEL) + +#define RM_AIC2_TRI RM(FM_AIC2_TRI, FB_AIC2_TRI) +#define RM_AIC2_BLRCM RM(FM_AIC2_BLRCM, FB_AIC2_BLRCM) + +/* Register Values */ +#define RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED \ + RV(FV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED, FB_AIC2_BLRCM) + + +/****************************** + * R_CNVRTR0 (0x16) * + ******************************/ + +/* Field Offsets */ +#define FB_CNVRTR0_ADCPOLR 7 +#define FB_CNVRTR0_ADCPOLL 6 +#define FB_CNVRTR0_AMONOMIX 4 +#define FB_CNVRTR0_ADCMU 3 +#define FB_CNVRTR0_HPOR 2 +#define FB_CNVRTR0_ADCHPDR 1 +#define FB_CNVRTR0_ADCHPDL 0 + +/* Field Masks */ +#define FM_CNVRTR0_ADCPOLR 0X1 +#define FM_CNVRTR0_ADCPOLL 0X1 +#define FM_CNVRTR0_AMONOMIX 0X3 +#define FM_CNVRTR0_ADCMU 0X1 +#define FM_CNVRTR0_HPOR 0X1 +#define FM_CNVRTR0_ADCHPDR 0X1 +#define FM_CNVRTR0_ADCHPDL 0X1 + +/* Field Values */ +#define FV_CNVRTR0_ADCPOLR_INVERT 0x1 +#define FV_CNVRTR0_ADCPOLR_NORMAL 0x0 +#define FV_CNVRTR0_ADCPOLL_INVERT 0x1 +#define FV_CNVRTR0_ADCPOLL_NORMAL 0x0 +#define FV_CNVRTR0_ADCMU_ENABLE 0x1 +#define FV_CNVRTR0_ADCMU_DISABLE 0x0 +#define FV_CNVRTR0_ADCHPDR_ENABLE 0x1 +#define FV_CNVRTR0_ADCHPDR_DISABLE 0x0 +#define FV_CNVRTR0_ADCHPDL_ENABLE 0x1 +#define FV_CNVRTR0_ADCHPDL_DISABLE 0x0 + +/* Register Masks */ +#define RM_CNVRTR0_ADCPOLR \ + RM(FM_CNVRTR0_ADCPOLR, FB_CNVRTR0_ADCPOLR) + +#define RM_CNVRTR0_ADCPOLL \ + RM(FM_CNVRTR0_ADCPOLL, FB_CNVRTR0_ADCPOLL) + +#define RM_CNVRTR0_AMONOMIX \ + RM(FM_CNVRTR0_AMONOMIX, FB_CNVRTR0_AMONOMIX) + +#define RM_CNVRTR0_ADCMU \ + RM(FM_CNVRTR0_ADCMU, FB_CNVRTR0_ADCMU) + +#define RM_CNVRTR0_HPOR \ + RM(FM_CNVRTR0_HPOR, FB_CNVRTR0_HPOR) + +#define RM_CNVRTR0_ADCHPDR \ + RM(FM_CNVRTR0_ADCHPDR, FB_CNVRTR0_ADCHPDR) + +#define RM_CNVRTR0_ADCHPDL \ + RM(FM_CNVRTR0_ADCHPDL, FB_CNVRTR0_ADCHPDL) + + +/* Register Values */ +#define RV_CNVRTR0_ADCPOLR_INVERT \ + RV(FV_CNVRTR0_ADCPOLR_INVERT, FB_CNVRTR0_ADCPOLR) + +#define RV_CNVRTR0_ADCPOLR_NORMAL \ + RV(FV_CNVRTR0_ADCPOLR_NORMAL, FB_CNVRTR0_ADCPOLR) + +#define RV_CNVRTR0_ADCPOLL_INVERT \ + RV(FV_CNVRTR0_ADCPOLL_INVERT, FB_CNVRTR0_ADCPOLL) + +#define RV_CNVRTR0_ADCPOLL_NORMAL \ + RV(FV_CNVRTR0_ADCPOLL_NORMAL, FB_CNVRTR0_ADCPOLL) + +#define RV_CNVRTR0_ADCMU_ENABLE \ + RV(FV_CNVRTR0_ADCMU_ENABLE, FB_CNVRTR0_ADCMU) + +#define RV_CNVRTR0_ADCMU_DISABLE \ + RV(FV_CNVRTR0_ADCMU_DISABLE, FB_CNVRTR0_ADCMU) + +#define RV_CNVRTR0_ADCHPDR_ENABLE \ + RV(FV_CNVRTR0_ADCHPDR_ENABLE, FB_CNVRTR0_ADCHPDR) + +#define RV_CNVRTR0_ADCHPDR_DISABLE \ + RV(FV_CNVRTR0_ADCHPDR_DISABLE, FB_CNVRTR0_ADCHPDR) + +#define RV_CNVRTR0_ADCHPDL_ENABLE \ + RV(FV_CNVRTR0_ADCHPDL_ENABLE, FB_CNVRTR0_ADCHPDL) + +#define RV_CNVRTR0_ADCHPDL_DISABLE \ + RV(FV_CNVRTR0_ADCHPDL_DISABLE, FB_CNVRTR0_ADCHPDL) + + +/**************************** + * R_ADCSR (0x17) * + ****************************/ + +/* Field Offsets */ +#define FB_ADCSR_ABCM 6 +#define FB_ADCSR_ABR 3 +#define FB_ADCSR_ABM 0 + +/* Field Masks */ +#define FM_ADCSR_ABCM 0X3 +#define FM_ADCSR_ABR 0X3 +#define FM_ADCSR_ABM 0X7 + +/* Field Values */ +#define FV_ADCSR_ABCM_AUTO 0x0 +#define FV_ADCSR_ABCM_32 0x1 +#define FV_ADCSR_ABCM_40 0x2 +#define FV_ADCSR_ABCM_64 0x3 +#define FV_ADCSR_ABR_32 0x0 +#define FV_ADCSR_ABR_44_1 0x1 +#define FV_ADCSR_ABR_48 0x2 +#define FV_ADCSR_ABM_PT25 0x0 +#define FV_ADCSR_ABM_PT5 0x1 +#define FV_ADCSR_ABM_1 0x2 +#define FV_ADCSR_ABM_2 0x3 + +/* Register Masks */ +#define RM_ADCSR_ABCM RM(FM_ADCSR_ABCM, FB_ADCSR_ABCM) +#define RM_ADCSR_ABR RM(FM_ADCSR_ABR, FB_ADCSR_ABR) +#define RM_ADCSR_ABM RM(FM_ADCSR_ABM, FB_ADCSR_ABM) + +/* Register Values */ +#define RV_ADCSR_ABCM_AUTO \ + RV(FV_ADCSR_ABCM_AUTO, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABCM_32 \ + RV(FV_ADCSR_ABCM_32, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABCM_40 \ + RV(FV_ADCSR_ABCM_40, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABCM_64 \ + RV(FV_ADCSR_ABCM_64, FB_ADCSR_ABCM) + +#define RV_ADCSR_ABR_32 RV(FV_ADCSR_ABR_32, FB_ADCSR_ABR) +#define RV_ADCSR_ABR_44_1 \ + RV(FV_ADCSR_ABR_44_1, FB_ADCSR_ABR) + +#define RV_ADCSR_ABR_48 RV(FV_ADCSR_ABR_48, FB_ADCSR_ABR) +#define RV_ADCSR_ABR_ RV(FV_ADCSR_ABR_, FB_ADCSR_ABR) +#define RV_ADCSR_ABM_PT25 \ + RV(FV_ADCSR_ABM_PT25, FB_ADCSR_ABM) + +#define RV_ADCSR_ABM_PT5 RV(FV_ADCSR_ABM_PT5, FB_ADCSR_ABM) +#define RV_ADCSR_ABM_1 RV(FV_ADCSR_ABM_1, FB_ADCSR_ABM) +#define RV_ADCSR_ABM_2 RV(FV_ADCSR_ABM_2, FB_ADCSR_ABM) + +/****************************** + * R_CNVRTR1 (0x18) * + ******************************/ + +/* Field Offsets */ +#define FB_CNVRTR1_DACPOLR 7 +#define FB_CNVRTR1_DACPOLL 6 +#define FB_CNVRTR1_DMONOMIX 4 +#define FB_CNVRTR1_DACMU 3 +#define FB_CNVRTR1_DEEMPH 2 +#define FB_CNVRTR1_DACDITH 0 + +/* Field Masks */ +#define FM_CNVRTR1_DACPOLR 0X1 +#define FM_CNVRTR1_DACPOLL 0X1 +#define FM_CNVRTR1_DMONOMIX 0X3 +#define FM_CNVRTR1_DACMU 0X1 +#define FM_CNVRTR1_DEEMPH 0X1 +#define FM_CNVRTR1_DACDITH 0X3 + +/* Field Values */ +#define FV_CNVRTR1_DACPOLR_INVERT 0x1 +#define FV_CNVRTR1_DACPOLR_NORMAL 0x0 +#define FV_CNVRTR1_DACPOLL_INVERT 0x1 +#define FV_CNVRTR1_DACPOLL_NORMAL 0x0 +#define FV_CNVRTR1_DMONOMIX_ENABLE 0x1 +#define FV_CNVRTR1_DMONOMIX_DISABLE 0x0 +#define FV_CNVRTR1_DACMU_ENABLE 0x1 +#define FV_CNVRTR1_DACMU_DISABLE 0x0 + +/* Register Masks */ +#define RM_CNVRTR1_DACPOLR \ + RM(FM_CNVRTR1_DACPOLR, FB_CNVRTR1_DACPOLR) + +#define RM_CNVRTR1_DACPOLL \ + RM(FM_CNVRTR1_DACPOLL, FB_CNVRTR1_DACPOLL) + +#define RM_CNVRTR1_DMONOMIX \ + RM(FM_CNVRTR1_DMONOMIX, FB_CNVRTR1_DMONOMIX) + +#define RM_CNVRTR1_DACMU \ + RM(FM_CNVRTR1_DACMU, FB_CNVRTR1_DACMU) + +#define RM_CNVRTR1_DEEMPH \ + RM(FM_CNVRTR1_DEEMPH, FB_CNVRTR1_DEEMPH) + +#define RM_CNVRTR1_DACDITH \ + RM(FM_CNVRTR1_DACDITH, FB_CNVRTR1_DACDITH) + + +/* Register Values */ +#define RV_CNVRTR1_DACPOLR_INVERT \ + RV(FV_CNVRTR1_DACPOLR_INVERT, FB_CNVRTR1_DACPOLR) + +#define RV_CNVRTR1_DACPOLR_NORMAL \ + RV(FV_CNVRTR1_DACPOLR_NORMAL, FB_CNVRTR1_DACPOLR) + +#define RV_CNVRTR1_DACPOLL_INVERT \ + RV(FV_CNVRTR1_DACPOLL_INVERT, FB_CNVRTR1_DACPOLL) + +#define RV_CNVRTR1_DACPOLL_NORMAL \ + RV(FV_CNVRTR1_DACPOLL_NORMAL, FB_CNVRTR1_DACPOLL) + +#define RV_CNVRTR1_DMONOMIX_ENABLE \ + RV(FV_CNVRTR1_DMONOMIX_ENABLE, FB_CNVRTR1_DMONOMIX) + +#define RV_CNVRTR1_DMONOMIX_DISABLE \ + RV(FV_CNVRTR1_DMONOMIX_DISABLE, FB_CNVRTR1_DMONOMIX) + +#define RV_CNVRTR1_DACMU_ENABLE \ + RV(FV_CNVRTR1_DACMU_ENABLE, FB_CNVRTR1_DACMU) + +#define RV_CNVRTR1_DACMU_DISABLE \ + RV(FV_CNVRTR1_DACMU_DISABLE, FB_CNVRTR1_DACMU) + + +/**************************** + * R_DACSR (0x19) * + ****************************/ + +/* Field Offsets */ +#define FB_DACSR_DBCM 6 +#define FB_DACSR_DBR 3 +#define FB_DACSR_DBM 0 + +/* Field Masks */ +#define FM_DACSR_DBCM 0X3 +#define FM_DACSR_DBR 0X3 +#define FM_DACSR_DBM 0X7 + +/* Field Values */ +#define FV_DACSR_DBCM_AUTO 0x0 +#define FV_DACSR_DBCM_32 0x1 +#define FV_DACSR_DBCM_40 0x2 +#define FV_DACSR_DBCM_64 0x3 +#define FV_DACSR_DBR_32 0x0 +#define FV_DACSR_DBR_44_1 0x1 +#define FV_DACSR_DBR_48 0x2 +#define FV_DACSR_DBM_PT25 0x0 +#define FV_DACSR_DBM_PT5 0x1 +#define FV_DACSR_DBM_1 0x2 +#define FV_DACSR_DBM_2 0x3 + +/* Register Masks */ +#define RM_DACSR_DBCM RM(FM_DACSR_DBCM, FB_DACSR_DBCM) +#define RM_DACSR_DBR RM(FM_DACSR_DBR, FB_DACSR_DBR) +#define RM_DACSR_DBM RM(FM_DACSR_DBM, FB_DACSR_DBM) + +/* Register Values */ +#define RV_DACSR_DBCM_AUTO \ + RV(FV_DACSR_DBCM_AUTO, FB_DACSR_DBCM) + +#define RV_DACSR_DBCM_32 \ + RV(FV_DACSR_DBCM_32, FB_DACSR_DBCM) + +#define RV_DACSR_DBCM_40 \ + RV(FV_DACSR_DBCM_40, FB_DACSR_DBCM) + +#define RV_DACSR_DBCM_64 \ + RV(FV_DACSR_DBCM_64, FB_DACSR_DBCM) + +#define RV_DACSR_DBR_32 RV(FV_DACSR_DBR_32, FB_DACSR_DBR) +#define RV_DACSR_DBR_44_1 \ + RV(FV_DACSR_DBR_44_1, FB_DACSR_DBR) + +#define RV_DACSR_DBR_48 RV(FV_DACSR_DBR_48, FB_DACSR_DBR) +#define RV_DACSR_DBM_PT25 \ + RV(FV_DACSR_DBM_PT25, FB_DACSR_DBM) + +#define RV_DACSR_DBM_PT5 RV(FV_DACSR_DBM_PT5, FB_DACSR_DBM) +#define RV_DACSR_DBM_1 RV(FV_DACSR_DBM_1, FB_DACSR_DBM) +#define RV_DACSR_DBM_2 RV(FV_DACSR_DBM_2, FB_DACSR_DBM) + +/**************************** + * R_PWRM1 (0x1A) * + ****************************/ + +/* Field Offsets */ +#define FB_PWRM1_BSTL 7 +#define FB_PWRM1_BSTR 6 +#define FB_PWRM1_PGAL 5 +#define FB_PWRM1_PGAR 4 +#define FB_PWRM1_ADCL 3 +#define FB_PWRM1_ADCR 2 +#define FB_PWRM1_MICB 1 +#define FB_PWRM1_DIGENB 0 + +/* Field Masks */ +#define FM_PWRM1_BSTL 0X1 +#define FM_PWRM1_BSTR 0X1 +#define FM_PWRM1_PGAL 0X1 +#define FM_PWRM1_PGAR 0X1 +#define FM_PWRM1_ADCL 0X1 +#define FM_PWRM1_ADCR 0X1 +#define FM_PWRM1_MICB 0X1 +#define FM_PWRM1_DIGENB 0X1 + +/* Field Values */ +#define FV_PWRM1_BSTL_ENABLE 0x1 +#define FV_PWRM1_BSTL_DISABLE 0x0 +#define FV_PWRM1_BSTR_ENABLE 0x1 +#define FV_PWRM1_BSTR_DISABLE 0x0 +#define FV_PWRM1_PGAL_ENABLE 0x1 +#define FV_PWRM1_PGAL_DISABLE 0x0 +#define FV_PWRM1_PGAR_ENABLE 0x1 +#define FV_PWRM1_PGAR_DISABLE 0x0 +#define FV_PWRM1_ADCL_ENABLE 0x1 +#define FV_PWRM1_ADCL_DISABLE 0x0 +#define FV_PWRM1_ADCR_ENABLE 0x1 +#define FV_PWRM1_ADCR_DISABLE 0x0 +#define FV_PWRM1_MICB_ENABLE 0x1 +#define FV_PWRM1_MICB_DISABLE 0x0 +#define FV_PWRM1_DIGENB_DISABLE 0x1 +#define FV_PWRM1_DIGENB_ENABLE 0x0 + +/* Register Masks */ +#define RM_PWRM1_BSTL RM(FM_PWRM1_BSTL, FB_PWRM1_BSTL) +#define RM_PWRM1_BSTR RM(FM_PWRM1_BSTR, FB_PWRM1_BSTR) +#define RM_PWRM1_PGAL RM(FM_PWRM1_PGAL, FB_PWRM1_PGAL) +#define RM_PWRM1_PGAR RM(FM_PWRM1_PGAR, FB_PWRM1_PGAR) +#define RM_PWRM1_ADCL RM(FM_PWRM1_ADCL, FB_PWRM1_ADCL) +#define RM_PWRM1_ADCR RM(FM_PWRM1_ADCR, FB_PWRM1_ADCR) +#define RM_PWRM1_MICB RM(FM_PWRM1_MICB, FB_PWRM1_MICB) +#define RM_PWRM1_DIGENB \ + RM(FM_PWRM1_DIGENB, FB_PWRM1_DIGENB) + + +/* Register Values */ +#define RV_PWRM1_BSTL_ENABLE \ + RV(FV_PWRM1_BSTL_ENABLE, FB_PWRM1_BSTL) + +#define RV_PWRM1_BSTL_DISABLE \ + RV(FV_PWRM1_BSTL_DISABLE, FB_PWRM1_BSTL) + +#define RV_PWRM1_BSTR_ENABLE \ + RV(FV_PWRM1_BSTR_ENABLE, FB_PWRM1_BSTR) + +#define RV_PWRM1_BSTR_DISABLE \ + RV(FV_PWRM1_BSTR_DISABLE, FB_PWRM1_BSTR) + +#define RV_PWRM1_PGAL_ENABLE \ + RV(FV_PWRM1_PGAL_ENABLE, FB_PWRM1_PGAL) + +#define RV_PWRM1_PGAL_DISABLE \ + RV(FV_PWRM1_PGAL_DISABLE, FB_PWRM1_PGAL) + +#define RV_PWRM1_PGAR_ENABLE \ + RV(FV_PWRM1_PGAR_ENABLE, FB_PWRM1_PGAR) + +#define RV_PWRM1_PGAR_DISABLE \ + RV(FV_PWRM1_PGAR_DISABLE, FB_PWRM1_PGAR) + +#define RV_PWRM1_ADCL_ENABLE \ + RV(FV_PWRM1_ADCL_ENABLE, FB_PWRM1_ADCL) + +#define RV_PWRM1_ADCL_DISABLE \ + RV(FV_PWRM1_ADCL_DISABLE, FB_PWRM1_ADCL) + +#define RV_PWRM1_ADCR_ENABLE \ + RV(FV_PWRM1_ADCR_ENABLE, FB_PWRM1_ADCR) + +#define RV_PWRM1_ADCR_DISABLE \ + RV(FV_PWRM1_ADCR_DISABLE, FB_PWRM1_ADCR) + +#define RV_PWRM1_MICB_ENABLE \ + RV(FV_PWRM1_MICB_ENABLE, FB_PWRM1_MICB) + +#define RV_PWRM1_MICB_DISABLE \ + RV(FV_PWRM1_MICB_DISABLE, FB_PWRM1_MICB) + +#define RV_PWRM1_DIGENB_DISABLE \ + RV(FV_PWRM1_DIGENB_DISABLE, FB_PWRM1_DIGENB) + +#define RV_PWRM1_DIGENB_ENABLE \ + RV(FV_PWRM1_DIGENB_ENABLE, FB_PWRM1_DIGENB) + + +/**************************** + * R_PWRM2 (0x1B) * + ****************************/ + +/* Field Offsets */ +#define FB_PWRM2_D2S 7 +#define FB_PWRM2_HPL 6 +#define FB_PWRM2_HPR 5 +#define FB_PWRM2_SPKL 4 +#define FB_PWRM2_SPKR 3 +#define FB_PWRM2_INSELL 2 +#define FB_PWRM2_INSELR 1 +#define FB_PWRM2_VREF 0 + +/* Field Masks */ +#define FM_PWRM2_D2S 0X1 +#define FM_PWRM2_HPL 0X1 +#define FM_PWRM2_HPR 0X1 +#define FM_PWRM2_SPKL 0X1 +#define FM_PWRM2_SPKR 0X1 +#define FM_PWRM2_INSELL 0X1 +#define FM_PWRM2_INSELR 0X1 +#define FM_PWRM2_VREF 0X1 + +/* Field Values */ +#define FV_PWRM2_D2S_ENABLE 0x1 +#define FV_PWRM2_D2S_DISABLE 0x0 +#define FV_PWRM2_HPL_ENABLE 0x1 +#define FV_PWRM2_HPL_DISABLE 0x0 +#define FV_PWRM2_HPR_ENABLE 0x1 +#define FV_PWRM2_HPR_DISABLE 0x0 +#define FV_PWRM2_SPKL_ENABLE 0x1 +#define FV_PWRM2_SPKL_DISABLE 0x0 +#define FV_PWRM2_SPKR_ENABLE 0x1 +#define FV_PWRM2_SPKR_DISABLE 0x0 +#define FV_PWRM2_INSELL_ENABLE 0x1 +#define FV_PWRM2_INSELL_DISABLE 0x0 +#define FV_PWRM2_INSELR_ENABLE 0x1 +#define FV_PWRM2_INSELR_DISABLE 0x0 +#define FV_PWRM2_VREF_ENABLE 0x1 +#define FV_PWRM2_VREF_DISABLE 0x0 + +/* Register Masks */ +#define RM_PWRM2_D2S RM(FM_PWRM2_D2S, FB_PWRM2_D2S) +#define RM_PWRM2_HPL RM(FM_PWRM2_HPL, FB_PWRM2_HPL) +#define RM_PWRM2_HPR RM(FM_PWRM2_HPR, FB_PWRM2_HPR) +#define RM_PWRM2_SPKL RM(FM_PWRM2_SPKL, FB_PWRM2_SPKL) +#define RM_PWRM2_SPKR RM(FM_PWRM2_SPKR, FB_PWRM2_SPKR) +#define RM_PWRM2_INSELL \ + RM(FM_PWRM2_INSELL, FB_PWRM2_INSELL) + +#define RM_PWRM2_INSELR \ + RM(FM_PWRM2_INSELR, FB_PWRM2_INSELR) + +#define RM_PWRM2_VREF RM(FM_PWRM2_VREF, FB_PWRM2_VREF) + +/* Register Values */ +#define RV_PWRM2_D2S_ENABLE \ + RV(FV_PWRM2_D2S_ENABLE, FB_PWRM2_D2S) + +#define RV_PWRM2_D2S_DISABLE \ + RV(FV_PWRM2_D2S_DISABLE, FB_PWRM2_D2S) + +#define RV_PWRM2_HPL_ENABLE \ + RV(FV_PWRM2_HPL_ENABLE, FB_PWRM2_HPL) + +#define RV_PWRM2_HPL_DISABLE \ + RV(FV_PWRM2_HPL_DISABLE, FB_PWRM2_HPL) + +#define RV_PWRM2_HPR_ENABLE \ + RV(FV_PWRM2_HPR_ENABLE, FB_PWRM2_HPR) + +#define RV_PWRM2_HPR_DISABLE \ + RV(FV_PWRM2_HPR_DISABLE, FB_PWRM2_HPR) + +#define RV_PWRM2_SPKL_ENABLE \ + RV(FV_PWRM2_SPKL_ENABLE, FB_PWRM2_SPKL) + +#define RV_PWRM2_SPKL_DISABLE \ + RV(FV_PWRM2_SPKL_DISABLE, FB_PWRM2_SPKL) + +#define RV_PWRM2_SPKR_ENABLE \ + RV(FV_PWRM2_SPKR_ENABLE, FB_PWRM2_SPKR) + +#define RV_PWRM2_SPKR_DISABLE \ + RV(FV_PWRM2_SPKR_DISABLE, FB_PWRM2_SPKR) + +#define RV_PWRM2_INSELL_ENABLE \ + RV(FV_PWRM2_INSELL_ENABLE, FB_PWRM2_INSELL) + +#define RV_PWRM2_INSELL_DISABLE \ + RV(FV_PWRM2_INSELL_DISABLE, FB_PWRM2_INSELL) + +#define RV_PWRM2_INSELR_ENABLE \ + RV(FV_PWRM2_INSELR_ENABLE, FB_PWRM2_INSELR) + +#define RV_PWRM2_INSELR_DISABLE \ + RV(FV_PWRM2_INSELR_DISABLE, FB_PWRM2_INSELR) + +#define RV_PWRM2_VREF_ENABLE \ + RV(FV_PWRM2_VREF_ENABLE, FB_PWRM2_VREF) + +#define RV_PWRM2_VREF_DISABLE \ + RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF) + + +/****************************** + * R_CONFIG0 (0x1F) * + ******************************/ + +/* Field Offsets */ +#define FB_CONFIG0_ASDM 6 +#define FB_CONFIG0_DSDM 4 +#define FB_CONFIG0_DC_BYPASS 1 +#define FB_CONFIG0_SD_FORCE_ON 0 + +/* Field Masks */ +#define FM_CONFIG0_ASDM 0X3 +#define FM_CONFIG0_DSDM 0X3 +#define FM_CONFIG0_DC_BYPASS 0X1 +#define FM_CONFIG0_SD_FORCE_ON 0X1 + +/* Field Values */ +#define FV_CONFIG0_ASDM_HALF 0x1 +#define FV_CONFIG0_ASDM_FULL 0x2 +#define FV_CONFIG0_ASDM_AUTO 0x3 +#define FV_CONFIG0_DSDM_HALF 0x1 +#define FV_CONFIG0_DSDM_FULL 0x2 +#define FV_CONFIG0_DSDM_AUTO 0x3 +#define FV_CONFIG0_DC_BYPASS_ENABLE 0x1 +#define FV_CONFIG0_DC_BYPASS_DISABLE 0x0 +#define FV_CONFIG0_SD_FORCE_ON_ENABLE 0x1 +#define FV_CONFIG0_SD_FORCE_ON_DISABLE 0x0 + +/* Register Masks */ +#define RM_CONFIG0_ASDM \ + RM(FM_CONFIG0_ASDM, FB_CONFIG0_ASDM) + +#define RM_CONFIG0_DSDM \ + RM(FM_CONFIG0_DSDM, FB_CONFIG0_DSDM) + +#define RM_CONFIG0_DC_BYPASS \ + RM(FM_CONFIG0_DC_BYPASS, FB_CONFIG0_DC_BYPASS) + +#define RM_CONFIG0_SD_FORCE_ON \ + RM(FM_CONFIG0_SD_FORCE_ON, FB_CONFIG0_SD_FORCE_ON) + + +/* Register Values */ +#define RV_CONFIG0_ASDM_HALF \ + RV(FV_CONFIG0_ASDM_HALF, FB_CONFIG0_ASDM) + +#define RV_CONFIG0_ASDM_FULL \ + RV(FV_CONFIG0_ASDM_FULL, FB_CONFIG0_ASDM) + +#define RV_CONFIG0_ASDM_AUTO \ + RV(FV_CONFIG0_ASDM_AUTO, FB_CONFIG0_ASDM) + +#define RV_CONFIG0_DSDM_HALF \ + RV(FV_CONFIG0_DSDM_HALF, FB_CONFIG0_DSDM) + +#define RV_CONFIG0_DSDM_FULL \ + RV(FV_CONFIG0_DSDM_FULL, FB_CONFIG0_DSDM) + +#define RV_CONFIG0_DSDM_AUTO \ + RV(FV_CONFIG0_DSDM_AUTO, FB_CONFIG0_DSDM) + +#define RV_CONFIG0_DC_BYPASS_ENABLE \ + RV(FV_CONFIG0_DC_BYPASS_ENABLE, FB_CONFIG0_DC_BYPASS) + +#define RV_CONFIG0_DC_BYPASS_DISABLE \ + RV(FV_CONFIG0_DC_BYPASS_DISABLE, FB_CONFIG0_DC_BYPASS) + +#define RV_CONFIG0_SD_FORCE_ON_ENABLE \ + RV(FV_CONFIG0_SD_FORCE_ON_ENABLE, FB_CONFIG0_SD_FORCE_ON) + +#define RV_CONFIG0_SD_FORCE_ON_DISABLE \ + RV(FV_CONFIG0_SD_FORCE_ON_DISABLE, FB_CONFIG0_SD_FORCE_ON) + + +/****************************** + * R_CONFIG1 (0x20) * + ******************************/ + +/* Field Offsets */ +#define FB_CONFIG1_EQ2_EN 7 +#define FB_CONFIG1_EQ2_BE 4 +#define FB_CONFIG1_EQ1_EN 3 +#define FB_CONFIG1_EQ1_BE 0 + +/* Field Masks */ +#define FM_CONFIG1_EQ2_EN 0X1 +#define FM_CONFIG1_EQ2_BE 0X7 +#define FM_CONFIG1_EQ1_EN 0X1 +#define FM_CONFIG1_EQ1_BE 0X7 + +/* Field Values */ +#define FV_CONFIG1_EQ2_EN_ENABLE 0x1 +#define FV_CONFIG1_EQ2_EN_DISABLE 0x0 +#define FV_CONFIG1_EQ2_BE_PRE 0x0 +#define FV_CONFIG1_EQ2_BE_PRE_EQ_0 0x1 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_1 0x2 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_2 0x3 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_3 0x4 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_4 0x5 +#define FV_CONFIG1_EQ2_BE_PRE_EQ0_5 0x6 +#define FV_CONFIG1_EQ1_EN_ENABLE 0x1 +#define FV_CONFIG1_EQ1_EN_DISABLE 0x0 +#define FV_CONFIG1_EQ1_BE_PRE 0x0 +#define FV_CONFIG1_EQ1_BE_PRE_EQ_0 0x1 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_1 0x2 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_2 0x3 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_3 0x4 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_4 0x5 +#define FV_CONFIG1_EQ1_BE_PRE_EQ0_5 0x6 + +/* Register Masks */ +#define RM_CONFIG1_EQ2_EN \ + RM(FM_CONFIG1_EQ2_EN, FB_CONFIG1_EQ2_EN) + +#define RM_CONFIG1_EQ2_BE \ + RM(FM_CONFIG1_EQ2_BE, FB_CONFIG1_EQ2_BE) + +#define RM_CONFIG1_EQ1_EN \ + RM(FM_CONFIG1_EQ1_EN, FB_CONFIG1_EQ1_EN) + +#define RM_CONFIG1_EQ1_BE \ + RM(FM_CONFIG1_EQ1_BE, FB_CONFIG1_EQ1_BE) + + +/* Register Values */ +#define RV_CONFIG1_EQ2_EN_ENABLE \ + RV(FV_CONFIG1_EQ2_EN_ENABLE, FB_CONFIG1_EQ2_EN) + +#define RV_CONFIG1_EQ2_EN_DISABLE \ + RV(FV_CONFIG1_EQ2_EN_DISABLE, FB_CONFIG1_EQ2_EN) + +#define RV_CONFIG1_EQ2_BE_PRE \ + RV(FV_CONFIG1_EQ2_BE_PRE, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ_0 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ_0, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_1 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_1, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_2 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_2, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_3 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_3, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_4 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_4, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ2_BE_PRE_EQ0_5 \ + RV(FV_CONFIG1_EQ2_BE_PRE_EQ0_5, FB_CONFIG1_EQ2_BE) + +#define RV_CONFIG1_EQ1_EN_ENABLE \ + RV(FV_CONFIG1_EQ1_EN_ENABLE, FB_CONFIG1_EQ1_EN) + +#define RV_CONFIG1_EQ1_EN_DISABLE \ + RV(FV_CONFIG1_EQ1_EN_DISABLE, FB_CONFIG1_EQ1_EN) + +#define RV_CONFIG1_EQ1_BE_PRE \ + RV(FV_CONFIG1_EQ1_BE_PRE, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ_0 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ_0, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_1 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_1, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_2 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_2, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_3 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_3, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_4 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_4, FB_CONFIG1_EQ1_BE) + +#define RV_CONFIG1_EQ1_BE_PRE_EQ0_5 \ + RV(FV_CONFIG1_EQ1_BE_PRE_EQ0_5, FB_CONFIG1_EQ1_BE) + + +/****************************** + * R_DMICCTL (0x24) * + ******************************/ + +/* Field Offsets */ +#define FB_DMICCTL_DMICEN 7 +#define FB_DMICCTL_DMONO 4 +#define FB_DMICCTL_DMPHADJ 2 +#define FB_DMICCTL_DMRATE 0 + +/* Field Masks */ +#define FM_DMICCTL_DMICEN 0X1 +#define FM_DMICCTL_DMONO 0X1 +#define FM_DMICCTL_DMPHADJ 0X3 +#define FM_DMICCTL_DMRATE 0X3 + +/* Field Values */ +#define FV_DMICCTL_DMICEN_ENABLE 0x1 +#define FV_DMICCTL_DMICEN_DISABLE 0x0 +#define FV_DMICCTL_DMONO_STEREO 0x0 +#define FV_DMICCTL_DMONO_MONO 0x1 + +/* Register Masks */ +#define RM_DMICCTL_DMICEN \ + RM(FM_DMICCTL_DMICEN, FB_DMICCTL_DMICEN) + +#define RM_DMICCTL_DMONO \ + RM(FM_DMICCTL_DMONO, FB_DMICCTL_DMONO) + +#define RM_DMICCTL_DMPHADJ \ + RM(FM_DMICCTL_DMPHADJ, FB_DMICCTL_DMPHADJ) + +#define RM_DMICCTL_DMRATE \ + RM(FM_DMICCTL_DMRATE, FB_DMICCTL_DMRATE) + + +/* Register Values */ +#define RV_DMICCTL_DMICEN_ENABLE \ + RV(FV_DMICCTL_DMICEN_ENABLE, FB_DMICCTL_DMICEN) + +#define RV_DMICCTL_DMICEN_DISABLE \ + RV(FV_DMICCTL_DMICEN_DISABLE, FB_DMICCTL_DMICEN) + +#define RV_DMICCTL_DMONO_STEREO \ + RV(FV_DMICCTL_DMONO_STEREO, FB_DMICCTL_DMONO) + +#define RV_DMICCTL_DMONO_MONO \ + RV(FV_DMICCTL_DMONO_MONO, FB_DMICCTL_DMONO) + + +/***************************** + * R_CLECTL (0x25) * + *****************************/ + +/* Field Offsets */ +#define FB_CLECTL_LVL_MODE 4 +#define FB_CLECTL_WINDOWSEL 3 +#define FB_CLECTL_EXP_EN 2 +#define FB_CLECTL_LIMIT_EN 1 +#define FB_CLECTL_COMP_EN 0 + +/* Field Masks */ +#define FM_CLECTL_LVL_MODE 0X1 +#define FM_CLECTL_WINDOWSEL 0X1 +#define FM_CLECTL_EXP_EN 0X1 +#define FM_CLECTL_LIMIT_EN 0X1 +#define FM_CLECTL_COMP_EN 0X1 + +/* Field Values */ +#define FV_CLECTL_LVL_MODE_AVG 0x0 +#define FV_CLECTL_LVL_MODE_PEAK 0x1 +#define FV_CLECTL_WINDOWSEL_512 0x0 +#define FV_CLECTL_WINDOWSEL_64 0x1 +#define FV_CLECTL_EXP_EN_ENABLE 0x1 +#define FV_CLECTL_EXP_EN_DISABLE 0x0 +#define FV_CLECTL_LIMIT_EN_ENABLE 0x1 +#define FV_CLECTL_LIMIT_EN_DISABLE 0x0 +#define FV_CLECTL_COMP_EN_ENABLE 0x1 +#define FV_CLECTL_COMP_EN_DISABLE 0x0 + +/* Register Masks */ +#define RM_CLECTL_LVL_MODE \ + RM(FM_CLECTL_LVL_MODE, FB_CLECTL_LVL_MODE) + +#define RM_CLECTL_WINDOWSEL \ + RM(FM_CLECTL_WINDOWSEL, FB_CLECTL_WINDOWSEL) + +#define RM_CLECTL_EXP_EN \ + RM(FM_CLECTL_EXP_EN, FB_CLECTL_EXP_EN) + +#define RM_CLECTL_LIMIT_EN \ + RM(FM_CLECTL_LIMIT_EN, FB_CLECTL_LIMIT_EN) + +#define RM_CLECTL_COMP_EN \ + RM(FM_CLECTL_COMP_EN, FB_CLECTL_COMP_EN) + + +/* Register Values */ +#define RV_CLECTL_LVL_MODE_AVG \ + RV(FV_CLECTL_LVL_MODE_AVG, FB_CLECTL_LVL_MODE) + +#define RV_CLECTL_LVL_MODE_PEAK \ + RV(FV_CLECTL_LVL_MODE_PEAK, FB_CLECTL_LVL_MODE) + +#define RV_CLECTL_WINDOWSEL_512 \ + RV(FV_CLECTL_WINDOWSEL_512, FB_CLECTL_WINDOWSEL) + +#define RV_CLECTL_WINDOWSEL_64 \ + RV(FV_CLECTL_WINDOWSEL_64, FB_CLECTL_WINDOWSEL) + +#define RV_CLECTL_EXP_EN_ENABLE \ + RV(FV_CLECTL_EXP_EN_ENABLE, FB_CLECTL_EXP_EN) + +#define RV_CLECTL_EXP_EN_DISABLE \ + RV(FV_CLECTL_EXP_EN_DISABLE, FB_CLECTL_EXP_EN) + +#define RV_CLECTL_LIMIT_EN_ENABLE \ + RV(FV_CLECTL_LIMIT_EN_ENABLE, FB_CLECTL_LIMIT_EN) + +#define RV_CLECTL_LIMIT_EN_DISABLE \ + RV(FV_CLECTL_LIMIT_EN_DISABLE, FB_CLECTL_LIMIT_EN) + +#define RV_CLECTL_COMP_EN_ENABLE \ + RV(FV_CLECTL_COMP_EN_ENABLE, FB_CLECTL_COMP_EN) + +#define RV_CLECTL_COMP_EN_DISABLE \ + RV(FV_CLECTL_COMP_EN_DISABLE, FB_CLECTL_COMP_EN) + + +/***************************** + * R_MUGAIN (0x26) * + *****************************/ + +/* Field Offsets */ +#define FB_MUGAIN_CLEMUG 0 + +/* Field Masks */ +#define FM_MUGAIN_CLEMUG 0X1F + +/* Field Values */ +#define FV_MUGAIN_CLEMUG_46PT5DB 0x1F +#define FV_MUGAIN_CLEMUG_0DB 0x0 + +/* Register Masks */ +#define RM_MUGAIN_CLEMUG \ + RM(FM_MUGAIN_CLEMUG, FB_MUGAIN_CLEMUG) + + +/* Register Values */ +#define RV_MUGAIN_CLEMUG_46PT5DB \ + RV(FV_MUGAIN_CLEMUG_46PT5DB, FB_MUGAIN_CLEMUG) + +#define RV_MUGAIN_CLEMUG_0DB \ + RV(FV_MUGAIN_CLEMUG_0DB, FB_MUGAIN_CLEMUG) + + +/***************************** + * R_COMPTH (0x27) * + *****************************/ + +/* Field Offsets */ +#define FB_COMPTH 0 + +/* Field Masks */ +#define FM_COMPTH 0XFF + +/* Field Values */ +#define FV_COMPTH_0DB 0xFF +#define FV_COMPTH_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_COMPTH RM(FM_COMPTH, FB_COMPTH) + +/* Register Values */ +#define RV_COMPTH_0DB RV(FV_COMPTH_0DB, FB_COMPTH) +#define RV_COMPTH_N95PT625DB \ + RV(FV_COMPTH_N95PT625DB, FB_COMPTH) + + +/***************************** + * R_CMPRAT (0x28) * + *****************************/ + +/* Field Offsets */ +#define FB_CMPRAT 0 + +/* Field Masks */ +#define FM_CMPRAT 0X1F + +/* Register Masks */ +#define RM_CMPRAT RM(FM_CMPRAT, FB_CMPRAT) + +/****************************** + * R_CATKTCL (0x29) * + ******************************/ + +/* Field Offsets */ +#define FB_CATKTCL 0 + +/* Field Masks */ +#define FM_CATKTCL 0XFF + +/* Register Masks */ +#define RM_CATKTCL RM(FM_CATKTCL, FB_CATKTCL) + +/****************************** + * R_CATKTCH (0x2A) * + ******************************/ + +/* Field Offsets */ +#define FB_CATKTCH 0 + +/* Field Masks */ +#define FM_CATKTCH 0XFF + +/* Register Masks */ +#define RM_CATKTCH RM(FM_CATKTCH, FB_CATKTCH) + +/****************************** + * R_CRELTCL (0x2B) * + ******************************/ + +/* Field Offsets */ +#define FB_CRELTCL 0 + +/* Field Masks */ +#define FM_CRELTCL 0XFF + +/* Register Masks */ +#define RM_CRELTCL RM(FM_CRELTCL, FB_CRELTCL) + +/****************************** + * R_CRELTCH (0x2C) * + ******************************/ + +/* Field Offsets */ +#define FB_CRELTCH 0 + +/* Field Masks */ +#define FM_CRELTCH 0XFF + +/* Register Masks */ +#define RM_CRELTCH RM(FM_CRELTCH, FB_CRELTCH) + +/**************************** + * R_LIMTH (0x2D) * + ****************************/ + +/* Field Offsets */ +#define FB_LIMTH 0 + +/* Field Masks */ +#define FM_LIMTH 0XFF + +/* Field Values */ +#define FV_LIMTH_0DB 0xFF +#define FV_LIMTH_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_LIMTH RM(FM_LIMTH, FB_LIMTH) + +/* Register Values */ +#define RV_LIMTH_0DB RV(FV_LIMTH_0DB, FB_LIMTH) +#define RV_LIMTH_N95PT625DB RV(FV_LIMTH_N95PT625DB, FB_LIMTH) + +/***************************** + * R_LIMTGT (0x2E) * + *****************************/ + +/* Field Offsets */ +#define FB_LIMTGT 0 + +/* Field Masks */ +#define FM_LIMTGT 0XFF + +/* Field Values */ +#define FV_LIMTGT_0DB 0xFF +#define FV_LIMTGT_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_LIMTGT RM(FM_LIMTGT, FB_LIMTGT) + +/* Register Values */ +#define RV_LIMTGT_0DB RV(FV_LIMTGT_0DB, FB_LIMTGT) +#define RV_LIMTGT_N95PT625DB \ + RV(FV_LIMTGT_N95PT625DB, FB_LIMTGT) + + +/****************************** + * R_LATKTCL (0x2F) * + ******************************/ + +/* Field Offsets */ +#define FB_LATKTCL 0 + +/* Field Masks */ +#define FM_LATKTCL 0XFF + +/* Register Masks */ +#define RM_LATKTCL RM(FM_LATKTCL, FB_LATKTCL) + +/****************************** + * R_LATKTCH (0x30) * + ******************************/ + +/* Field Offsets */ +#define FB_LATKTCH 0 + +/* Field Masks */ +#define FM_LATKTCH 0XFF + +/* Register Masks */ +#define RM_LATKTCH RM(FM_LATKTCH, FB_LATKTCH) + +/****************************** + * R_LRELTCL (0x31) * + ******************************/ + +/* Field Offsets */ +#define FB_LRELTCL 0 + +/* Field Masks */ +#define FM_LRELTCL 0XFF + +/* Register Masks */ +#define RM_LRELTCL RM(FM_LRELTCL, FB_LRELTCL) + +/****************************** + * R_LRELTCH (0x32) * + ******************************/ + +/* Field Offsets */ +#define FB_LRELTCH 0 + +/* Field Masks */ +#define FM_LRELTCH 0XFF + +/* Register Masks */ +#define RM_LRELTCH RM(FM_LRELTCH, FB_LRELTCH) + +/**************************** + * R_EXPTH (0x33) * + ****************************/ + +/* Field Offsets */ +#define FB_EXPTH 0 + +/* Field Masks */ +#define FM_EXPTH 0XFF + +/* Field Values */ +#define FV_EXPTH_0DB 0xFF +#define FV_EXPTH_N95PT625DB 0x0 + +/* Register Masks */ +#define RM_EXPTH RM(FM_EXPTH, FB_EXPTH) + +/* Register Values */ +#define RV_EXPTH_0DB RV(FV_EXPTH_0DB, FB_EXPTH) +#define RV_EXPTH_N95PT625DB RV(FV_EXPTH_N95PT625DB, FB_EXPTH) + +/***************************** + * R_EXPRAT (0x34) * + *****************************/ + +/* Field Offsets */ +#define FB_EXPRAT 0 + +/* Field Masks */ +#define FM_EXPRAT 0X7 + +/* Register Masks */ +#define RM_EXPRAT RM(FM_EXPRAT, FB_EXPRAT) + +/****************************** + * R_XATKTCL (0x35) * + ******************************/ + +/* Field Offsets */ +#define FB_XATKTCL 0 + +/* Field Masks */ +#define FM_XATKTCL 0XFF + +/* Register Masks */ +#define RM_XATKTCL RM(FM_XATKTCL, FB_XATKTCL) + +/****************************** + * R_XATKTCH (0x36) * + ******************************/ + +/* Field Offsets */ +#define FB_XATKTCH 0 + +/* Field Masks */ +#define FM_XATKTCH 0XFF + +/* Register Masks */ +#define RM_XATKTCH RM(FM_XATKTCH, FB_XATKTCH) + +/****************************** + * R_XRELTCL (0x37) * + ******************************/ + +/* Field Offsets */ +#define FB_XRELTCL 0 + +/* Field Masks */ +#define FM_XRELTCL 0XFF + +/* Register Masks */ +#define RM_XRELTCL RM(FM_XRELTCL, FB_XRELTCL) + +/****************************** + * R_XRELTCH (0x38) * + ******************************/ + +/* Field Offsets */ +#define FB_XRELTCH 0 + +/* Field Masks */ +#define FM_XRELTCH 0XFF + +/* Register Masks */ +#define RM_XRELTCH RM(FM_XRELTCH, FB_XRELTCH) + +/**************************** + * R_FXCTL (0x39) * + ****************************/ + +/* Field Offsets */ +#define FB_FXCTL_3DEN 4 +#define FB_FXCTL_TEEN 3 +#define FB_FXCTL_TNLFBYPASS 2 +#define FB_FXCTL_BEEN 1 +#define FB_FXCTL_BNLFBYPASS 0 + +/* Field Masks */ +#define FM_FXCTL_3DEN 0X1 +#define FM_FXCTL_TEEN 0X1 +#define FM_FXCTL_TNLFBYPASS 0X1 +#define FM_FXCTL_BEEN 0X1 +#define FM_FXCTL_BNLFBYPASS 0X1 + +/* Field Values */ +#define FV_FXCTL_3DEN_ENABLE 0x1 +#define FV_FXCTL_3DEN_DISABLE 0x0 +#define FV_FXCTL_TEEN_ENABLE 0x1 +#define FV_FXCTL_TEEN_DISABLE 0x0 +#define FV_FXCTL_TNLFBYPASS_ENABLE 0x1 +#define FV_FXCTL_TNLFBYPASS_DISABLE 0x0 +#define FV_FXCTL_BEEN_ENABLE 0x1 +#define FV_FXCTL_BEEN_DISABLE 0x0 +#define FV_FXCTL_BNLFBYPASS_ENABLE 0x1 +#define FV_FXCTL_BNLFBYPASS_DISABLE 0x0 + +/* Register Masks */ +#define RM_FXCTL_3DEN RM(FM_FXCTL_3DEN, FB_FXCTL_3DEN) +#define RM_FXCTL_TEEN RM(FM_FXCTL_TEEN, FB_FXCTL_TEEN) +#define RM_FXCTL_TNLFBYPASS \ + RM(FM_FXCTL_TNLFBYPASS, FB_FXCTL_TNLFBYPASS) + +#define RM_FXCTL_BEEN RM(FM_FXCTL_BEEN, FB_FXCTL_BEEN) +#define RM_FXCTL_BNLFBYPASS \ + RM(FM_FXCTL_BNLFBYPASS, FB_FXCTL_BNLFBYPASS) + + +/* Register Values */ +#define RV_FXCTL_3DEN_ENABLE \ + RV(FV_FXCTL_3DEN_ENABLE, FB_FXCTL_3DEN) + +#define RV_FXCTL_3DEN_DISABLE \ + RV(FV_FXCTL_3DEN_DISABLE, FB_FXCTL_3DEN) + +#define RV_FXCTL_TEEN_ENABLE \ + RV(FV_FXCTL_TEEN_ENABLE, FB_FXCTL_TEEN) + +#define RV_FXCTL_TEEN_DISABLE \ + RV(FV_FXCTL_TEEN_DISABLE, FB_FXCTL_TEEN) + +#define RV_FXCTL_TNLFBYPASS_ENABLE \ + RV(FV_FXCTL_TNLFBYPASS_ENABLE, FB_FXCTL_TNLFBYPASS) + +#define RV_FXCTL_TNLFBYPASS_DISABLE \ + RV(FV_FXCTL_TNLFBYPASS_DISABLE, FB_FXCTL_TNLFBYPASS) + +#define RV_FXCTL_BEEN_ENABLE \ + RV(FV_FXCTL_BEEN_ENABLE, FB_FXCTL_BEEN) + +#define RV_FXCTL_BEEN_DISABLE \ + RV(FV_FXCTL_BEEN_DISABLE, FB_FXCTL_BEEN) + +#define RV_FXCTL_BNLFBYPASS_ENABLE \ + RV(FV_FXCTL_BNLFBYPASS_ENABLE, FB_FXCTL_BNLFBYPASS) + +#define RV_FXCTL_BNLFBYPASS_DISABLE \ + RV(FV_FXCTL_BNLFBYPASS_DISABLE, FB_FXCTL_BNLFBYPASS) + + +/******************************* + * R_DACCRWRL (0x3A) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRWRL_DACCRWDL 0 + +/* Field Masks */ +#define FM_DACCRWRL_DACCRWDL 0XFF + +/* Register Masks */ +#define RM_DACCRWRL_DACCRWDL \ + RM(FM_DACCRWRL_DACCRWDL, FB_DACCRWRL_DACCRWDL) + + +/******************************* + * R_DACCRWRM (0x3B) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRWRM_DACCRWDM 0 + +/* Field Masks */ +#define FM_DACCRWRM_DACCRWDM 0XFF + +/* Register Masks */ +#define RM_DACCRWRM_DACCRWDM \ + RM(FM_DACCRWRM_DACCRWDM, FB_DACCRWRM_DACCRWDM) + + +/******************************* + * R_DACCRWRH (0x3C) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRWRH_DACCRWDH 0 + +/* Field Masks */ +#define FM_DACCRWRH_DACCRWDH 0XFF + +/* Register Masks */ +#define RM_DACCRWRH_DACCRWDH \ + RM(FM_DACCRWRH_DACCRWDH, FB_DACCRWRH_DACCRWDH) + + +/******************************* + * R_DACCRRDL (0x3D) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRRDL 0 + +/* Field Masks */ +#define FM_DACCRRDL 0XFF + +/* Register Masks */ +#define RM_DACCRRDL RM(FM_DACCRRDL, FB_DACCRRDL) + +/******************************* + * R_DACCRRDM (0x3E) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRRDM 0 + +/* Field Masks */ +#define FM_DACCRRDM 0XFF + +/* Register Masks */ +#define RM_DACCRRDM RM(FM_DACCRRDM, FB_DACCRRDM) + +/******************************* + * R_DACCRRDH (0x3F) * + *******************************/ + +/* Field Offsets */ +#define FB_DACCRRDH 0 + +/* Field Masks */ +#define FM_DACCRRDH 0XFF + +/* Register Masks */ +#define RM_DACCRRDH RM(FM_DACCRRDH, FB_DACCRRDH) + +/******************************** + * R_DACCRADDR (0x40) * + ********************************/ + +/* Field Offsets */ +#define FB_DACCRADDR_DACCRADD 0 + +/* Field Masks */ +#define FM_DACCRADDR_DACCRADD 0XFF + +/* Register Masks */ +#define RM_DACCRADDR_DACCRADD \ + RM(FM_DACCRADDR_DACCRADD, FB_DACCRADDR_DACCRADD) + + +/****************************** + * R_DCOFSEL (0x41) * + ******************************/ + +/* Field Offsets */ +#define FB_DCOFSEL_DC_COEF_SEL 0 + +/* Field Masks */ +#define FM_DCOFSEL_DC_COEF_SEL 0X7 + +/* Field Values */ +#define FV_DCOFSEL_DC_COEF_SEL_2_N8 0x0 +#define FV_DCOFSEL_DC_COEF_SEL_2_N9 0x1 +#define FV_DCOFSEL_DC_COEF_SEL_2_N10 0x2 +#define FV_DCOFSEL_DC_COEF_SEL_2_N11 0x3 +#define FV_DCOFSEL_DC_COEF_SEL_2_N12 0x4 +#define FV_DCOFSEL_DC_COEF_SEL_2_N13 0x5 +#define FV_DCOFSEL_DC_COEF_SEL_2_N14 0x6 +#define FV_DCOFSEL_DC_COEF_SEL_2_N15 0x7 + +/* Register Masks */ +#define RM_DCOFSEL_DC_COEF_SEL \ + RM(FM_DCOFSEL_DC_COEF_SEL, FB_DCOFSEL_DC_COEF_SEL) + + +/* Register Values */ +#define RV_DCOFSEL_DC_COEF_SEL_2_N8 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N8, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N9 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N9, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N10 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N10, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N11 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N11, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N12 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N12, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N13 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N13, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N14 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N14, FB_DCOFSEL_DC_COEF_SEL) + +#define RV_DCOFSEL_DC_COEF_SEL_2_N15 \ + RV(FV_DCOFSEL_DC_COEF_SEL_2_N15, FB_DCOFSEL_DC_COEF_SEL) + + +/****************************** + * R_PLLCTL9 (0x4E) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTL9_REFDIV_PLL1 0 + +/* Field Masks */ +#define FM_PLLCTL9_REFDIV_PLL1 0XFF + +/* Register Masks */ +#define RM_PLLCTL9_REFDIV_PLL1 \ + RM(FM_PLLCTL9_REFDIV_PLL1, FB_PLLCTL9_REFDIV_PLL1) + + +/****************************** + * R_PLLCTLA (0x4F) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLA_OUTDIV_PLL1 0 + +/* Field Masks */ +#define FM_PLLCTLA_OUTDIV_PLL1 0XFF + +/* Register Masks */ +#define RM_PLLCTLA_OUTDIV_PLL1 \ + RM(FM_PLLCTLA_OUTDIV_PLL1, FB_PLLCTLA_OUTDIV_PLL1) + + +/****************************** + * R_PLLCTLB (0x50) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLB_FBDIV_PLL1L 0 + +/* Field Masks */ +#define FM_PLLCTLB_FBDIV_PLL1L 0XFF + +/* Register Masks */ +#define RM_PLLCTLB_FBDIV_PLL1L \ + RM(FM_PLLCTLB_FBDIV_PLL1L, FB_PLLCTLB_FBDIV_PLL1L) + + +/****************************** + * R_PLLCTLC (0x51) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLC_FBDIV_PLL1H 0 + +/* Field Masks */ +#define FM_PLLCTLC_FBDIV_PLL1H 0X7 + +/* Register Masks */ +#define RM_PLLCTLC_FBDIV_PLL1H \ + RM(FM_PLLCTLC_FBDIV_PLL1H, FB_PLLCTLC_FBDIV_PLL1H) + + +/****************************** + * R_PLLCTLD (0x52) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLD_RZ_PLL1 3 +#define FB_PLLCTLD_CP_PLL1 0 + +/* Field Masks */ +#define FM_PLLCTLD_RZ_PLL1 0X7 +#define FM_PLLCTLD_CP_PLL1 0X7 + +/* Register Masks */ +#define RM_PLLCTLD_RZ_PLL1 \ + RM(FM_PLLCTLD_RZ_PLL1, FB_PLLCTLD_RZ_PLL1) + +#define RM_PLLCTLD_CP_PLL1 \ + RM(FM_PLLCTLD_CP_PLL1, FB_PLLCTLD_CP_PLL1) + + +/****************************** + * R_PLLCTLE (0x53) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLE_REFDIV_PLL2 0 + +/* Field Masks */ +#define FM_PLLCTLE_REFDIV_PLL2 0XFF + +/* Register Masks */ +#define RM_PLLCTLE_REFDIV_PLL2 \ + RM(FM_PLLCTLE_REFDIV_PLL2, FB_PLLCTLE_REFDIV_PLL2) + + +/****************************** + * R_PLLCTLF (0x54) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTLF_OUTDIV_PLL2 0 + +/* Field Masks */ +#define FM_PLLCTLF_OUTDIV_PLL2 0XFF + +/* Register Masks */ +#define RM_PLLCTLF_OUTDIV_PLL2 \ + RM(FM_PLLCTLF_OUTDIV_PLL2, FB_PLLCTLF_OUTDIV_PLL2) + + +/******************************* + * R_PLLCTL10 (0x55) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL10_FBDIV_PLL2L 0 + +/* Field Masks */ +#define FM_PLLCTL10_FBDIV_PLL2L 0XFF + +/* Register Masks */ +#define RM_PLLCTL10_FBDIV_PLL2L \ + RM(FM_PLLCTL10_FBDIV_PLL2L, FB_PLLCTL10_FBDIV_PLL2L) + + +/******************************* + * R_PLLCTL11 (0x56) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL11_FBDIV_PLL2H 0 + +/* Field Masks */ +#define FM_PLLCTL11_FBDIV_PLL2H 0X7 + +/* Register Masks */ +#define RM_PLLCTL11_FBDIV_PLL2H \ + RM(FM_PLLCTL11_FBDIV_PLL2H, FB_PLLCTL11_FBDIV_PLL2H) + + +/******************************* + * R_PLLCTL12 (0x57) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL12_RZ_PLL2 3 +#define FB_PLLCTL12_CP_PLL2 0 + +/* Field Masks */ +#define FM_PLLCTL12_RZ_PLL2 0X7 +#define FM_PLLCTL12_CP_PLL2 0X7 + +/* Register Masks */ +#define RM_PLLCTL12_RZ_PLL2 \ + RM(FM_PLLCTL12_RZ_PLL2, FB_PLLCTL12_RZ_PLL2) + +#define RM_PLLCTL12_CP_PLL2 \ + RM(FM_PLLCTL12_CP_PLL2, FB_PLLCTL12_CP_PLL2) + + +/******************************* + * R_PLLCTL1B (0x60) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL1B_VCOI_PLL2 4 +#define FB_PLLCTL1B_VCOI_PLL1 2 + +/* Field Masks */ +#define FM_PLLCTL1B_VCOI_PLL2 0X3 +#define FM_PLLCTL1B_VCOI_PLL1 0X3 + +/* Register Masks */ +#define RM_PLLCTL1B_VCOI_PLL2 \ + RM(FM_PLLCTL1B_VCOI_PLL2, FB_PLLCTL1B_VCOI_PLL2) + +#define RM_PLLCTL1B_VCOI_PLL1 \ + RM(FM_PLLCTL1B_VCOI_PLL1, FB_PLLCTL1B_VCOI_PLL1) + + +/******************************* + * R_PLLCTL1C (0x61) * + *******************************/ + +/* Field Offsets */ +#define FB_PLLCTL1C_PDB_PLL2 2 +#define FB_PLLCTL1C_PDB_PLL1 1 + +/* Field Masks */ +#define FM_PLLCTL1C_PDB_PLL2 0X1 +#define FM_PLLCTL1C_PDB_PLL1 0X1 + +/* Field Values */ +#define FV_PLLCTL1C_PDB_PLL2_ENABLE 0x1 +#define FV_PLLCTL1C_PDB_PLL2_DISABLE 0x0 +#define FV_PLLCTL1C_PDB_PLL1_ENABLE 0x1 +#define FV_PLLCTL1C_PDB_PLL1_DISABLE 0x0 + +/* Register Masks */ +#define RM_PLLCTL1C_PDB_PLL2 \ + RM(FM_PLLCTL1C_PDB_PLL2, FB_PLLCTL1C_PDB_PLL2) + +#define RM_PLLCTL1C_PDB_PLL1 \ + RM(FM_PLLCTL1C_PDB_PLL1, FB_PLLCTL1C_PDB_PLL1) + + +/* Register Values */ +#define RV_PLLCTL1C_PDB_PLL2_ENABLE \ + RV(FV_PLLCTL1C_PDB_PLL2_ENABLE, FB_PLLCTL1C_PDB_PLL2) + +#define RV_PLLCTL1C_PDB_PLL2_DISABLE \ + RV(FV_PLLCTL1C_PDB_PLL2_DISABLE, FB_PLLCTL1C_PDB_PLL2) + +#define RV_PLLCTL1C_PDB_PLL1_ENABLE \ + RV(FV_PLLCTL1C_PDB_PLL1_ENABLE, FB_PLLCTL1C_PDB_PLL1) + +#define RV_PLLCTL1C_PDB_PLL1_DISABLE \ + RV(FV_PLLCTL1C_PDB_PLL1_DISABLE, FB_PLLCTL1C_PDB_PLL1) + + +/******************************* + * R_TIMEBASE (0x77) * + *******************************/ + +/* Field Offsets */ +#define FB_TIMEBASE_DIVIDER 0 + +/* Field Masks */ +#define FM_TIMEBASE_DIVIDER 0XFF + +/* Register Masks */ +#define RM_TIMEBASE_DIVIDER \ + RM(FM_TIMEBASE_DIVIDER, FB_TIMEBASE_DIVIDER) + + +/***************************** + * R_DEVIDL (0x7D) * + *****************************/ + +/* Field Offsets */ +#define FB_DEVIDL_DIDL 0 + +/* Field Masks */ +#define FM_DEVIDL_DIDL 0XFF + +/* Register Masks */ +#define RM_DEVIDL_DIDL RM(FM_DEVIDL_DIDL, FB_DEVIDL_DIDL) + +/***************************** + * R_DEVIDH (0x7E) * + *****************************/ + +/* Field Offsets */ +#define FB_DEVIDH_DIDH 0 + +/* Field Masks */ +#define FM_DEVIDH_DIDH 0XFF + +/* Register Masks */ +#define RM_DEVIDH_DIDH RM(FM_DEVIDH_DIDH, FB_DEVIDH_DIDH) + +/**************************** + * R_RESET (0x80) * + ****************************/ + +/* Field Offsets */ +#define FB_RESET 0 + +/* Field Masks */ +#define FM_RESET 0XFF + +/* Field Values */ +#define FV_RESET_ENABLE 0x85 + +/* Register Masks */ +#define RM_RESET RM(FM_RESET, FB_RESET) + +/* Register Values */ +#define RV_RESET_ENABLE RV(FV_RESET_ENABLE, FB_RESET) + +/******************************** + * R_DACCRSTAT (0x8A) * + ********************************/ + +/* Field Offsets */ +#define FB_DACCRSTAT_DACCR_BUSY 7 + +/* Field Masks */ +#define FM_DACCRSTAT_DACCR_BUSY 0X1 + +/* Register Masks */ +#define RM_DACCRSTAT_DACCR_BUSY \ + RM(FM_DACCRSTAT_DACCR_BUSY, FB_DACCRSTAT_DACCR_BUSY) + + +/****************************** + * R_PLLCTL0 (0x8E) * + ******************************/ + +/* Field Offsets */ +#define FB_PLLCTL0_PLL2_LOCK 1 +#define FB_PLLCTL0_PLL1_LOCK 0 + +/* Field Masks */ +#define FM_PLLCTL0_PLL2_LOCK 0X1 +#define FM_PLLCTL0_PLL1_LOCK 0X1 + +/* Register Masks */ +#define RM_PLLCTL0_PLL2_LOCK \ + RM(FM_PLLCTL0_PLL2_LOCK, FB_PLLCTL0_PLL2_LOCK) + +#define RM_PLLCTL0_PLL1_LOCK \ + RM(FM_PLLCTL0_PLL1_LOCK, FB_PLLCTL0_PLL1_LOCK) + + +/******************************** + * R_PLLREFSEL (0x8F) * + ********************************/ + +/* Field Offsets */ +#define FB_PLLREFSEL_PLL2_REF_SEL 4 +#define FB_PLLREFSEL_PLL1_REF_SEL 0 + +/* Field Masks */ +#define FM_PLLREFSEL_PLL2_REF_SEL 0X7 +#define FM_PLLREFSEL_PLL1_REF_SEL 0X7 + +/* Field Values */ +#define FV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1 0x0 +#define FV_PLLREFSEL_PLL2_REF_SEL_MCLK2 0x1 +#define FV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1 0x0 +#define FV_PLLREFSEL_PLL1_REF_SEL_MCLK2 0x1 + +/* Register Masks */ +#define RM_PLLREFSEL_PLL2_REF_SEL \ + RM(FM_PLLREFSEL_PLL2_REF_SEL, FB_PLLREFSEL_PLL2_REF_SEL) + +#define RM_PLLREFSEL_PLL1_REF_SEL \ + RM(FM_PLLREFSEL_PLL1_REF_SEL, FB_PLLREFSEL_PLL1_REF_SEL) + + +/* Register Values */ +#define RV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1 \ + RV(FV_PLLREFSEL_PLL2_REF_SEL_XTAL_MCLK1, FB_PLLREFSEL_PLL2_REF_SEL) + +#define RV_PLLREFSEL_PLL2_REF_SEL_MCLK2 \ + RV(FV_PLLREFSEL_PLL2_REF_SEL_MCLK2, FB_PLLREFSEL_PLL2_REF_SEL) + +#define RV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1 \ + RV(FV_PLLREFSEL_PLL1_REF_SEL_XTAL_MCLK1, FB_PLLREFSEL_PLL1_REF_SEL) + +#define RV_PLLREFSEL_PLL1_REF_SEL_MCLK2 \ + RV(FV_PLLREFSEL_PLL1_REF_SEL_MCLK2, FB_PLLREFSEL_PLL1_REF_SEL) + + +/******************************* + * R_DACMBCEN (0xC7) * + *******************************/ + +/* Field Offsets */ +#define FB_DACMBCEN_MBCEN3 2 +#define FB_DACMBCEN_MBCEN2 1 +#define FB_DACMBCEN_MBCEN1 0 + +/* Field Masks */ +#define FM_DACMBCEN_MBCEN3 0X1 +#define FM_DACMBCEN_MBCEN2 0X1 +#define FM_DACMBCEN_MBCEN1 0X1 + +/* Register Masks */ +#define RM_DACMBCEN_MBCEN3 \ + RM(FM_DACMBCEN_MBCEN3, FB_DACMBCEN_MBCEN3) + +#define RM_DACMBCEN_MBCEN2 \ + RM(FM_DACMBCEN_MBCEN2, FB_DACMBCEN_MBCEN2) + +#define RM_DACMBCEN_MBCEN1 \ + RM(FM_DACMBCEN_MBCEN1, FB_DACMBCEN_MBCEN1) + + +/******************************** + * R_DACMBCCTL (0xC8) * + ********************************/ + +/* Field Offsets */ +#define FB_DACMBCCTL_LVLMODE3 5 +#define FB_DACMBCCTL_WINSEL3 4 +#define FB_DACMBCCTL_LVLMODE2 3 +#define FB_DACMBCCTL_WINSEL2 2 +#define FB_DACMBCCTL_LVLMODE1 1 +#define FB_DACMBCCTL_WINSEL1 0 + +/* Field Masks */ +#define FM_DACMBCCTL_LVLMODE3 0X1 +#define FM_DACMBCCTL_WINSEL3 0X1 +#define FM_DACMBCCTL_LVLMODE2 0X1 +#define FM_DACMBCCTL_WINSEL2 0X1 +#define FM_DACMBCCTL_LVLMODE1 0X1 +#define FM_DACMBCCTL_WINSEL1 0X1 + +/* Register Masks */ +#define RM_DACMBCCTL_LVLMODE3 \ + RM(FM_DACMBCCTL_LVLMODE3, FB_DACMBCCTL_LVLMODE3) + +#define RM_DACMBCCTL_WINSEL3 \ + RM(FM_DACMBCCTL_WINSEL3, FB_DACMBCCTL_WINSEL3) + +#define RM_DACMBCCTL_LVLMODE2 \ + RM(FM_DACMBCCTL_LVLMODE2, FB_DACMBCCTL_LVLMODE2) + +#define RM_DACMBCCTL_WINSEL2 \ + RM(FM_DACMBCCTL_WINSEL2, FB_DACMBCCTL_WINSEL2) + +#define RM_DACMBCCTL_LVLMODE1 \ + RM(FM_DACMBCCTL_LVLMODE1, FB_DACMBCCTL_LVLMODE1) + +#define RM_DACMBCCTL_WINSEL1 \ + RM(FM_DACMBCCTL_WINSEL1, FB_DACMBCCTL_WINSEL1) + + +/********************************* + * R_DACMBCMUG1 (0xC9) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCMUG1_PHASE 5 +#define FB_DACMBCMUG1_MUGAIN 0 + +/* Field Masks */ +#define FM_DACMBCMUG1_PHASE 0X1 +#define FM_DACMBCMUG1_MUGAIN 0X1F + +/* Register Masks */ +#define RM_DACMBCMUG1_PHASE \ + RM(FM_DACMBCMUG1_PHASE, FB_DACMBCMUG1_PHASE) + +#define RM_DACMBCMUG1_MUGAIN \ + RM(FM_DACMBCMUG1_MUGAIN, FB_DACMBCMUG1_MUGAIN) + + +/********************************* + * R_DACMBCTHR1 (0xCA) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCTHR1_THRESH 0 + +/* Field Masks */ +#define FM_DACMBCTHR1_THRESH 0XFF + +/* Register Masks */ +#define RM_DACMBCTHR1_THRESH \ + RM(FM_DACMBCTHR1_THRESH, FB_DACMBCTHR1_THRESH) + + +/********************************* + * R_DACMBCRAT1 (0xCB) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCRAT1_RATIO 0 + +/* Field Masks */ +#define FM_DACMBCRAT1_RATIO 0X1F + +/* Register Masks */ +#define RM_DACMBCRAT1_RATIO \ + RM(FM_DACMBCRAT1_RATIO, FB_DACMBCRAT1_RATIO) + + +/********************************** + * R_DACMBCATK1L (0xCC) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK1L_TCATKL 0 + +/* Field Masks */ +#define FM_DACMBCATK1L_TCATKL 0XFF + +/* Register Masks */ +#define RM_DACMBCATK1L_TCATKL \ + RM(FM_DACMBCATK1L_TCATKL, FB_DACMBCATK1L_TCATKL) + + +/********************************** + * R_DACMBCATK1H (0xCD) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK1H_TCATKH 0 + +/* Field Masks */ +#define FM_DACMBCATK1H_TCATKH 0XFF + +/* Register Masks */ +#define RM_DACMBCATK1H_TCATKH \ + RM(FM_DACMBCATK1H_TCATKH, FB_DACMBCATK1H_TCATKH) + + +/********************************** + * R_DACMBCREL1L (0xCE) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL1L_TCRELL 0 + +/* Field Masks */ +#define FM_DACMBCREL1L_TCRELL 0XFF + +/* Register Masks */ +#define RM_DACMBCREL1L_TCRELL \ + RM(FM_DACMBCREL1L_TCRELL, FB_DACMBCREL1L_TCRELL) + + +/********************************** + * R_DACMBCREL1H (0xCF) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL1H_TCRELH 0 + +/* Field Masks */ +#define FM_DACMBCREL1H_TCRELH 0XFF + +/* Register Masks */ +#define RM_DACMBCREL1H_TCRELH \ + RM(FM_DACMBCREL1H_TCRELH, FB_DACMBCREL1H_TCRELH) + + +/********************************* + * R_DACMBCMUG2 (0xD0) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCMUG2_PHASE 5 +#define FB_DACMBCMUG2_MUGAIN 0 + +/* Field Masks */ +#define FM_DACMBCMUG2_PHASE 0X1 +#define FM_DACMBCMUG2_MUGAIN 0X1F + +/* Register Masks */ +#define RM_DACMBCMUG2_PHASE \ + RM(FM_DACMBCMUG2_PHASE, FB_DACMBCMUG2_PHASE) + +#define RM_DACMBCMUG2_MUGAIN \ + RM(FM_DACMBCMUG2_MUGAIN, FB_DACMBCMUG2_MUGAIN) + + +/********************************* + * R_DACMBCTHR2 (0xD1) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCTHR2_THRESH 0 + +/* Field Masks */ +#define FM_DACMBCTHR2_THRESH 0XFF + +/* Register Masks */ +#define RM_DACMBCTHR2_THRESH \ + RM(FM_DACMBCTHR2_THRESH, FB_DACMBCTHR2_THRESH) + + +/********************************* + * R_DACMBCRAT2 (0xD2) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCRAT2_RATIO 0 + +/* Field Masks */ +#define FM_DACMBCRAT2_RATIO 0X1F + +/* Register Masks */ +#define RM_DACMBCRAT2_RATIO \ + RM(FM_DACMBCRAT2_RATIO, FB_DACMBCRAT2_RATIO) + + +/********************************** + * R_DACMBCATK2L (0xD3) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK2L_TCATKL 0 + +/* Field Masks */ +#define FM_DACMBCATK2L_TCATKL 0XFF + +/* Register Masks */ +#define RM_DACMBCATK2L_TCATKL \ + RM(FM_DACMBCATK2L_TCATKL, FB_DACMBCATK2L_TCATKL) + + +/********************************** + * R_DACMBCATK2H (0xD4) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK2H_TCATKH 0 + +/* Field Masks */ +#define FM_DACMBCATK2H_TCATKH 0XFF + +/* Register Masks */ +#define RM_DACMBCATK2H_TCATKH \ + RM(FM_DACMBCATK2H_TCATKH, FB_DACMBCATK2H_TCATKH) + + +/********************************** + * R_DACMBCREL2L (0xD5) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL2L_TCRELL 0 + +/* Field Masks */ +#define FM_DACMBCREL2L_TCRELL 0XFF + +/* Register Masks */ +#define RM_DACMBCREL2L_TCRELL \ + RM(FM_DACMBCREL2L_TCRELL, FB_DACMBCREL2L_TCRELL) + + +/********************************** + * R_DACMBCREL2H (0xD6) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL2H_TCRELH 0 + +/* Field Masks */ +#define FM_DACMBCREL2H_TCRELH 0XFF + +/* Register Masks */ +#define RM_DACMBCREL2H_TCRELH \ + RM(FM_DACMBCREL2H_TCRELH, FB_DACMBCREL2H_TCRELH) + + +/********************************* + * R_DACMBCMUG3 (0xD7) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCMUG3_PHASE 5 +#define FB_DACMBCMUG3_MUGAIN 0 + +/* Field Masks */ +#define FM_DACMBCMUG3_PHASE 0X1 +#define FM_DACMBCMUG3_MUGAIN 0X1F + +/* Register Masks */ +#define RM_DACMBCMUG3_PHASE \ + RM(FM_DACMBCMUG3_PHASE, FB_DACMBCMUG3_PHASE) + +#define RM_DACMBCMUG3_MUGAIN \ + RM(FM_DACMBCMUG3_MUGAIN, FB_DACMBCMUG3_MUGAIN) + + +/********************************* + * R_DACMBCTHR3 (0xD8) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCTHR3_THRESH 0 + +/* Field Masks */ +#define FM_DACMBCTHR3_THRESH 0XFF + +/* Register Masks */ +#define RM_DACMBCTHR3_THRESH \ + RM(FM_DACMBCTHR3_THRESH, FB_DACMBCTHR3_THRESH) + + +/********************************* + * R_DACMBCRAT3 (0xD9) * + *********************************/ + +/* Field Offsets */ +#define FB_DACMBCRAT3_RATIO 0 + +/* Field Masks */ +#define FM_DACMBCRAT3_RATIO 0X1F + +/* Register Masks */ +#define RM_DACMBCRAT3_RATIO \ + RM(FM_DACMBCRAT3_RATIO, FB_DACMBCRAT3_RATIO) + + +/********************************** + * R_DACMBCATK3L (0xDA) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK3L_TCATKL 0 + +/* Field Masks */ +#define FM_DACMBCATK3L_TCATKL 0XFF + +/* Register Masks */ +#define RM_DACMBCATK3L_TCATKL \ + RM(FM_DACMBCATK3L_TCATKL, FB_DACMBCATK3L_TCATKL) + + +/********************************** + * R_DACMBCATK3H (0xDB) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCATK3H_TCATKH 0 + +/* Field Masks */ +#define FM_DACMBCATK3H_TCATKH 0XFF + +/* Register Masks */ +#define RM_DACMBCATK3H_TCATKH \ + RM(FM_DACMBCATK3H_TCATKH, FB_DACMBCATK3H_TCATKH) + + +/********************************** + * R_DACMBCREL3L (0xDC) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL3L_TCRELL 0 + +/* Field Masks */ +#define FM_DACMBCREL3L_TCRELL 0XFF + +/* Register Masks */ +#define RM_DACMBCREL3L_TCRELL \ + RM(FM_DACMBCREL3L_TCRELL, FB_DACMBCREL3L_TCRELL) + + +/********************************** + * R_DACMBCREL3H (0xDD) * + **********************************/ + +/* Field Offsets */ +#define FB_DACMBCREL3H_TCRELH 0 + +/* Field Masks */ +#define FM_DACMBCREL3H_TCRELH 0XFF + +/* Register Masks */ +#define RM_DACMBCREL3H_TCRELH \ + RM(FM_DACMBCREL3H_TCRELH, FB_DACMBCREL3H_TCRELH) + + +#endif /* __WOOKIE_H__ */ -- 2.11.0

1 1

[alsa-devel] [PATCH 0/3] ALSA: ens1370: Fine-tuning for six function implementations
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 14 Nov 2017 18:38:28 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 15 function calls together with a variable assignment Use common error handling code in snd_audiopci_probe() Use common error handling code in snd_ensoniq_create() sound/pci/ens1370.c | 110 +++++++++++++++++++++++++++++----------------------- 1 file changed, 61 insertions(+), 49 deletions(-) -- 2.15.0

1 3

[alsa-devel] dsnoop, hwpointer and avail
by Henrik Eriksson 14 Nov '17

14 Nov '17

Hello Is there any documentation/rationale for how pcm_dsnoop works in snd_pcm_{dsnoop_}status? I am particularly wondering about the ordering of the snd_pcm_dsnoop_sync_ptr and the snd_pcm_status call on the slave pcm. I see occasional spikes (>1000 frames) of difference between the slave pcm hwpointer after the snd_pcm_status call on it and the pointers in the dsnoop pcm, and a correspondingly bad avail count. I suspect the mismatches are due to process scheduling (the slave pcm status is delayed because the application process not running and the hardware progresses during that delay). Does this seem plausible? If so, could the code be simplified to reduce the the number of systemcalls needed? The status of the slave pcm seems to provide much of the information used to sync the pointers. Or is there some other way to get a tighter coupling between the dsnoop status htstamp and avail count? Regards, /henrik

1 2

[alsa-devel] [PATCH] ALSA: emu10k1: Use common error handling code in two functions
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 14 Nov 2017 16:40:32 +0100 * Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix ten affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/emu10k1/emu10k1x.c | 62 +++++++++++++++++++++----------------------- sound/pci/emu10k1/emupcm.c | 31 +++++++++++++--------- 2 files changed, 49 insertions(+), 44 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2c2b12a06177..4f700907b566 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1569,38 +1569,33 @@ static int snd_emu10k1x_probe(struct pci_dev *pci, if (err < 0) return err; - if ((err = snd_emu10k1x_create(card, pci, &chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_emu10k1x_create(card, pci, &chip); + if (err < 0) + goto free_card; - if ((err = snd_emu10k1x_pcm(chip, 0)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_emu10k1x_pcm(chip, 1)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_emu10k1x_pcm(chip, 2)) < 0) { - snd_card_free(card); - return err; - } + err = snd_emu10k1x_pcm(chip, 0); + if (err < 0) + goto free_card; - if ((err = snd_emu10k1x_ac97(chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_emu10k1x_pcm(chip, 1); + if (err < 0) + goto free_card; - if ((err = snd_emu10k1x_mixer(chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_emu10k1x_pcm(chip, 2); + if (err < 0) + goto free_card; + + err = snd_emu10k1x_ac97(chip); + if (err < 0) + goto free_card; + + err = snd_emu10k1x_mixer(chip); + if (err < 0) + goto free_card; - if ((err = snd_emu10k1x_midi(chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_emu10k1x_midi(chip); + if (err < 0) + goto free_card; snd_emu10k1x_proc_init(chip); @@ -1609,14 +1604,17 @@ static int snd_emu10k1x_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->port, chip->irq); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto free_card; pci_set_drvdata(pci, card); dev++; return 0; + +free_card: + snd_card_free(card); + return err; } static void snd_emu10k1x_remove(struct pci_dev *pci) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 2683b9717215..088741f90f91 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1143,23 +1143,26 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) runtime->private_data = epcm; runtime->private_free = snd_emu10k1_pcm_free_substream; runtime->hw = snd_emu10k1_playback; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) { - kfree(epcm); - return err; - } - if ((err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX)) < 0) { - kfree(epcm); - return err; - } + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + goto free_pcm; + + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + 256, + UINT_MAX); + if (err < 0) + goto free_pcm; + if (emu->card_capabilities->emu_model && emu->emu1010.internal_clock == 0) sample_rate = 44100; else sample_rate = 48000; err = snd_pcm_hw_rule_noresample(runtime, sample_rate); - if (err < 0) { - kfree(epcm); - return err; - } + if (err < 0) + goto free_pcm; + mix = &emu->pcm_mixer[substream->number]; for (i = 0; i < 4; i++) mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i; @@ -1170,6 +1173,10 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) mix->epcm = epcm; snd_emu10k1_pcm_mixer_notify(emu, substream->number, 1); return 0; + +free_pcm: + kfree(epcm); + return err; } static int snd_emu10k1_playback_close(struct snd_pcm_substream *substream) -- 2.15.0

1 0

[alsa-devel] [PATCH 0/7] ALSA: Fine-tuning for several function implementations
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 24 Jan 2017 11:02:34 +0100 Some update suggestions were taken into account from static source code analysis. Markus Elfring (7): seq: Delete unnecessary checks in snd_seq_ioctl_query_subs() seq: Delete unnecessary checks in snd_seq_ioctl_get_subscription() seq: Delete unnecessary checks in snd_seq_ioctl_unsubscribe_port() seq: Delete unnecessary checks in snd_seq_ioctl_subscribe_port() seq: Adjust 16 function calls together with a variable assignment dmasound_core: Move two assignments for the variable "ret" in state_open() dmasound_core: Adjust six function calls together with a variable assignment sound/core/seq/seq_clientmgr.c | 163 ++++++++++++++++++++----------------- sound/oss/dmasound/dmasound_core.c | 33 +++++--- 2 files changed, 109 insertions(+), 87 deletions(-) -- 2.11.0

2 9

[alsa-devel] [PATCH 0/5] ASoC: cleanup random snd_soc_read/write - step [4/6]
by Kuninori Morimoto 14 Nov '17

14 Nov '17

Hi Mark Previous step [3/6] will be applied after merge window, thus, I try to post next step [4/6]. Current Codec framework is supporting its original reg_cache, but, we have "regmap" now, and common snd_soc_component_read/write are using it. This "legacy reg_cache" user driver is using its own xxx_read/write which is using reg_cache inside. If we can copy "Codec framework reg_cache" into its driver, we can remove original reg_cache from Codec framework. No problem, the user driver is very few. Let's do it (and remove original reg_cache in the future) (= for [1/5][2/5][3/5]). About [3/5] case, cx20442 driver is using reg_cache, but it is only for 1byte at this point. and this driver doesn't use snd_soc_write/read (which is using reg_cache). The reg_cache user is only debugfs. Let's just clean reg_cache for now, but keep cx20442_write. It seems Lars-Peter has some plan/patch for this driver, and it will update read/write related things which will use this cx20442_write. Our common snd_soc_read/write is wrapper of snd_soc_component_read/write now. snd_soc_component_read/write will try to use regmap first. And will try Codec framework's original .read/.write if driver doesn't use regmap. But, almost all driver which doesn't use regmap is using their original xxx_read/write instead of snd_soc_component_read/write. We can remove Codec framework's .read/.write if all these driver uses its own xxx_read/write instead of snd_soc_read/write (= for [4/5][5/5]). Kuninori Morimoto (5): ASoC: use internal reg_cache on uda1380 ASoC: use internal reg_cache on tlv320dac33 ASoC: cx20442: don't use reg_cache ASoC: don't use snd_soc_write/read on twl6040 ASoC: don't use snd_soc_write/read on twl4030 sound/soc/codecs/cx20442.c | 21 ++++----------------- sound/soc/codecs/tlv320dac33.c | 33 +++++++++++++-------------------- sound/soc/codecs/twl4030.c | 10 ++++------ sound/soc/codecs/twl6040.c | 4 +--- sound/soc/codecs/uda1380.c | 24 ++++++++++++++---------- 5 files changed, 36 insertions(+), 56 deletions(-) -- 1.9.1

1 5

[alsa-devel] [PATCH 0/9] Remove use of "gpio-reset" from DT
by Andrew F. Davis 13 Nov '17

13 Nov '17

Hello all, I was working on fixing up the tlv320aic31xx driver when I noticed its gpio reset was not working, it was due to this driver looking for "gpio-reset" instead of the usual "reset-gpio". A quick check shows this mistake is rare and only copied by one other audio CODECs: $ git grep "reset-gpio" | wc -l 630 $ git grep "gpio-reset" | wc -l 6 Luckliy the two effected drivers only use this reset line when power has been cut to the device, so not only were these optional properties but they had no real functional effect anyway. Lets just fix this typo before is spreads to drivers were it matters. I've also added fixes tags to each patch so it can be individually back-ported to where this bug was introduced if one wanted to. I'm hoping this can sit on next for a while to get the most testing, just in case I'm wrong about this not breaking anything :) Thanks, Andrew Andrew F. Davis (9): ASoC: tlv320aic31xx: Fix typo in DT binding documentation ASoC: tlv320aic3x: Fix typo in DT binding documentation ASoC: cs42l56: bindings: sound: Fix reset GPIO name in example DT binding ARM: dts: am335x-pepper: Fix the audio CODEC's reset pin ARM: dts: imx6: RDU2: Fix the audio CODEC's reset pin ARM: dts: imx: Fix the audio CODEC's reset pin ARM: dts: omap3-n900: Fix the audio CODEC's reset pin ASoC: tlv320aic31xx: Fix the reset GPIO OF name ASoC: tlv320aic3x: Fix the reset GPIO OF name Documentation/devicetree/bindings/sound/cs42l56.txt | 2 +- Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 5 ++++- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 6 +++++- arch/arm/boot/dts/am335x-pepper.dts | 2 +- arch/arm/boot/dts/imx6qdl-gw5903.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-zii-rdu2.dtsi | 4 ++-- arch/arm/boot/dts/omap3-n900.dts | 4 ++-- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- 9 files changed, 18 insertions(+), 11 deletions(-) -- 2.15.0

5 21

[alsa-devel] [PATCH] ALSA: echoaudio: Use common error handling code in two functions
by SF Markus Elfring 13 Nov '17

13 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 13 Nov 2017 21:57:57 +0100 * Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix six affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/echoaudio/echoaudio.c | 55 ++++++++++++++++++++++------------------- 1 file changed, 30 insertions(+), 25 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d68f99e076a8..ae58c48aaaf9 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1913,7 +1913,8 @@ static int snd_echo_create(struct snd_card *card, pci_write_config_byte(pci, PCI_LATENCY_TIMER, 0xC0); - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; pci_set_master(pci); @@ -1945,11 +1946,11 @@ static int snd_echo_create(struct snd_card *card, if (sz > PAGE_SIZE) sz = PAGE_SIZE; /* We map only the required part */ - if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz, - ECHOCARD_NAME)) == NULL) { + chip->iores = request_mem_region(chip->dsp_registers_phys, sz, + ECHOCARD_NAME); + if (!chip->iores) { dev_err(chip->card->dev, "cannot get memory region\n"); - snd_echo_free(chip); - return -EBUSY; + goto e_busy; } chip->dsp_registers = (volatile u32 __iomem *) ioremap_nocache(chip->dsp_registers_phys, sz); @@ -1957,8 +1958,7 @@ static int snd_echo_create(struct snd_card *card, if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(chip->card->dev, "cannot grab irq\n"); - snd_echo_free(chip); - return -EBUSY; + goto e_busy; } chip->irq = pci->irq; dev_dbg(card->dev, "pci=%p irq=%d subdev=%04x Init hardware...\n", @@ -1970,8 +1970,8 @@ static int snd_echo_create(struct snd_card *card, sizeof(struct comm_page), &chip->commpage_dma_buf) < 0) { dev_err(chip->card->dev, "cannot allocate the comm page\n"); - snd_echo_free(chip); - return -ENOMEM; + err = -ENOMEM; + goto free_sound_chip; } chip->comm_page_phys = chip->commpage_dma_buf.addr; chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; @@ -1981,17 +1981,22 @@ static int snd_echo_create(struct snd_card *card, err = set_mixer_defaults(chip); if (err < 0) { dev_err(card->dev, "init_hw err=%d\n", err); - snd_echo_free(chip); - return err; + goto free_sound_chip; } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_echo_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto free_sound_chip; + *rchip = chip; /* Init done ! */ return 0; + +e_busy: + err = -EBUSY; +free_sound_chip: + snd_echo_free(chip); + return err; } @@ -2020,10 +2025,9 @@ static int snd_echo_probe(struct pci_dev *pci, return err; chip = NULL; /* Tells snd_echo_create to allocate chip */ - if ((err = snd_echo_create(card, pci, &chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_echo_create(card, pci, &chip); + if (err < 0) + goto free_card; strcpy(card->driver, "Echo_" ECHOCARD_NAME); strcpy(card->shortname, chip->card_name); @@ -2036,18 +2040,18 @@ static int snd_echo_probe(struct pci_dev *pci, card->shortname, pci_id->subdevice & 0x000f, dsp, chip->dsp_registers_phys, chip->irq); - if ((err = snd_echo_new_pcm(chip)) < 0) { + err = snd_echo_new_pcm(chip); + if (err < 0) { dev_err(chip->card->dev, "new pcm error %d\n", err); - snd_card_free(card); - return err; + goto free_card; } #ifdef ECHOCARD_HAS_MIDI if (chip->has_midi) { /* Some Mia's do not have midi */ - if ((err = snd_echo_midi_create(card, chip)) < 0) { + err = snd_echo_midi_create(card, chip); + if (err < 0) { dev_err(chip->card->dev, "new midi error %d\n", err); - snd_card_free(card); - return err; + goto free_card; } } #endif @@ -2152,6 +2156,7 @@ static int snd_echo_probe(struct pci_dev *pci, ctl_error: dev_err(card->dev, "new control error %d\n", err); +free_card: snd_card_free(card); return err; } -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: cs5530: Use common error handling code in snd_cs5530_create()
by SF Markus Elfring 13 Nov '17

13 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 13 Nov 2017 20:33:07 +0100 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/cs5530.c | 48 ++++++++++++++++++++++++------------------------ 1 file changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 0a8cf94c4858..88f253489075 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -124,8 +124,8 @@ static int snd_cs5530_create(struct snd_card *card, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } chip->card = card; @@ -134,15 +134,14 @@ static int snd_cs5530_create(struct snd_card *card, err = pci_request_regions(pci, "CS5530"); if (err < 0) { kfree(chip); - pci_disable_device(pci); - return err; + goto disable_device; } chip->pci_base = pci_resource_start(pci, 0); mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { - snd_cs5530_free(chip); - return -EBUSY; + err = -EBUSY; + goto free_sound_chip; } map = readw(mem + 0x18); @@ -163,8 +162,7 @@ static int snd_cs5530_create(struct snd_card *card, dev_info(card->dev, "XpressAudio at 0x%lx\n", sb_base); else { dev_err(card->dev, "Could not find XpressAudio!\n"); - snd_cs5530_free(chip); - return -ENODEV; + goto e_nodev; } if (map & (1<<5)) @@ -183,8 +181,7 @@ static int snd_cs5530_create(struct snd_card *card, dma16 = 7; else { dev_err(card->dev, "No 16bit DMA enabled\n"); - snd_cs5530_free(chip); - return -ENODEV; + goto e_nodev; } if (dma8 & 0x01) @@ -195,8 +192,7 @@ static int snd_cs5530_create(struct snd_card *card, dma8 = 3; else { dev_err(card->dev, "No 8bit DMA enabled\n"); - snd_cs5530_free(chip); - return -ENODEV; + goto e_nodev; } if (irq & 1) @@ -209,8 +205,7 @@ static int snd_cs5530_create(struct snd_card *card, irq = 10; else { dev_err(card->dev, "SoundBlaster IRQ not set\n"); - snd_cs5530_free(chip); - return -ENODEV; + goto e_nodev; } dev_info(card->dev, "IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, dma16); @@ -219,32 +214,37 @@ static int snd_cs5530_create(struct snd_card *card, dma16, SB_HW_CS5530, &chip->sb); if (err < 0) { dev_err(card->dev, "Could not create SoundBlaster\n"); - snd_cs5530_free(chip); - return err; + goto free_sound_chip; } err = snd_sb16dsp_pcm(chip->sb, 0); if (err < 0) { dev_err(card->dev, "Could not create PCM\n"); - snd_cs5530_free(chip); - return err; + goto free_sound_chip; } err = snd_sbmixer_new(chip->sb); if (err < 0) { dev_err(card->dev, "Could not create Mixer\n"); - snd_cs5530_free(chip); - return err; + goto free_sound_chip; } err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_cs5530_free(chip); - return err; - } + if (err < 0) + goto free_sound_chip; *rchip = chip; return 0; + +disable_device: + pci_disable_device(pci); + return err; + +e_nodev: + err = -ENODEV; +free_sound_chip: + snd_cs5530_free(chip); + return err; } static int snd_cs5530_probe(struct pci_dev *pci, -- 2.15.0

1 0

[alsa-devel] [PATCH 0/4] ALSA: cs46xx: Adjustments for some function implementations
by SF Markus Elfring 13 Nov '17

13 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 13 Nov 2017 19:14:56 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (4): Adjust 33 function calls together with a variable assignment Use common error handling code in two functions Improve a size determination in cs46xx_dsp_proc_register_scb_desc() Adjust 35 checks for null pointers sound/pci/cs46xx/cs46xx.c | 75 ++++++++++---------- sound/pci/cs46xx/cs46xx_lib.c | 134 +++++++++++++++++++++--------------- sound/pci/cs46xx/dsp_spos.c | 60 +++++++++------- sound/pci/cs46xx/dsp_spos_scb_lib.c | 36 +++++----- 4 files changed, 166 insertions(+), 139 deletions(-) -- 2.15.0

2 5

[alsa-devel] [PATCH 0/1] USB Audio Device Class 3.0 support
by Ruslan Bilovol 13 Nov '17

13 Nov '17

Hi Takashi, This patch adds initial USB Audio Device Class 3.0 [1] support to the ALSA that we discussed at ELCE. The patch was tested with UAC3 gadget [2] that I posted to USB mailing list before. It is good for working with BADD device which implements such topologies like BAIF (Basic Audio Input Functions), BAOF (Basic Audio Output Functions) or compbination of both. UAC3 spec has changed descriptors laout and many other codes comparing to UAC2 spec, thus making reuse of existing sources impossible or quite complex. There are still many areas of improvement, as this patch doesn't have UAC3 Mixer Unit support nor some new features like Power Management I tested this with BeagleBone Black as UAC3 gadget device. [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] http://www.spinics.net/lists/linux-usb/msg162482.html Ruslan Bilovol (1): ALSA: usb: initial USB Audio Device Class 3.0 support include/linux/usb/audio-v2.h | 4 +- include/linux/usb/audio-v3.h | 343 +++++++++++++++++++++++++++++++++++++++++ include/uapi/linux/usb/audio.h | 1 + sound/usb/card.c | 7 +- sound/usb/card.h | 2 +- sound/usb/clock.c | 228 ++++++++++++++++++++++++--- sound/usb/clock.h | 4 +- sound/usb/format.c | 87 +++++++++-- sound/usb/format.h | 6 +- sound/usb/mixer.c | 332 ++++++++++++++++++++++++++------------- sound/usb/stream.c | 296 ++++++++++++++++++++++++++++++----- 11 files changed, 1117 insertions(+), 193 deletions(-) create mode 100644 include/linux/usb/audio-v3.h -- 1.9.1

5 13

[alsa-devel] [alsa-lib][PATCH v2 0/2] ctl: deprecate APIs of dimension information
by Takashi Sakamoto 13 Nov '17

13 Nov '17

Hi, This patchset is to update my previous one: [alsa-devel] [alsa-lib][PATCH 0/2] ctl: deprecate APIs of dimension information http://mailman.alsa-project.org/pipermail/alsa-devel/2017-November/126972.h… In ALSA control interface of asound.h, 'struct snd_ctl_elem_info' has 'dimen' member to deliver information for multi-dimensional array, however there's no common way to handle the member. As a result, drivers can force userspace applications to handle the information by inconsistent ways. A series of echoaudio drivers is an actual example of the inconsistent usage. This issue was addressed in a commit 51db452df07b ('Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"') to Linux kernel[1]. Fortunately, at present, this feature of interface is just used by the drivers, and there's a way to obsolete usage of the feature. We can obsolete it without large impacts to userland. As a result of discussion at Linux miniconference 2017, usage of 'dimen' member of 'struct snd_ctl_elem_info' is going to be deprecated for future removal. This patchset is for proposed tasks at a development period of Linux kernel v4.15[3]. Some APIs of dimension information are deprecated. In a short discussion in my previous patchset, it's clear that actual removal of kernel interface is uncertain depending on several conditions. Thus the timing to remove the library APIs is not decided yet. This updated version reflects the issue in API documentation (but simply removes statement of the schedule...). [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/s… [2] [alsa-devel] [ANNOUNCE] Audio Mini Summit 2017 at Prague, Oct 27 http://mailman.alsa-project.org/pipermail/alsa-devel/2017-July/123110.html [3] https://github.com/takaswie/presentations/blob/master/20171027/contents.md Takashi Sakamoto (2): ctl: deprecate APIs of dimensional information test: obsolete usage of APIs of dimensional information src/control/control.c | 9 +++++++++ test/user-ctl-element-set.c | 31 ------------------------------- 2 files changed, 9 insertions(+), 31 deletions(-) -- 2.14.1

2 4

[alsa-devel] [PATCH 0/3] ALSA: cs4281: Fine-tuning for five function implementations
by SF Markus Elfring 13 Nov '17

13 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 13 Nov 2017 13:46:23 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust 18 function calls together with a variable assignment Use common error handling code in snd_cs4281_probe() Use common error handling code in snd_cs4281_create() sound/pci/cs4281.c | 136 ++++++++++++++++++++++++++++++----------------------- 1 file changed, 77 insertions(+), 59 deletions(-) -- 2.15.0

1 3

[alsa-devel] Broken child-node lookup in sound/soc/mediatek/mt8173
by Johan Hovold 13 Nov '17

13 Nov '17

Hi, I'm trying to fix up incorrect usage of of_find_node_by_name() to lookup child nodes, and found the following code in sound/soc/mediatek/mt8173/mt8173-rt5650.c: static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { ... platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); ... if (of_find_node_by_name(platform_node, "codec-capture")) { np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); if (!np) { dev_err(&pdev->dev, "%s: Can't find codec-capture DT node\n", __func__); return -EINVAL; } ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); if (ret < 0) { dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); return ret; } mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; } added by commit d349caeb0510 ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver"). First of all the "codec-capture" node is indeed documented as a child node of the sound node, so the tree-wide depth-first search from the platform_node looks entirely bogus. Note that of_find_node_by_name() also drops a reference to its first argument, in this case the sound node, which could end up being prematurely freed. And then the reference to any returned codec-capture node (from either lookup) is never dropped. And since support for this second codec was added retrospectively and is documented as optional, that -EINVAL in case the node is missing looks broken too. I figured I better just report this one to the author of the patch and the maintainers to be straightened out. Thanks, Johan

1 0

[alsa-devel] [bug report] ASoC: Add EQ and filter to max98095 CODEC driver
by Dan Carpenter 13 Nov '17

13 Nov '17

[ This is an old patch. I'm working on some new Smatch stuff and I'm looking at this code, and I don't see how it's possible for this code to work. I must be missing something and it would help me if someone could explain to me what's going on so I can fix my software. - dan ] Hello Peter Hsiang, The patch dad31ec133ad: "ASoC: Add EQ and filter to max98095 CODEC driver" from Apr 19, 2011, leads to the following static checker warning: sound/soc/codecs/max98095.c:1524 max98095_put_eq_enum() warn: constraint overflow 'max98095->eq_texts' required = '(struct max98095_priv)->eq_textcnt' The error message is saying that Smatch expects we should verify that the index into the max98095->eq_texts[] array is less than max98095->eq_textcnt. sound/soc/codecs/max98095.c 1494 static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, 1495 struct snd_ctl_elem_value *ucontrol) 1496 { 1497 struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); 1498 struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); 1499 struct max98095_pdata *pdata = max98095->pdata; 1500 int channel = max98095_get_eq_channel(kcontrol->id.name); 1501 struct max98095_cdata *cdata; 1502 unsigned int sel = ucontrol->value.enumerated.item[0]; 1503 struct max98095_eq_cfg *coef_set; 1504 int fs, best, best_val, i; 1505 int regmask, regsave; 1506 1507 if (WARN_ON(channel > 1)) 1508 return -EINVAL; 1509 1510 if (!pdata || !max98095->eq_textcnt) ^^^^^^^^^^^^^^^^^^^^^ Here we check that max98095->eq_textcnt is non-zero. 1511 return 0; 1512 1513 if (sel >= pdata->eq_cfgcnt) ^^^^^^^^^^^^^^^^^^^^^^^ Check here 1514 return -EINVAL; 1515 1516 cdata = &max98095->dai[channel]; 1517 cdata->eq_sel = sel; 1518 fs = cdata->rate; 1519 1520 /* Find the selected configuration with nearest sample rate */ 1521 best = 0; 1522 best_val = INT_MAX; 1523 for (i = 0; i < pdata->eq_cfgcnt; i++) { ^^^^^^^^^^^^^^^^ According to Smatch the only place where pdata->eq_cfgcnt get set is from the assignment in max98095_i2c_probe() where we do: max98095->pdata = i2c->dev.platform_data; I don't know how i2c->dev.platform_data gets set. I would have thought it would just be a zeroed buffer. 1524 if (strcmp(pdata->eq_cfg[i].name, max98095->eq_texts[sel]) == 0 && ^^^^^^^^^^^^^ This is where the warning is generated. 1525 abs(pdata->eq_cfg[i].rate - fs) < best_val) { 1526 best = i; 1527 best_val = abs(pdata->eq_cfg[i].rate - fs); 1528 } 1529 } regards, dan carpenter

1 0

[alsa-devel] [PATCH] ALSA: aw2: Use common error handling code in three functions
by SF Markus Elfring 13 Nov '17

13 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 13 Nov 2017 09:49:15 +0100 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/aw2/aw2-alsa.c | 85 +++++++++++++++++++++++------------------------- 1 file changed, 40 insertions(+), 45 deletions(-) diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 9a49e4243a9c..25710043c79d 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -261,13 +261,13 @@ static int snd_aw2_create(struct snd_card *card, if ((dma_set_mask(&pci->dev, DMA_BIT_MASK(32)) < 0) || (dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(32)) < 0)) { dev_err(card->dev, "Impossible to set 32bit mask DMA\n"); - pci_disable_device(pci); - return -ENXIO; + err = -ENXIO; + goto disable_device; } chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } /* initialize the stuff */ @@ -277,11 +277,9 @@ static int snd_aw2_create(struct snd_card *card, /* (1) PCI resource allocation */ err = pci_request_regions(pci, "Audiowerk2"); - if (err < 0) { - pci_disable_device(pci); - kfree(chip); - return err; - } + if (err < 0) + goto free_data; + chip->iobase_phys = pci_resource_start(pci, 0); chip->iobase_virt = ioremap_nocache(chip->iobase_phys, @@ -289,10 +287,8 @@ static int snd_aw2_create(struct snd_card *card, if (chip->iobase_virt == NULL) { dev_err(card->dev, "unable to remap memory region"); - pci_release_regions(pci); - pci_disable_device(pci); - kfree(chip); - return -ENOMEM; + err = -ENOMEM; + goto release_regions; } /* (2) initialization of the chip hardware */ @@ -301,23 +297,15 @@ static int snd_aw2_create(struct snd_card *card, if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "Cannot grab irq %d\n", pci->irq); - - iounmap(chip->iobase_virt); - pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - kfree(chip); - return -EBUSY; + err = -EBUSY; + goto unmap_io; } chip->irq = pci->irq; err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { free_irq(chip->irq, (void *)chip); - iounmap(chip->iobase_virt); - pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - kfree(chip); - return err; + goto unmap_io; } *rchip = chip; @@ -325,6 +313,16 @@ static int snd_aw2_create(struct snd_card *card, dev_info(card->dev, "Audiowerk 2 sound card (saa7146 chipset) detected and managed\n"); return 0; + +unmap_io: + iounmap(chip->iobase_virt); +release_regions: + pci_release_regions(pci); +free_data: + kfree(chip); +disable_device: + pci_disable_device(pci); + return err; } /* constructor */ @@ -352,10 +350,8 @@ static int snd_aw2_probe(struct pci_dev *pci, /* (3) Create main component */ err = snd_aw2_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; /* initialize mutex */ mutex_init(&chip->mtx); @@ -373,16 +369,18 @@ static int snd_aw2_probe(struct pci_dev *pci, /* (6) Register card instance */ err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto free_card; /* (7) Set PCI driver data */ pci_set_drvdata(pci, card); dev++; return 0; + +free_card: + snd_card_free(card); + return err; } /* destructor */ @@ -599,10 +597,8 @@ static int snd_aw2_new_pcm(struct aw2 *chip) err = snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0, &pcm_playback_ana); - if (err < 0) { - dev_err(chip->card->dev, "snd_pcm_new error (0x%X)\n", err); - return err; - } + if (err < 0) + goto report_failure; /* Creation ok */ pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_ANA]; @@ -636,11 +632,9 @@ static int snd_aw2_new_pcm(struct aw2 *chip) err = snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0, &pcm_playback_num); + if (err < 0) + goto report_failure; - if (err < 0) { - dev_err(chip->card->dev, "snd_pcm_new error (0x%X)\n", err); - return err; - } /* Creation ok */ pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_DIG]; @@ -673,11 +667,8 @@ static int snd_aw2_new_pcm(struct aw2 *chip) err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1, &pcm_capture); - - if (err < 0) { - dev_err(chip->card->dev, "snd_pcm_new error (0x%X)\n", err); - return err; - } + if (err < 0) + goto report_failure; /* Creation ok */ pcm_device = &chip->device_capture[NUM_STREAM_CAPTURE_ANA]; @@ -718,6 +709,10 @@ static int snd_aw2_new_pcm(struct aw2 *chip) } return 0; + +report_failure: + dev_err(chip->card->dev, "snd_pcm_new error (0x%X)\n", err); + return err; } static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, -- 2.15.0

1 0

[alsa-devel] [PATCH 0/6] ASoC: don't use codec hw_write - step [3/6]
by Kuninori Morimoto 13 Nov '17

13 Nov '17

Hi Mark These are 3/6 step of prepare patches for "replace component". Some codec drivers are using codec->hw_write, codec->control_data, but these are very redundant code, and component doesn't want to have such code. These patches cleanup it. You can check whole replace component patch-set on below. This is one of them Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 Kuninori Morimoto (6): ASoC: don't use codec hw_write on uda1380 ASoC: don't use codec hw_write on tlv320dac33 ASoC: don't use codec hw_write on cx20442/omap-ams-delta ASoC: don't use codec hw_write on twl6040 ASoC: don't use codec hw_write on max98926 ASoC: don't use codec hw_write on max98927 sound/soc/codecs/cx20442.c | 23 ++++++++++------------- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/max98927.c | 1 - sound/soc/codecs/tlv320dac33.c | 12 +++++------- sound/soc/codecs/twl6040.c | 16 ++++++++-------- sound/soc/codecs/uda1380.c | 20 ++++++++++---------- sound/soc/omap/ams-delta.c | 4 ++-- 7 files changed, 36 insertions(+), 42 deletions(-)

2 8

[alsa-devel] snd-hda-codec-hdmi option static_hdmi_pcm=1 causes glitchy audio
by VDR User 13 Nov '17

13 Nov '17

Hi. I was checking my surround sound setup with speaker-test and noticed that setting static_hdmi_pcm=1 for snd-hda-codec-hdmi causes audio to sound glitchy. Not sure what's going on there but thought I should report it. I'm using an Nvidia GT520 video card with audio/video being sent to my surround receiver via HDMI. Kernel 4.13.12, Debian Testing, nvidia driver 375.82 Thanks

1 0

[alsa-devel] [PATCH 0/2] ALSA: au88x0: Fine-tuning for some function implementations
by SF Markus Elfring 12 Nov '17

12 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 12 Nov 2017 22:30:44 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust 40 function calls together with a variable assignment Use common error handling code in two functions sound/pci/au88x0/au88x0.c | 112 +++++++++++++++++++-------------------- sound/pci/au88x0/au88x0_a3d.c | 28 +++++----- sound/pci/au88x0/au88x0_core.c | 78 +++++++++++++-------------- sound/pci/au88x0/au88x0_eq.c | 21 +++++--- sound/pci/au88x0/au88x0_mixer.c | 3 +- sound/pci/au88x0/au88x0_mpu401.c | 14 ++--- sound/pci/au88x0/au88x0_pcm.c | 16 +++--- 7 files changed, 140 insertions(+), 132 deletions(-) -- 2.15.0

1 2

[alsa-devel] [PATCH] ALSA: als4000: Use common error handling code in snd_card_als4000_probe()
by SF Markus Elfring 12 Nov '17

12 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 12 Nov 2017 13:50:48 +0100 * Adjust jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix eight affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/als4000.c | 90 ++++++++++++++++++++++++++++------------------------- 1 file changed, 47 insertions(+), 43 deletions(-) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 26b097edec8c..e63598c036f6 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -867,21 +867,22 @@ static int snd_card_als4000_probe(struct pci_dev *pci, } /* enable PCI device */ - if ((err = pci_enable_device(pci)) < 0) { + err = pci_enable_device(pci); + if (err < 0) return err; - } + /* check, if we can restrict PCI DMA transfers to 24 bits */ if (dma_set_mask(&pci->dev, DMA_BIT_MASK(24)) < 0 || dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(24)) < 0) { dev_err(&pci->dev, "architecture does not support 24bit PCI busmaster DMA\n"); - pci_disable_device(pci); - return -ENXIO; + err = -ENXIO; + goto disable_device; } - if ((err = pci_request_regions(pci, "ALS4000")) < 0) { - pci_disable_device(pci); - return err; - } + err = pci_request_regions(pci, "ALS4000"); + if (err < 0) + goto disable_device; + iobase = pci_resource_start(pci, 0); pci_read_config_word(pci, PCI_COMMAND, &word); @@ -893,8 +894,7 @@ static int snd_card_als4000_probe(struct pci_dev *pci, &card); if (err < 0) { pci_release_regions(pci); - pci_disable_device(pci); - return err; + goto disable_device; } acard = card->private_data; @@ -905,17 +905,18 @@ static int snd_card_als4000_probe(struct pci_dev *pci, /* disable all legacy ISA stuff */ snd_als4000_set_addr(acard->iobase, 0, 0, 0, 0); - if ((err = snd_sbdsp_create(card, - iobase + ALS4K_IOB_10_ADLIB_ADDR0, - pci->irq, + err = snd_sbdsp_create(card, + iobase + ALS4K_IOB_10_ADLIB_ADDR0, + pci->irq, /* internally registered as IRQF_SHARED in case of ALS4000 SB */ - snd_als4000_interrupt, - -1, - -1, - SB_HW_ALS4000, - &chip)) < 0) { - goto out_err; - } + snd_als4000_interrupt, + -1, + -1, + SB_HW_ALS4000, + &chip); + if (err < 0) + goto free_card; + acard->chip = chip; chip->pci = pci; @@ -928,14 +929,14 @@ static int snd_card_als4000_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->alt_port, chip->irq); - if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, - iobase + ALS4K_IOB_30_MIDI_DATA, - MPU401_INFO_INTEGRATED | - MPU401_INFO_IRQ_HOOK, - -1, &chip->rmidi)) < 0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_ALS4000, + iobase + ALS4K_IOB_30_MIDI_DATA, + MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); + if (err < 0) { dev_err(&pci->dev, "no MPU-401 device at 0x%lx?\n", iobase + ALS4K_IOB_30_MIDI_DATA); - goto out_err; + goto free_card; } /* FIXME: ALS4000 has interesting MPU401 configuration features * at ALS4K_CR1A_MPU401_UART_MODE_CONTROL @@ -943,12 +944,13 @@ static int snd_card_als4000_probe(struct pci_dev *pci, * however there doesn't seem to be an ALSA API for this... * SPECS_PAGE: 21 */ - if ((err = snd_als4000_pcm(chip, 0)) < 0) { - goto out_err; - } - if ((err = snd_sbmixer_new(chip)) < 0) { - goto out_err; - } + err = snd_als4000_pcm(chip, 0); + if (err < 0) + goto free_card; + + err = snd_sbmixer_new(chip); + if (err < 0) + goto free_card; if (snd_opl3_create(card, iobase + ALS4K_IOB_10_ADLIB_ADDR0, @@ -958,25 +960,27 @@ static int snd_card_als4000_probe(struct pci_dev *pci, iobase + ALS4K_IOB_10_ADLIB_ADDR0, iobase + ALS4K_IOB_12_ADLIB_ADDR2); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { - goto out_err; - } + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto free_card; } snd_als4000_create_gameport(acard, dev); - if ((err = snd_card_register(card)) < 0) { - goto out_err; - } + err = snd_card_register(card); + if (err < 0) + goto free_card; + pci_set_drvdata(pci, card); dev++; - err = 0; - goto out; + return 0; -out_err: +disable_device: + pci_disable_device(pci); + return err; + +free_card: snd_card_free(card); - -out: return err; } -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: als300: Use common error handling code in two functions
by SF Markus Elfring 12 Nov '17

12 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 12 Nov 2017 12:52:50 +0100 * Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix six affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/als300.c | 65 +++++++++++++++++++++++++++++++----------------------- 1 file changed, 37 insertions(+), 28 deletions(-) diff --git a/sound/pci/als300.c b/sound/pci/als300.c index eaa2d853d922..fd5ba75595cc 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -655,21 +655,22 @@ static int snd_als300_create(struct snd_card *card, }; *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; if (dma_set_mask(&pci->dev, DMA_BIT_MASK(28)) < 0 || dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(28)) < 0) { dev_err(card->dev, "error setting 28bit DMA mask\n"); - pci_disable_device(pci); - return -ENXIO; + err = -ENXIO; + goto disable_device; } pci_set_master(pci); chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } chip->card = card; @@ -678,10 +679,10 @@ static int snd_als300_create(struct snd_card *card, chip->chip_type = chip_type; spin_lock_init(&chip->reg_lock); - if ((err = pci_request_regions(pci, "ALS300")) < 0) { + err = pci_request_regions(pci, "ALS300"); + if (err < 0) { kfree(chip); - pci_disable_device(pci); - return err; + goto disable_device; } chip->port = pci_resource_start(pci, 0); @@ -693,8 +694,8 @@ static int snd_als300_create(struct snd_card *card, if (request_irq(pci->irq, irq_handler, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_als300_free(chip); - return -EBUSY; + err = -EBUSY; + goto free_sound_chip; } chip->irq = pci->irq; @@ -704,24 +705,29 @@ static int snd_als300_create(struct snd_card *card, err = snd_als300_ac97(chip); if (err < 0) { dev_err(card->dev, "Could not create ac97\n"); - snd_als300_free(chip); - return err; + goto free_sound_chip; } - if ((err = snd_als300_new_pcm(chip)) < 0) { + err = snd_als300_new_pcm(chip); + if (err < 0) { dev_err(card->dev, "Could not create PCM\n"); - snd_als300_free(chip); - return err; + goto free_sound_chip; } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, - chip, &ops)) < 0) { - snd_als300_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto free_sound_chip; *rchip = chip; return 0; + +disable_device: + pci_disable_device(pci); + return err; + +free_sound_chip: + snd_als300_free(chip); + return err; } #ifdef CONFIG_PM_SLEEP @@ -776,11 +782,10 @@ static int snd_als300_probe(struct pci_dev *pci, return err; chip_type = pci_id->driver_data; + err = snd_als300_create(card, pci, chip_type, &chip); + if (err < 0) + goto free_card; - if ((err = snd_als300_create(card, pci, chip_type, &chip)) < 0) { - snd_card_free(card); - return err; - } card->private_data = chip; strcpy(card->driver, "ALS300"); @@ -794,13 +799,17 @@ static int snd_als300_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->port, chip->irq); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto free_card; + pci_set_drvdata(pci, card); dev++; return 0; + +free_card: + snd_card_free(card); + return err; } static struct pci_driver als300_driver = { -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: ak4531_codec: Use common error handling code in snd_ak4531_mixer()
by SF Markus Elfring 12 Nov '17

12 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 12 Nov 2017 10:56:19 +0100 * Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix three affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ak4531_codec.c | 28 ++++++++++++++++------------ 1 file changed, 16 insertions(+), 12 deletions(-) diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 2fb1fbba3e5e..208c96d6ae0f 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -399,10 +399,10 @@ int snd_ak4531_mixer(struct snd_card *card, return -ENOMEM; *ak4531 = *_ak4531; mutex_init(&ak4531->reg_mutex); - if ((err = snd_component_add(card, "AK4531")) < 0) { - snd_ak4531_free(ak4531); - return err; - } + err = snd_component_add(card, "AK4531"); + if (err < 0) + goto free_ac; + strcpy(card->mixername, "Asahi Kasei AK4531"); ak4531->write(ak4531, AK4531_RESET, 0x03); /* no RST, PD */ udelay(100); @@ -413,16 +413,16 @@ int snd_ak4531_mixer(struct snd_card *card, ak4531->write(ak4531, idx, ak4531->regs[idx] = snd_ak4531_initial_map[idx]); /* recording source is mixer */ } for (idx = 0; idx < ARRAY_SIZE(snd_ak4531_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ak4531_controls[idx], ak4531))) < 0) { - snd_ak4531_free(ak4531); - return err; - } + err = snd_ctl_add(card, + snd_ctl_new1(&snd_ak4531_controls[idx], + ak4531)); + if (err < 0) + goto free_ac; } snd_ak4531_proc_init(card, ak4531); - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ak4531, &ops)) < 0) { - snd_ak4531_free(ak4531); - return err; - } + err = snd_device_new(card, SNDRV_DEV_CODEC, ak4531, &ops); + if (err < 0) + goto free_ac; #if 0 snd_ak4531_dump(ak4531); @@ -430,6 +430,10 @@ int snd_ak4531_mixer(struct snd_card *card, if (rak4531) *rak4531 = ak4531; return 0; + +free_ac: + snd_ak4531_free(ak4531); + return err; } /* -- 2.15.0

2 1

[alsa-devel] [PATCH] ALSA: ad1889: Use common error handling code in snd_ad1889_create()
by SF Markus Elfring 12 Nov '17

12 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 12 Nov 2017 10:10:53 +0100 * Adjust jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix five affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ad1889.c | 45 +++++++++++++++++++++++++-------------------- 1 file changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 0bf2c04eeada..9f00e63937a5 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -894,21 +894,23 @@ snd_ad1889_create(struct snd_card *card, *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; /* check PCI availability (32bit DMA) */ if (dma_set_mask(&pci->dev, DMA_BIT_MASK(32)) < 0 || dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(32)) < 0) { dev_err(card->dev, "error setting 32-bit DMA mask.\n"); - pci_disable_device(pci); - return -ENXIO; + err = -ENXIO; + goto disable_device; } /* allocate chip specific data with zero-filled memory */ - if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) { - pci_disable_device(pci); - return -ENOMEM; + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + err = -ENOMEM; + goto disable_device; } chip->card = card; @@ -917,15 +919,16 @@ snd_ad1889_create(struct snd_card *card, chip->irq = -1; /* (1) PCI resource allocation */ - if ((err = pci_request_regions(pci, card->driver)) < 0) - goto free_and_ret; + err = pci_request_regions(pci, card->driver); + if (err < 0) + goto free_data; chip->bar = pci_resource_start(pci, 0); chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { dev_err(card->dev, "unable to reserve region.\n"); err = -EBUSY; - goto free_and_ret; + goto free_data; } pci_set_master(pci); @@ -935,32 +938,34 @@ snd_ad1889_create(struct snd_card *card, if (request_irq(pci->irq, snd_ad1889_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "cannot obtain IRQ %d\n", pci->irq); - snd_ad1889_free(chip); - return -EBUSY; + err = -EBUSY; + goto free_sound_chip; } chip->irq = pci->irq; synchronize_irq(chip->irq); /* (2) initialization of the chip hardware */ - if ((err = snd_ad1889_init(chip)) < 0) { - snd_ad1889_free(chip); - return err; - } + err = snd_ad1889_init(chip); + if (err < 0) + goto free_sound_chip; - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_ad1889_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto free_sound_chip; *rchip = chip; return 0; -free_and_ret: +free_data: kfree(chip); +disable_device: pci_disable_device(pci); + return err; +free_sound_chip: + snd_ad1889_free(chip); return err; } -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: ac97_codec: Use common error handling code in snd_ac97_mixer()
by SF Markus Elfring 11 Nov '17

11 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 11 Nov 2017 22:30:51 +0100 * Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. * The script "checkpatch.pl" pointed information out like the following. ERROR: do not use assignment in if condition Thus fix three affected source code places. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ac97/ac97_codec.c | 50 +++++++++++++++++++++++---------------------- 1 file changed, 26 insertions(+), 24 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 1ef7cdf1d3e8..2289cdcf2d48 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2103,8 +2103,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, ac97_err(ac97, "AC'97 %d access is not valid [0x%x], removing mixer.\n", ac97->num, ac97->id); - snd_ac97_free(ac97); - return -EIO; + err = -EIO; + goto free_ac; } pid = look_for_codec_id(snd_ac97_codec_ids, ac97->id); if (pid) @@ -2138,8 +2138,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, ac97_err(ac97, "AC'97 %d access error (not audio or modem codec)\n", ac97->num); - snd_ac97_free(ac97); - return -EACCES; + err = -EACCES; + goto free_ac; } if (bus->ops->reset) // FIXME: always skipping? @@ -2267,14 +2267,12 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } } sprintf(comp, "AC97a:%08x", ac97->id); - if ((err = snd_component_add(card, comp)) < 0) { - snd_ac97_free(ac97); - return err; - } - if (snd_ac97_mixer_build(ac97) < 0) { - snd_ac97_free(ac97); - return -ENOMEM; - } + err = snd_component_add(card, comp); + if (err < 0) + goto free_ac; + + if (snd_ac97_mixer_build(ac97) < 0) + goto e_nomem; } if (ac97_is_modem(ac97)) { char comp[16]; @@ -2287,24 +2285,28 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } } sprintf(comp, "AC97m:%08x", ac97->id); - if ((err = snd_component_add(card, comp)) < 0) { - snd_ac97_free(ac97); - return err; - } - if (snd_ac97_modem_build(card, ac97) < 0) { - snd_ac97_free(ac97); - return -ENOMEM; - } + err = snd_component_add(card, comp); + if (err < 0) + goto free_ac; + + if (snd_ac97_modem_build(card, ac97) < 0) + goto e_nomem; } if (ac97_is_audio(ac97)) update_power_regs(ac97); snd_ac97_proc_init(ac97); - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ac97, &ops)) < 0) { - snd_ac97_free(ac97); - return err; - } + err = snd_device_new(card, SNDRV_DEV_CODEC, ac97, &ops); + if (err < 0) + goto free_ac; + *rac97 = ac97; return 0; + +e_nomem: + err = -ENOMEM; +free_ac: + snd_ac97_free(ac97); + return err; } EXPORT_SYMBOL(snd_ac97_mixer); -- 2.15.0

1 0

[alsa-devel] [PATCH] ALSA: i2c: Use common error handling code in two functions
by SF Markus Elfring 11 Nov '17

11 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 11 Nov 2017 17:10:24 +0100 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/i2c/i2c.c | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index ef2a9afe9e19..9292aa60f3a2 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -282,20 +282,22 @@ static int snd_i2c_bit_sendbytes(struct snd_i2c_device *device, return -EIO; /* not yet implemented */ snd_i2c_bit_start(bus); err = snd_i2c_bit_sendbyte(bus, device->addr << 1); - if (err < 0) { - snd_i2c_bit_hw_stop(bus); - return err; - } + if (err < 0) + goto stop_bus; + while (count-- > 0) { err = snd_i2c_bit_sendbyte(bus, *bytes++); - if (err < 0) { - snd_i2c_bit_hw_stop(bus); - return err; - } + if (err < 0) + goto stop_bus; + res++; } snd_i2c_bit_stop(bus); return res; + +stop_bus: + snd_i2c_bit_hw_stop(bus); + return err; } static int snd_i2c_bit_readbytes(struct snd_i2c_device *device, @@ -308,21 +310,23 @@ static int snd_i2c_bit_readbytes(struct snd_i2c_device *device, return -EIO; /* not yet implemented */ snd_i2c_bit_start(bus); err = snd_i2c_bit_sendbyte(bus, (device->addr << 1) | 1); - if (err < 0) { - snd_i2c_bit_hw_stop(bus); - return err; - } + if (err < 0) + goto stop_bus; + while (count-- > 0) { err = snd_i2c_bit_readbyte(bus, count == 0); - if (err < 0) { - snd_i2c_bit_hw_stop(bus); - return err; - } + if (err < 0) + goto stop_bus; + *bytes++ = (unsigned char)err; res++; } snd_i2c_bit_stop(bus); return res; + +stop_bus: + snd_i2c_bit_hw_stop(bus); + return err; } static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr) -- 2.15.0

1 0

[alsa-devel] [PATCH 0/3] ALSA: ml403-ac97cr: Fine-tuning for five function implementations
by SF Markus Elfring 11 Nov '17

11 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 11 Nov 2017 14:13:12 +0100 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use common error handling code in two functions Delete an unnecessary return statement in snd_ml403_ac97cr_codec_write() Adjust four checks for null pointers sound/drivers/ml403-ac97cr.c | 61 ++++++++++++++++++++++---------------------- 1 file changed, 30 insertions(+), 31 deletions(-) -- 2.15.0

1 3

[alsa-devel] [PATCH 0/3] ALSA: rawmidi: Adjustments for some function implementations
by SF Markus Elfring 11 Nov '17

11 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sat, 11 Nov 2017 12:02:12 +0100 A few update suggestions were taken into account from static source code analysis. Markus Elfring (3): Adjust seven function calls together with a variable assignment Use common error handling code in snd_rawmidi_new() Adjust 21 checks for null pointers sound/core/rawmidi.c | 103 +++++++++++++++++++++++++++++---------------------- 1 file changed, 58 insertions(+), 45 deletions(-) -- 2.15.0

1 3

[alsa-devel] [PATCH] ASoC: rt5663: Check the JD status in the button pushing
by Oder Chiou 10 Nov '17

10 Nov '17

Check the JD status in the button pushing to prevent the IRQ that is locked by button pushing event while the jack unpluging. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 43 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 18dc1ec..b036c9d 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -52,7 +52,7 @@ struct rt5663_priv { struct snd_soc_codec *codec; struct rt5663_platform_data pdata; struct regmap *regmap; - struct delayed_work jack_detect_work; + struct delayed_work jack_detect_work, jd_unplug_work; struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; struct impedance_mapping_table *imp_table; @@ -1941,8 +1941,15 @@ static void rt5663_jack_detect_work(struct work_struct *work) break; } /* button release or spurious interrput*/ - if (btn_type == 0) + if (btn_type == 0) { report = rt5663->jack_type; + cancel_delayed_work_sync( + &rt5663->jd_unplug_work); + } else { + queue_delayed_work(system_wq, + &rt5663->jd_unplug_work, + msecs_to_jiffies(500)); + } } } else { /* jack out */ @@ -1963,6 +1970,37 @@ static void rt5663_jack_detect_work(struct work_struct *work) SND_JACK_BTN_2 | SND_JACK_BTN_3); } +static void rt5663_jd_unplug_work(struct work_struct *work) +{ + struct rt5663_priv *rt5663 = + container_of(work, struct rt5663_priv, jd_unplug_work.work); + struct snd_soc_codec *codec = rt5663->codec; + + if (!codec) + return; + + if (!rt5663_check_jd_status(codec)) { + /* jack out */ + switch (rt5663->codec_ver) { + case CODEC_VER_1: + rt5663_v2_jack_detect(rt5663->codec, 0); + break; + case CODEC_VER_0: + rt5663_jack_detect(rt5663->codec, 0); + break; + default: + dev_err(codec->dev, "Unknown CODEC Version\n"); + } + + snd_soc_jack_report(rt5663->hs_jack, 0, SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + queue_delayed_work(system_wq, &rt5663->jd_unplug_work, + msecs_to_jiffies(500)); + } +} + static const struct snd_kcontrol_new rt5663_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC Playback Volume", RT5663_STO1_DAC_DIG_VOL, @@ -3557,6 +3595,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, } INIT_DELAYED_WORK(&rt5663->jack_detect_work, rt5663_jack_detect_work); + INIT_DELAYED_WORK(&rt5663->jd_unplug_work, rt5663_jd_unplug_work); if (i2c->irq) { ret = request_irq(i2c->irq, rt5663_irq, -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5514: mark PM functions as __maybe_unused
by Arnd Bergmann 10 Nov '17

10 Nov '17

The new functions are only used when CONFIG_PM is enabled, leading to a harmless warning: sound/soc/codecs/rt5514-spi.c:474:12: error: 'rt5514_resume' defined but not used [-Werror=unused-function] sound/soc/codecs/rt5514-spi.c:464:12: error: 'rt5514_suspend' defined but not used [-Werror=unused-function] This marks them as __maybe_unused to make the build silent again. Fixes: 58f1c07d23cd ("ASoC: rt5514: Voice wakeup support.") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/rt5514-spi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 36e0a58ffc87..2df91db765ac 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -461,7 +461,7 @@ static int rt5514_spi_probe(struct spi_device *spi) return 0; } -static int rt5514_suspend(struct device *dev) +static int __maybe_unused rt5514_suspend(struct device *dev) { int irq = to_spi_device(dev)->irq; @@ -471,7 +471,7 @@ static int rt5514_suspend(struct device *dev) return 0; } -static int rt5514_resume(struct device *dev) +static int __maybe_unused rt5514_resume(struct device *dev) { struct snd_soc_platform *platform = snd_soc_lookup_platform(dev); struct rt5514_dsp *rt5514_dsp = -- 2.9.0

2 3

[alsa-devel] [PATCH 0/2] ASoC: add mclk-fs support to audio graph card
by Olivier Moysan 10 Nov '17

10 Nov '17

The aim of these patches is to port mclk-fs property to audio graph card, as originally implemented in simple card. Olivier Moysan (2): ASoC: add mclk-fs to audio graph card binding ASoC: add mclk-fs support to audio graph card .../devicetree/bindings/sound/audio-graph-card.txt | 1 + sound/soc/generic/audio-graph-card.c | 47 ++++++++++++++++++---- 2 files changed, 40 insertions(+), 8 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH 0/4] ASoC: bcm2835: Additional modes and constraint updates
by Matthias Reichl 10 Nov '17

10 Nov '17

Main point of this series is support for justified and DSP modes and implementation of the set_tdm_slot interface in the bcm2835 I2S driver. I've also updated the constraints: The switch from discrete rates to RATE_CONTINUOUS and bump of maximum rates to 384kHz has been tested with various cards. Enforcement of full symmetry fixes issues with split-codec setups (eg separate ADC and DAC). This series brings the bcm2835-i2s driver in line with the downstream rpi tree where we've carried these patches for several months.

3 7

[alsa-devel] [PATCH 1/1] ASoC: bcm2835-i2s: Support 384k sample rate
by Clive Messer 10 Nov '17

10 Nov '17

Add support for sample rates up to 384k to bcm2835-i2s driver, with SNDRV_PCM_RATE_CONTINUOUS and specifying 8k min and 384k max, for both playback and capture. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/bcm/bcm2835-i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..4a67db2 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -578,7 +578,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE @@ -586,7 +588,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.5.5

3 2

[alsa-devel] Applied "ASoC: bcm2835: Support left/right justified and DSP modes" to the asoc tree
by Mark Brown 10 Nov '17

10 Nov '17

The patch ASoC: bcm2835: Support left/right justified and DSP modes has been applied to the asoc tree at https://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 abd4f0e1c24fd5b7241c77506006e9c0982128c0 Mon Sep 17 00:00:00 2001 From: Matthias Reichl <hias(a)horus.com> Date: Wed, 8 Nov 2017 21:03:30 +0100 Subject: [PATCH] ASoC: bcm2835: Support left/right justified and DSP modes DSP modes and left/right justified modes can be supported on bcm2835 by configuring the frame sync polarity and frame sync length registers and by adjusting the channel data position registers. Clock and frame sync polarity handling in hw_params has been refactored to make the interaction between logical rising/falling edge frame start and physical configuration (changed by normal/inverted polarity modes) clearer. Modes where the first active data bit is transmitted immediately after frame start (eg DSP mode B with slot 0 active) only work reliable if bcm2835 is configured as frame master. In frame slave mode channel swap (or shift, this isn't quite clear yet) can occur. Currently the driver only warns if an unstable configuration is detected but doensn't prevent using them. Signed-off-by: Matthias Reichl <hias(a)horus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/bcm2835-i2s.c | 152 +++++++++++++++++++++++++++++--------------- 1 file changed, 99 insertions(+), 53 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 111b4ef7ce35..2e449d7173fc 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -344,6 +344,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int rx_mask, tx_mask; unsigned int rx_ch1_pos, rx_ch2_pos, tx_ch1_pos, tx_ch2_pos; unsigned int mode, format; + bool bit_clock_master = false; + bool frame_sync_master = false; + bool frame_start_falling_edge = false; uint32_t csreg; int ret = 0; @@ -387,16 +390,39 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, if (data_length > slot_width) return -EINVAL; - /* Clock should only be set up here if CPU is clock master */ + /* Check if CPU is bit clock master */ switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: case SND_SOC_DAIFMT_CBS_CFM: - ret = clk_set_rate(dev->clk, bclk_rate); - if (ret) - return ret; + bit_clock_master = true; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + bit_clock_master = false; break; default: + return -EINVAL; + } + + /* Check if CPU is frame sync master */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + frame_sync_master = true; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFM: + frame_sync_master = false; break; + default: + return -EINVAL; + } + + /* Clock should only be set up here if CPU is clock master */ + if (bit_clock_master) { + ret = clk_set_rate(dev->clk, bclk_rate); + if (ret) + return ret; } /* Setup the frame format */ @@ -427,13 +453,41 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* Setup frame sync signal for 50% duty cycle */ framesync_length = frame_length / 2; + frame_start_falling_edge = true; + break; + case SND_SOC_DAIFMT_LEFT_J: + if (slots & 1) + return -EINVAL; + + odd_slot_offset = slots >> 1; + data_delay = 0; + framesync_length = frame_length / 2; + frame_start_falling_edge = false; + break; + case SND_SOC_DAIFMT_RIGHT_J: + if (slots & 1) + return -EINVAL; + + /* Odd frame lengths aren't supported */ + if (frame_length & 1) + return -EINVAL; + + odd_slot_offset = slots >> 1; + data_delay = slot_width - data_length; + framesync_length = frame_length / 2; + frame_start_falling_edge = false; + break; + case SND_SOC_DAIFMT_DSP_A: + data_delay = 1; + framesync_length = 1; + frame_start_falling_edge = false; + break; + case SND_SOC_DAIFMT_DSP_B: + data_delay = 0; + framesync_length = 1; + frame_start_falling_edge = false; break; default: - /* - * TODO - * Others are possible but are not implemented at the moment. - */ - dev_err(dev->dev, "%s:bad format\n", __func__); return -EINVAL; } @@ -442,6 +496,15 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, tx_mask, slot_width, data_delay, odd_slot_offset); + /* + * Transmitting data immediately after frame start, eg + * in left-justified or DSP mode A, only works stable + * if bcm2835 is the frame clock master. + */ + if ((!rx_ch1_pos || !tx_ch1_pos) && !frame_sync_master) + dev_warn(dev->dev, + "Unstable slave config detected, L/R may be swapped"); + /* * Set format for both streams. * We cannot set another frame length @@ -472,62 +535,38 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, mode |= BCM2835_I2S_FLEN(frame_length - 1); mode |= BCM2835_I2S_FSLEN(framesync_length); - /* Master or slave? */ - switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* CPU is master */ - break; - case SND_SOC_DAIFMT_CBM_CFS: - /* - * CODEC is bit clock master - * CPU is frame master - */ + /* CLKM selects bcm2835 clock slave mode */ + if (!bit_clock_master) mode |= BCM2835_I2S_CLKM; - break; - case SND_SOC_DAIFMT_CBS_CFM: - /* - * CODEC is frame master - * CPU is bit clock master - */ + + /* FSM selects bcm2835 frame sync slave mode */ + if (!frame_sync_master) mode |= BCM2835_I2S_FSM; + + /* CLKI selects normal clocking mode, sampling on rising edge */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + mode |= BCM2835_I2S_CLKI; break; - case SND_SOC_DAIFMT_CBM_CFM: - /* CODEC is master */ - mode |= BCM2835_I2S_CLKM; - mode |= BCM2835_I2S_FSM; + case SND_SOC_DAIFMT_IB_NF: + case SND_SOC_DAIFMT_IB_IF: break; default: - dev_err(dev->dev, "%s:bad master\n", __func__); return -EINVAL; } - /* - * Invert clocks? - * - * The BCM approach seems to be inverted to the classical I2S approach. - */ + /* FSI selects frame start on falling edge */ switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - /* None. Therefore, both for BCM */ - mode |= BCM2835_I2S_CLKI; - mode |= BCM2835_I2S_FSI; - break; - case SND_SOC_DAIFMT_IB_IF: - /* Both. Therefore, none for BCM */ + case SND_SOC_DAIFMT_IB_NF: + if (frame_start_falling_edge) + mode |= BCM2835_I2S_FSI; break; case SND_SOC_DAIFMT_NB_IF: - /* - * Invert only frame sync. Therefore, - * invert only bit clock for BCM - */ - mode |= BCM2835_I2S_CLKI; - break; - case SND_SOC_DAIFMT_IB_NF: - /* - * Invert only bit clock. Therefore, - * invert only frame sync for BCM - */ - mode |= BCM2835_I2S_FSI; + case SND_SOC_DAIFMT_IB_IF: + if (!frame_start_falling_edge) + mode |= BCM2835_I2S_FSI; break; default: return -EINVAL; @@ -563,6 +602,13 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, dev_dbg(dev->dev, "sampling rate: %d bclk rate: %d\n", params_rate(params), bclk_rate); + dev_dbg(dev->dev, "CLKM: %d CLKI: %d FSM: %d FSI: %d frame start: %s edge\n", + !!(mode & BCM2835_I2S_CLKM), + !!(mode & BCM2835_I2S_CLKI), + !!(mode & BCM2835_I2S_FSM), + !!(mode & BCM2835_I2S_FSI), + (mode & BCM2835_I2S_FSI) ? "falling" : "rising"); + return ret; } -- 2.15.0

1 0

[alsa-devel] [PATCH v2] ASoC: Replace snd_soc_acpi_check_hid with acpi_dev_present
by Jeremy Cline 10 Nov '17

10 Nov '17

Replace snd_soc_acpi_check_hid() with the generic acpi_dev_present() and remove the now unused snd_soc_acpi_check_hid function. This should have no functional change. Signed-off-by: Jeremy Cline <jeremy(a)jcline.org> --- Changes since v1 (https://patchwork.kernel.org/patch/10051951/): - Rebase against broonie/for-next include/sound/soc-acpi.h | 3 --- sound/soc/soc-acpi.c | 32 ++------------------------------ 2 files changed, 2 insertions(+), 33 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index a7d8d335b043..057805489af3 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -49,9 +49,6 @@ snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines); -/* acpi check hid */ -bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); - /** * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are * related to the hardware, except for the firmware and topology file names. diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index f21df28bc28e..8cdeae977a00 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -49,41 +49,13 @@ const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) } EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid); -static acpi_status snd_soc_acpi_mach_match(acpi_handle handle, u32 level, - void *context, void **ret) -{ - unsigned long long sta; - acpi_status status; - - *(bool *)context = true; - status = acpi_evaluate_integer(handle, "_STA", NULL, &sta); - if (ACPI_FAILURE(status) || !(sta & ACPI_STA_DEVICE_PRESENT)) - *(bool *)context = false; - - return AE_OK; -} - -bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) -{ - acpi_status status; - bool found = false; - - status = acpi_get_devices(hid, snd_soc_acpi_mach_match, &found, NULL); - - if (ACPI_FAILURE(status)) - return false; - - return found; -} -EXPORT_SYMBOL_GPL(snd_soc_acpi_check_hid); - struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { struct snd_soc_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (snd_soc_acpi_check_hid(mach->id) == true) { + if (acpi_dev_present(mach->id, NULL, -1) == true) { if (mach->machine_quirk == NULL) return mach; @@ -163,7 +135,7 @@ struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (snd_soc_acpi_check_hid(codec_list->codecs[i]) != true) + if (acpi_dev_present(codec_list->codecs[i], NULL, -1) != true) return NULL; } -- 2.14.3

2 1

[alsa-devel] [PATCH] ASoC: Intel: common: Replace sst_acpi_check_hid with acpi_dev_present
by Jeremy Cline 10 Nov '17

10 Nov '17

Replace the sst_acpi_check_hid helper function added in commit 915ae2b9f0fe ("ASoC: Intel: Create a helper to search for matching machine") with the generic acpi_dev_present helper function and remove the now unused sst_acpi_check_hid function. This should have no functional change. Signed-off-by: Jeremy Cline <jeremy(a)jcline.org> --- sound/soc/intel/common/sst-acpi.h | 3 --- sound/soc/intel/common/sst-match-acpi.c | 32 ++------------------------------ 2 files changed, 2 insertions(+), 33 deletions(-) diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index afe9b87b8bd5..d7facac83b7e 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -43,9 +43,6 @@ static inline bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); -/* acpi check hid */ -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); - /* Descriptor for SST ASoC machine driver */ struct sst_acpi_mach { /* ACPI ID for the matching machine driver. Audio codec for instance */ diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/intel/common/sst-match-acpi.c index 56d26f36a3cb..f8e1b92de79a 100644 --- a/sound/soc/intel/common/sst-match-acpi.c +++ b/sound/soc/intel/common/sst-match-acpi.c @@ -49,40 +49,12 @@ const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) } EXPORT_SYMBOL_GPL(sst_acpi_find_name_from_hid); -static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, - void *context, void **ret) -{ - unsigned long long sta; - acpi_status status; - - *(bool *)context = true; - status = acpi_evaluate_integer(handle, "_STA", NULL, &sta); - if (ACPI_FAILURE(status) || !(sta & ACPI_STA_DEVICE_PRESENT)) - *(bool *)context = false; - - return AE_OK; -} - -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) -{ - acpi_status status; - bool found = false; - - status = acpi_get_devices(hid, sst_acpi_mach_match, &found, NULL); - - if (ACPI_FAILURE(status)) - return false; - - return found; -} -EXPORT_SYMBOL_GPL(sst_acpi_check_hid); - struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) { struct sst_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (sst_acpi_check_hid(mach->id) == true) { + if (acpi_dev_present(mach->id, NULL, -1) == true) { if (mach->machine_quirk == NULL) return mach; @@ -161,7 +133,7 @@ struct sst_acpi_mach *sst_acpi_codec_list(void *arg) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (sst_acpi_check_hid(codec_list->codecs[i]) != true) + if (acpi_dev_present(codec_list->codecs[i], NULL, -1) != true) return NULL; } -- 2.14.3

3 2

[alsa-devel] C-Media CM6533 limited sample rate
by Luigi Tarenga 10 Nov '17

10 Nov '17

hello everybody, I bought on Amazon a cheap usb soundcard based on CM6533 chip. For a amateur radio project I need a microphone able to sample at 96Khz and from the datasheet the CM6533 can sample up to this rate. Actually I have 2 different project, the first an echolink node that would live happily with a sample rate of 8Khz and another of a software modem that would work better with a sample rate of 96Khz. Unfortunately it seems I can't set this sample rate with arecord or sox. Actually I can only sample at 44100 or 48000 and only in stereo mode. The datasheet say the chip support PCM at 8K/11.025K/16K/22.05K/32K/ 44.1K/48K/96K Hz with 16 or 24 bit depth. (the datasheet can be found here https://www.cmedia.com.tw/support/download_center/25 ) with arecord the error I get is: $ arecord -D hw:2,0 -vv -f S16_LE -c 2 -r 96000 -t raw -d 10 > /dev/null Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 96000 Hz, Stereo Warning: rate is not accurate (requested = 96000Hz, got = 48000Hz) please, try the plug plugin Hardware PCM card 2 'USB Advanced Audio Device' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 #+ | 00%^CAborted by signal Interrupt... same error if I try 8000 Hz. note that another problem is that I have to use 2 channel for sampling while I would prefer to sample in mono since I would have to convert it anyway with an alsa plugin or in userspace and waste some cpu cicle. If I try to sample in mono: $ arecord -D hw:2,0 -vv -f S16_LE -c 1 -r 8000 -t raw -d 10 > /dev/null Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono arecord: set_params:1324: Channels count non available my software version: $ uname -a Linux bullet 4.13.11-1-ARCH #1 SMP PREEMPT Thu Nov 2 10:25:56 CET 2017 x86_64 GNU/Linux $ pacman -Q|grep alsa alsa-lib 1.1.4.1-2 alsa-tools 1.1.3-1 alsa-utils 1.1.4-1 zita-alsa-pcmi 0.2.0-3 lsusb give this output: lsusb -v -d 0d8c:0024 Bus 003 Device 007: ID 0d8c:0024 C-Media Electronics, Inc. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 16 idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0024 bcdDevice 1.11 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 393 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 95 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 6 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 8 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 7 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 8 bNrInPins 2 baSourceID( 0) 1 baSourceID( 1) 7 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bmControls 0x00 iMixer 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x89 EP 9 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 88200 tSamFreq[ 1] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 2 Discrete tSamFreq[ 0] 88200 tSamFreq[ 1] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 77 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 32 >From the output it seems it's reported support for only 44.1Khz or 48Khz. I'm trying to find out what could be the problem, is the snd_usb_audio driver that stick with a safe defaults? is the firmware on the chip that limit the hardware capabilities and when queried by the driver it report only 2 sample rates? The datasheet make few example of usb audio topology configured via firmware so maybe there is a configuration (actually undocumented) that limit the microphone sampe rate to only those 2 (just guessing). Reading the driver source (snd_usb_audio) I undestand that there are some quirks to customize the capabilities of different usb audio card, do you think adding a quirk for this chip and trying to overcome the firmware configuration and adding support for other sample rate can be a solution? If someone could help me going on track to modify the source I can try to prepare a patch to add quirks for this chip or if someone is interested in investigating into the problem I can sponsor a soundcard for testing and debugging (I'ts not much I know, very few dollars). If the firmware is the culprit of the problem does anyone know if there is alternative firmware available and the eeprom is still writeable after the OEM have wrote on it? many thanks Luigi

2 1

[alsa-devel] [PATCH 00/14] soundwire: Add a new SoundWire subsystem
by Vinod Koul 10 Nov '17

10 Nov '17

So today we celebrate Diwali, the festival of lights in this part of the world. On this occasion I would like to add a new Linux subsystem. This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v1 This is first version of the patch which incorporates lots of changes based on the RFC series posted last year https://lkml.org/lkml/2016/10/21/395 Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (10): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence module soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module Documentation/soundwire/summary.txt | 192 +++++++ drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 146 +++++ drivers/soundwire/Kconfig | 37 ++ drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 983 ++++++++++++++++++++++++++++++++ drivers/soundwire/bus.h | 119 ++++ drivers/soundwire/bus_type.c | 245 ++++++++ drivers/soundwire/cadence_master.c | 793 ++++++++++++++++++++++++++ drivers/soundwire/cadence_master.h | 94 +++ drivers/soundwire/intel.c | 395 +++++++++++++ drivers/soundwire/intel.h | 70 +++ drivers/soundwire/intel_init.c | 230 ++++++++ drivers/soundwire/mipi_disco.c | 421 ++++++++++++++ drivers/soundwire/slave.c | 173 ++++++ drivers/soundwire/sysfs.c | 396 +++++++++++++ include/linux/mod_devicetable.h | 7 + include/linux/regmap.h | 37 ++ include/linux/soundwire/sdw.h | 564 ++++++++++++++++++ include/linux/soundwire/sdw_intel.h | 72 +++ include/linux/soundwire/sdw_registers.h | 238 ++++++++ scripts/mod/devicetable-offsets.c | 5 + scripts/mod/file2alias.c | 16 + 26 files changed, 5259 insertions(+) create mode 100644 Documentation/soundwire/summary.txt create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h -- 2.7.4

9 93

[alsa-devel] [RFC PATCH v2 0/7] Fix year 2038 issue for sound subsystem
by Baolin Wang 10 Nov '17

10 Nov '17

Since many structures will use timespec type variables to record time stamp in uapi/asound.h, which are not year 2038 safe on 32bit system. This patchset tries to introduce new structures removing timespec type to compatible native mode and compat mode. Moreover this patchset also converts the internal structrures to use timespec64 type and related APIs. Changes since v1: - Flat all structues to make them more clear. - Re-modify the convertion for struct snd_timer_tread. - Use same coding style when converting struct snd_rawmidi_status. - Use struct snd_timer_user_status64 in compat mode directly when converting struct snd_timer_status. - Add #ifdef __KERNEL__ when converting struct snd_ctl_elem_value. - Re-modify the convertion for struct snd_pcm_sync_ptr. - Fix some building errors. Appreciated for any comments, especially for patch6 which I am not sure if it is correct. Baolin Wang (7): sound: Replace timespec with timespec64 sound: core: Avoid using timespec for struct snd_pcm_status sound: core: Avoid using timespec for struct snd_rawmidi_status sound: core: Avoid using timespec for struct snd_timer_status uapi: sound: Avoid using timespec for struct snd_ctl_elem_value sound: core: Avoid using timespec for struct snd_pcm_sync_ptr sound: core: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 126 +++++++++- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 19 +- sound/core/pcm.c | 20 +- sound/core/pcm_compat.c | 469 ++++++++++++++++++++++++++++--------- sound/core/pcm_lib.c | 33 ++- sound/core/pcm_native.c | 220 +++++++++++++---- sound/core/rawmidi.c | 81 ++++++- sound/core/rawmidi_compat.c | 121 +++++++--- sound/core/timer.c | 253 ++++++++++++++++---- sound/core/timer_compat.c | 67 ++---- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 13 files changed, 1090 insertions(+), 337 deletions(-) -- 1.7.9.5

4 19

[alsa-devel] [PATCH v3] ASoC: rt5663: Delay and retry reading rt5663 ID register
by Oder Chiou 09 Nov '17

09 Nov '17

In the probe, the codec may not be ready for I2C reading or there are some glitches on the i2c line. So if the i2c reading value is incorrect, it will read again after delay. This issue is similar the patch https://patchwork.kernel.org/patch/9681421/. In current project, these 2 devices were connected to the same i2c line, and they met the same problem. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index fcd1643..18dc1ec 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3429,7 +3429,16 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret); return ret; } - regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + + ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + if (ret || (val != RT5663_DEVICE_ID_2 && val != RT5663_DEVICE_ID_1)) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663, retry one time.\n", + val); + msleep(100); + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + } + switch (val) { case RT5663_DEVICE_ID_2: rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_v2_regmap); -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: sun8i-codec: Set the BCLK divider
by Maxime Ripard 09 Nov '17

09 Nov '17

While the current code was reporting to be able to work in master mode, it failed to do so because the BCLK divider wasn't programmed, meaning that the BCLK would run at the PLL's frequency no matter the sample rate. It was obviously a bit too fast. Add support to retrieve the divider to use, and set it. Since our PLL is not always able to generate a perfect multiple of the sample rate, we'll have to choose the closest divider that matches our setup. Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Cc: <stable(a)vger.kernel.org> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- Changes from v1: - Hardcoded the bit width to remain consistent with the rest of the driver --- sound/soc/sunxi/sun8i-codec.c | 51 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index c2ceca485d6a..7a312168f864 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -73,6 +73,7 @@ #define SUN8I_SYS_SR_CTRL_AIF2_FS_MASK GENMASK(11, 8) #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_MASK GENMASK(5, 4) #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK GENMASK(8, 6) +#define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK GENMASK(12, 9) struct sun8i_codec { struct device *dev; @@ -226,12 +227,57 @@ static int sun8i_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +struct sun8i_codec_clk_div { + u8 div; + u8 val; +}; + +static const struct sun8i_codec_clk_div sun8i_codec_bclk_div[] = { + { .div = 1, .val = 0 }, + { .div = 2, .val = 1 }, + { .div = 4, .val = 2 }, + { .div = 6, .val = 3 }, + { .div = 8, .val = 4 }, + { .div = 12, .val = 5 }, + { .div = 16, .val = 6 }, + { .div = 24, .val = 7 }, + { .div = 32, .val = 8 }, + { .div = 48, .val = 9 }, + { .div = 64, .val = 10 }, + { .div = 96, .val = 11 }, + { .div = 128, .val = 12 }, + { .div = 192, .val = 13 }, +}; + +static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec, + unsigned int rate, + unsigned int word_size) +{ + unsigned long clk_rate = clk_get_rate(scodec->clk_module); + unsigned int div = clk_rate / rate / word_size / 2; + unsigned int best_val = 0, best_diff = ~0; + int i; + + for (i = 0; i < ARRAY_SIZE(sun8i_codec_bclk_div); i++) { + const struct sun8i_codec_clk_div *bdiv = &sun8i_codec_bclk_div[i]; + unsigned int diff = abs(bdiv->div - div); + + if (diff < best_diff) { + best_diff = diff; + best_val = bdiv->val; + } + } + + return best_val; +} + static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct sun8i_codec *scodec = snd_soc_codec_get_drvdata(dai->codec); int sample_rate; + u8 bclk_div; /* * The CPU DAI handles only a sample of 16 bits. Configure the @@ -241,6 +287,11 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_MASK, SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16); + bclk_div = sun8i_codec_get_bclk_div(scodec, params_rate(params), 16); + regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, + SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK, + bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV); + regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK, SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16); -- 2.14.3

2 1

[alsa-devel] [PATCH] ASoC: sun8i-codec: Add a comment on the LRCK inversion
by Maxime Ripard 09 Nov '17

09 Nov '17

The current code might be a bit intriguing without having experienced the issue before, and might come up as a mistake. Make explicit what's going on by adding a comment. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun8i-codec.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 7a312168f864..3dd183be08a4 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -198,6 +198,17 @@ static int sun8i_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, BIT(SUN8I_AIF1CLK_CTRL_AIF1_BCLK_INV), value << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_INV); + + /* + * It appears that the DAI and the codec don't share the same + * polarity for the LRCK signal when they mean 'normal' and + * 'inverted' in the datasheet. + * + * Since the DAI here is our regular i2s driver that have been + * tested with way more codecs than just this one, it means + * that the codec probably gets it backward, and we have to + * invert the value here. + */ regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, BIT(SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV), !value << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV); -- 2.14.3

2 1

[alsa-devel] [PATCH] ASoC: da7213: add support for DSP modes
by Pierre-Louis Bossart 09 Nov '17

09 Nov '17

DSP modes are documented in the data sheet but not enabled in the driver. The work-around already implemented for DA7218/9 is also required to make sure the bit clock handling in DSP modes follows ASoC conventions. Tested with ARD-AUDIO-DA7212 and Minnowmax Turbot boards Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/da7213.c | 58 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 1 + 2 files changed, 49 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index cc0b2d2eaf15..41d9b1da27c2 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1220,6 +1220,7 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_codec *codec = codec_dai->codec; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); u8 dai_clk_mode = 0, dai_ctrl = 0; + u8 dai_offset = 0; /* Set master/slave mode */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1234,17 +1235,46 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } /* Set clock normal/inverted */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; - break; - case SND_SOC_DAIFMT_IB_NF: - dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | + DA7213_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } break; - case SND_SOC_DAIFMT_IB_IF: - dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | DA7213_DAI_CLK_POL_INV; + case SND_SOC_DAI_FORMAT_DSP_A: + case SND_SOC_DAI_FORMAT_DSP_B: + /* The bclk is inverted wrt ASoC conventions */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | + DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; + break; + default: + return -EINVAL; + } break; default: return -EINVAL; @@ -1261,6 +1291,13 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_RIGHT_J: dai_ctrl |= DA7213_DAI_FORMAT_RIGHT_J; break; + case SND_SOC_DAI_FORMAT_DSP_A: /* L data MSB after FRM LRC */ + dai_ctrl |= DA7213_DAI_FORMAT_DSP; + dai_offset = 1; + break; + case SND_SOC_DAI_FORMAT_DSP_B: /* L data MSB during FRM LRC */ + dai_ctrl |= DA7213_DAI_FORMAT_DSP; + break; default: return -EINVAL; } @@ -1271,6 +1308,7 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode); snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, dai_ctrl); + snd_soc_write(codec, DA7213_DAI_OFFSET, dai_offset); return 0; } diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 16ef56f77cd4..5a78dba1dcb5 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -188,6 +188,7 @@ #define DA7213_DAI_FORMAT_I2S_MODE (0x0 << 0) #define DA7213_DAI_FORMAT_LEFT_J (0x1 << 0) #define DA7213_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7213_DAI_FORMAT_DSP (0x3 << 0) #define DA7213_DAI_FORMAT_MASK (0x3 << 0) #define DA7213_DAI_WORD_LENGTH_S16_LE (0x0 << 2) #define DA7213_DAI_WORD_LENGTH_S20_LE (0x1 << 2) -- 2.11.0

3 2

[alsa-devel] Applied "ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+" to the asoc tree
by Mark Brown 09 Nov '17

09 Nov '17

The patch ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+ has been applied to the asoc tree at https://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 8a0f95ce48304364979066600eba7cc5c939ba83 Mon Sep 17 00:00:00 2001 From: Andrea Bondavalli <andrea.bondavalli74(a)gmail.com> Date: Tue, 31 Oct 2017 12:58:18 +0100 Subject: [PATCH] ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+ 32bit and 24bit audio capture formats for H3/H2+ are broken because the RX_SAMPLE_BITS and the RX_FIFO_MODE bits of AC_ADC_FIFOC register of the audio codec are not set to operate in 24bit mode but in 16bit mode only. The following patch sets the H3 audio codec registers and the DMA bus width properly when a 24/32bit capture is requested. Signed-off-by: Andrea Bondavalli <andrea.bondavalli74(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index baa9007464ed..5da4efe7a550 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -346,11 +346,6 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, 0x3 << 8, 0x1 << 8); - /* Fill most significant bits with valid data MSB */ - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); - return 0; } @@ -490,6 +485,30 @@ static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), 0); + /* Set the number of sample bits to either 16 or 24 bits */ + if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS)); + + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + 0); + + scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + } else { + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS), + 0); + + /* Fill most significant bits with valid data MSB */ + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); + + scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + } + return 0; } -- 2.15.0

1 0

[alsa-devel] [PATCH v3] ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2
by Alex Deucher 09 Nov '17

09 Nov '17

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Minimum time required between power On of codec and read of RT5645_VENDOR_ID2 is 400msec. We should wait that long before reading the value. TEST=Cold boot the device and check for sound device. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: rework the patch based on mailing list discussion. Just wait before reading the register. v3: Drop the extra read sound/soc/codecs/rt5645.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 23cc2cb8393f..5f24df4fae8e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -55,6 +55,8 @@ MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override"); #define RT5645_HWEQ_NUM 57 +#define TIME_TO_POWER_MS 400 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -3784,6 +3786,12 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); return ret; } + + /* + * Read after 400msec, as it is the interval required between + * read and power On. + */ + msleep(TIME_TO_POWER_MS); regmap_read(regmap, RT5645_VENDOR_ID2, &val); switch (val) { -- 2.13.6

2 3

[alsa-devel] [PATCH] ALSA: hda - Fix memory leak on snd_hdac_device_init error
by Guneshwor Singh 09 Nov '17

09 Nov '17

From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> Free codec vendor name on snd_hdac_device_init error. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/hda/hdac_device.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 19deb306facb..467b949544e2 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -111,6 +111,8 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, return 0; error: + kfree(codec->vendor_name); + put_device(&codec->dev); return err; } -- 2.15.0

4 3

[alsa-devel] [PATCH v2] ASoC: rt5663: Delay and retry reading rt5663 ID register
by Oder Chiou 09 Nov '17

09 Nov '17

In the probe, the codec may not be ready for I2C reading or there are some glitches on the i2c line. So if the i2c reading value is incorrect, it will read again after delay. This issue is similar the patch https://patchwork.kernel.org/patch/9681421/. In current project, these 2 devices were connected to the same i2c line, and they met the same problem. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index fcd1643..5985d8e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3429,7 +3429,16 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret); return ret; } - regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + + ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + if (ret || (val != RT5663_DEVICE_ID_2 && val != RT5663_DEVICE_ID_1)) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663," + " retry one time.\n", val); + msleep(100); + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + } + switch (val) { case RT5663_DEVICE_ID_2: rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_v2_regmap); -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/3] Fixes for AMD Stoney ACP audio
by Alex Deucher 09 Nov '17

09 Nov '17

This patch set is just a couple fixes for the Audio CoProcessor (ACP) on AMD Stoney platforms and a small general fix for the rt5645 codec driver. The entire patch set can also be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=sound-for-next-stoney2 Thanks! Alex v2: - Add a fix for a 64 bit divide in hw_ptr patch - rework rt5645 patch as per Mark's comments Akshu Agrawal (2): ASoC: AMD: Make the driver name consistent across files ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2 Guenter Roeck (1): FIXUP: FROMLIST: ASoC: amd: Report accurate hw_ptr during dma sound/soc/amd/Makefile | 4 ++-- sound/soc/amd/acp-pcm-dma.c | 9 +++++---- sound/soc/codecs/rt5645.c | 9 +++++++++ 3 files changed, 16 insertions(+), 6 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH] ALSA: hda - fix headset mic problem for Dell machines with alc274
by Hui Wang 09 Nov '17

09 Nov '17

Confirmed with Kailang of Realtek, the pin 0x19 is for Headset Mic, and the pin 0x1a is for Headphone Mic, he suggested to apply ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to fix this problem. And we verified applying this FIXUP can fix this problem. Cc: <stable(a)vger.kernel.org> Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b77d43..db1a376 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6568,6 +6568,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0xb7a60130}, + {0x13, 0xb8a61140}, + {0x16, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: cs42l56: Fix reset GPIO name in example DT binding" to the asoc tree
by Mark Brown 08 Nov '17

08 Nov '17

The patch ASoC: cs42l56: Fix reset GPIO name in example DT binding has been applied to the asoc tree at https://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 8adc430603d67e76a0f8491df21654f691acda62 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Wed, 8 Nov 2017 15:24:59 -0600 Subject: [PATCH] ASoC: cs42l56: Fix reset GPIO name in example DT binding The binding states the reset GPIO property shall be named "cirrus,gpio-nreset" and this is what the driver looks for, but the example uses "gpio-reset". Fix this here. Fixes: 3bb40619aca8 ("ASoC: cs42l56: bindings: sound: Add bindings for CS42L56 CODEC") Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/cs42l56.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cs42l56.txt b/Documentation/devicetree/bindings/sound/cs42l56.txt index 4feb0eb27ea4..4ba520a28ae8 100644 --- a/Documentation/devicetree/bindings/sound/cs42l56.txt +++ b/Documentation/devicetree/bindings/sound/cs42l56.txt @@ -55,7 +55,7 @@ Example: codec: codec@4b { compatible = "cirrus,cs42l56"; reg = <0x4b>; - gpio-reset = <&gpio 10 0>; + cirrus,gpio-nreset = <&gpio 10 0>; cirrus,chgfreq-divisor = <0x05>; cirrus.ain1_ref_cfg; cirrus,micbias-lvl = <5>; -- 2.15.0

1 0

[alsa-devel] [PATCH] ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds
by Alex Deucher 08 Nov '17

08 Nov '17

From: Guenter Roeck <groeck(a)chromium.org> ERROR: "__aeabi_uldivmod" [sound/soc/amd/snd-soc-acp-pcm.ko] undefined! 64-bit divides require special operations to avoid build errors on 32-bit systems. [Reword the commit message to make it clearer - Alex] fixes: 61add8147942 (ASoC: amd: Report accurate hw_ptr during dma) Signed-off-by: Guenter Roeck <groeck(a)chromium.org> Reviewed-on: https://chromium-review.googlesource.com/678919 Reviewed-by: Jason Clinton <jclinton(a)chromium.org> (cherry picked from commit 7ca726e80f21abdbaed9a5a70def1c33a26f8533) Reviewed-on: https://chromium-review.googlesource.com/681618 Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 13d040a4d26f..ef7e98ad960c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -856,12 +856,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (bytescount > rtd->renderbytescount) bytescount = bytescount - rtd->renderbytescount; - pos = bytescount % buffersize; } else { if (bytescount > rtd->capturebytescount) bytescount = bytescount - rtd->capturebytescount; - pos = bytescount % buffersize; } + pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } -- 2.13.6

2 1

[alsa-devel] [PATCH 0/3] ASoC: sun8i-codec: A33 codec fixes
by Maxime Ripard 08 Nov '17

08 Nov '17

Hi, The commit 043b8daa578f ("ASoC: sun4i-i2s: Update global enable with bitmask") has broken the A33 codec since it's been merged as part as 4.14-rc1. The reason has been that while that commit was obviously right, one of its side effect was to clear the bit that would set the I2S controller in slave mode, effectively forcing it to act as master. This was actually working by accident since the codec also had its condition on whether to act as a master or a slave backward, meaning that even though we were setting the i2s controller as slave and the codec as master, we were ending up with the exact opposite. The commit mentionned above (rightfully) broke that combination, and we ended up with the two devices in slave mode, which obviously didn't work. The way to fix that is in two parts. The first one is obviously to make the codec act as its proper role. However, that's not sufficient because some logic was missing in the initial driver that was merged to act as such. Indeed the BCLK divider was never programmed, meaning that we would generate a BCLK running at the PLL rate, which is way too fast. The second patch addresses that. The final fix is here to address an issue that has been there for quite some time too and would invert the two channels. It appears that the codec and I2S drivers don't share the same polarity for the LRCK signal. The fix is quite obvious and is to invert whatever value has been programmed in ALSA. All these commits should go as fix in 4.14 if possible, as without them the audio is completely broken. Let me know what you think, Maxime Maxime Ripard (3): ASoC: sun8i-codec: Invert Master / Slave condition ASoC: sun8i-codec: Set the BCLK divider ASoC: sun8i-codec: Fix left and right channels inversion sound/soc/sunxi/sun8i-codec.c | 63 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 58 insertions(+), 5 deletions(-) -- 2.14.3

3 8

[alsa-devel] [PATCH 0/9] ASoC: prepare for Component conversion - step 2
by Kuninori Morimoto 08 Nov '17

08 Nov '17

Hi Mark These are resend of one of "replace component" patch series. I had posted it as below. Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 Because it is big patch-set, to avoid sending patch bomb, I will repost it as small amount of patch set. To review from big picture, please check above patch-set. I will continue to resend remainings of it if this small patch-set were accepted Kuninori Morimoto (9): ASoC: snd_soc_component_driver has snd_pcm_ops ASoC: snd_soc_component_driver has snd_compr_ops ASoC: snd_soc_component_driver has pmdown_time ASoC: snd_soc_component_driver has endianness ASoC: snd_soc_component_driver has non_legacy_dai_naming ASoC: add snd_soc_component_read32 ASoC: add snd_soc_component_xxx_bias_level() ASoC: add snd_soc_component_cache_sync() ASoC: add snd_soc_dapm_kcontrol_component() include/sound/soc.h | 78 ++++++++ sound/soc/soc-compress.c | 461 +++++++++++++++++++++++++++++++++++++++++++---- sound/soc/soc-core.c | 83 +++++---- sound/soc/soc-io.c | 14 ++ sound/soc/soc-pcm.c | 378 ++++++++++++++++++++++++++++++++++++-- 5 files changed, 929 insertions(+), 85 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: add snd_soc_component_xxx_bias_level()" to the asoc tree
by Mark Brown 08 Nov '17

08 Nov '17

The patch ASoC: add snd_soc_component_xxx_bias_level() has been applied to the asoc tree at https://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 10e079d9874802c35dc3f51af922531e5423a6bf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 6 Nov 2017 01:48:37 +0000 Subject: [PATCH] ASoC: add snd_soc_component_xxx_bias_level() snd_soc_codec_xxx_bias_level() (= for Codec) will be removed soon. This patch Component version of it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 747588273c96..f69b13c4a641 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1439,6 +1439,21 @@ static inline void snd_soc_codec_init_bias_level(struct snd_soc_codec *codec, snd_soc_dapm_init_bias_level(snd_soc_codec_get_dapm(codec), level); } +/** + * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level + * @component: The COMPONENT for which to initialize the DAPM bias level + * @level: The DAPM level to initialize to + * + * Initializes the COMPONENT DAPM bias level. See snd_soc_dapm_init_bias_level(). + */ +static inline void +snd_soc_component_init_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + snd_soc_dapm_init_bias_level( + snd_soc_component_get_dapm(component), level); +} + /** * snd_soc_dapm_get_bias_level() - Get current CODEC DAPM bias level * @codec: The CODEC for which to get the DAPM bias level @@ -1451,6 +1466,19 @@ static inline enum snd_soc_bias_level snd_soc_codec_get_bias_level( return snd_soc_dapm_get_bias_level(snd_soc_codec_get_dapm(codec)); } +/** + * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level + * @component: The COMPONENT for which to get the DAPM bias level + * + * Returns: The current DAPM bias level of the COMPONENT. + */ +static inline enum snd_soc_bias_level +snd_soc_component_get_bias_level(struct snd_soc_component *component) +{ + return snd_soc_dapm_get_bias_level( + snd_soc_component_get_dapm(component)); +} + /** * snd_soc_codec_force_bias_level() - Set the CODEC DAPM bias level * @codec: The CODEC for which to set the level @@ -1466,6 +1494,23 @@ static inline int snd_soc_codec_force_bias_level(struct snd_soc_codec *codec, level); } +/** + * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level + * @component: The COMPONENT for which to set the level + * @level: The level to set to + * + * Forces the COMPONENT bias level to a specific state. See + * snd_soc_dapm_force_bias_level(). + */ +static inline int +snd_soc_component_force_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + return snd_soc_dapm_force_bias_level( + snd_soc_component_get_dapm(component), + level); +} + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol -- 2.15.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5514: Voice wakeup support.
by Oder Chiou 08 Nov '17

08 Nov '17

If the rt5514 Wake on Voice device is opened while suspended, it will be able to wake up the system when a voice command is detected. This patch also supports user-space policy to override wakeup behavior by /sys/bus/spi/drivers/rt5514/spi2.0/power/wakeup. Signed-off-by: Chinyue Chen <chinyue(a)chromium.org> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index b90d6d5..0c0e209 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -448,9 +448,35 @@ static int rt5514_spi_probe(struct spi_device *spi) return ret; } + device_init_wakeup(&spi->dev, true); + + return 0; +} + +static int rt5514_suspend(struct device *dev) +{ + int irq = to_spi_device(dev)->irq; + + if (device_may_wakeup(dev)) + enable_irq_wake(irq); + return 0; } +static int rt5514_resume(struct device *dev) +{ + int irq = to_spi_device(dev)->irq; + + if (device_may_wakeup(dev)) + disable_irq_wake(irq); + + return 0; +} + +static const struct dev_pm_ops rt5514_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(rt5514_suspend, rt5514_resume) +}; + static const struct of_device_id rt5514_of_match[] = { { .compatible = "realtek,rt5514", }, {}, @@ -460,6 +486,7 @@ MODULE_DEVICE_TABLE(of, rt5514_of_match); static struct spi_driver rt5514_spi_driver = { .driver = { .name = "rt5514", + .pm = &rt5514_pm_ops, .of_match_table = of_match_ptr(rt5514_of_match), }, .probe = rt5514_spi_probe, -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt5663: Delay and retry reading rt5663 ID register
by Mark Brown 08 Nov '17

08 Nov '17

On Tue, Nov 07, 2017 at 02:21:21AM +0000, Oder Chiou wrote: > The issue is similar the patch https://patchwork.kernel.org/patch/9681421/. > And in current project, these 2 devices were connected to the same i2c host, > and they meet the same problem. OK... can you resend with the log message all on one line please? That makes it easier for people to grep the kernel source to find where the error came from, it's more important than 80 columns.

1 0

[alsa-devel] ASoC: snd_soc_pcm512x_spi: no license, taints kernel
by Randy Dunlap 08 Nov '17

08 Nov '17

kernel is 4.14-rc8. [ 2600.284567] snd_soc_pcm512x_spi: module license 'unspecified' taints kernel. -- ~Randy

1 0

Re: [alsa-devel] [PATCH] ASoC: amd: fix report accurate hw_ptr during dma
by Alex Deucher 08 Nov '17

08 Nov '17

On Wed, Nov 8, 2017 at 2:30 PM, Guenter Roeck <groeck(a)google.com> wrote: > On Wed, Nov 8, 2017 at 11:18 AM, Deucher, Alexander > <Alexander.Deucher(a)amd.com> wrote: >> >> > -----Original Message----- >> > From: Mark Brown [mailto:broonie@kernel.org] >> > Sent: Wednesday, November 08, 2017 1:48 PM >> > To: Alex Deucher >> > Cc: amd-gfx list; alsa-devel(a)alsa-project.org; Maling list - DRI >> > developers; >> > Mukunda, Vijendar; Liam Girdwood; Takashi Iwai; Guenter Roeck; Deucher, >> > Alexander >> > Subject: Re: [PATCH] ASoC: amd: fix report accurate hw_ptr during dma >> > >> > On Wed, Nov 08, 2017 at 01:40:32PM -0500, Alex Deucher wrote: >> > > On Wed, Nov 8, 2017 at 1:22 PM, Mark Brown <broonie(a)kernel.org> >> > wrote: >> > >> > > > Like I said in reply to your other mail please don't resubmit >> > > > already >> > > > applied patches. The current tip of my topic/amd branch appears to >> > > > be >> > > > this very patch, if there's anything needs changing please send an >> > > > incremental patch. >> > >> > > I'm not seeing this one in your tree either. This is just a resend of >> > > Guenter's patch from an hour ago with the chromium stuff removed. >> > > Maybe you already applied it in the interim? >> > >> > Is this different to "ASoC: amd: Report accurate hw_ptr during dma" >> > which was applied at 16:07? >> >> Yes, this is a fix for that patch. It fixes a 64 bit division that wasn't >> properly handled. >> > > In that case, the subject should reflect the problem fixed, the description > should describe the problem, and there should be a Fixes: tag pointing to > the problematic patch. I updated the patch subject and added a fixes tag when I resent it after removing the chrome tags. The subject could have been bit better, but I wasn't sure how far to take it since it wasn't my patch originally. Alex > > Sorry, I was not aware that the problematic patch is already pending > upstream, or I would have submitted a proper patch upstream myself. > > Guenter > >> >> Alex >> >

2 1

Re: [alsa-devel] [PATCH] ASoC: amd: fix report accurate hw_ptr during dma
by Mark Brown 08 Nov '17

08 Nov '17

On Wed, Nov 08, 2017 at 11:13:50AM -0800, Guenter Roeck wrote: > On Wed, Nov 8, 2017 at 10:47 AM, Mark Brown <broonie(a)kernel.org> wrote: > > Is this different to "ASoC: amd: Report accurate hw_ptr during dma" > > which was applied at 16:07? > I suspect we may be getting hit by chromeos-isms again. In Chrome OS, we > don't use Fixes: tags but "FIXUP: <original description>" in the subject > (not that I like it, but it is what it is). So it probably is the same patch that was just resent with a slightly modified subject line? That's what it looked like initially.

2 1

[alsa-devel] [PATCH] ASoC: amd: fix report accurate hw_ptr during dma
by Alex Deucher 08 Nov '17

08 Nov '17

From: Guenter Roeck <groeck(a)chromium.org> ERROR: "__aeabi_uldivmod" [sound/soc/amd/snd-soc-acp-pcm.ko] undefined! 64-bit divides require special operations to avoid build errors on 32-bit systems. fixes: 61add8147942 (ASoC: amd: Report accurate hw_ptr during dma) Signed-off-by: Guenter Roeck <groeck(a)chromium.org> Reviewed-on: https://chromium-review.googlesource.com/678919 Reviewed-by: Jason Clinton <jclinton(a)chromium.org> (cherry picked from commit 7ca726e80f21abdbaed9a5a70def1c33a26f8533) Reviewed-on: https://chromium-review.googlesource.com/681618 Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: resend with the Chromium tags removed and the subject updated. sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 13d040a4d26f..ef7e98ad960c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -856,12 +856,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (bytescount > rtd->renderbytescount) bytescount = bytescount - rtd->renderbytescount; - pos = bytescount % buffersize; } else { if (bytescount > rtd->capturebytescount) bytescount = bytescount - rtd->capturebytescount; - pos = bytescount % buffersize; } + pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } -- 2.13.6

3 4

[alsa-devel] [PATCH 1/2] ASoC: Intel: improve DMADEVICES dependency
by Arnd Bergmann 08 Nov '17

08 Nov '17

As pointed out by Pierre-Louis Bossart, the depndency I added was broader than necessary, only Baytrail and Haswell/Broadwell actually need it, the others don't. At the same time, we have individual entries for the codecs that all have the 'select' statement but now don't need it any more. Fixes: f7a88db6fffd ("ASoC: Intel: fix Kconfig dependencies") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 3 ++- sound/soc/intel/boards/Kconfig | 5 ----- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e18118209b75..bb8be10b8437 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,19 +32,20 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_INTEL_SST_TOPLEVEL tristate "Intel ASoC SST drivers" depends on X86 || COMPILE_TEST - depends on DMADEVICES select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON config SND_SOC_INTEL_HASWELL tristate "Intel ASoC SST driver for Haswell/Broadwell" depends on SND_SOC_INTEL_SST_TOPLEVEL && SND_DMA_SGBUF + depends on DMADEVICES select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_FIRMWARE config SND_SOC_INTEL_BAYTRAIL tristate "Intel ASoC SST driver for Baytrail (legacy)" depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on DMADEVICES select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_FIRMWARE diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 449bc8baaa60..5063f15b4ca4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -19,7 +19,6 @@ config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on SND_SOC_INTEL_MACH depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DMADEVICES depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -32,7 +31,6 @@ config SND_SOC_INTEL_BDW_RT5677_MACH tristate "ASoC Audio driver for Intel Broadwell with RT5677 codec" depends on SND_SOC_INTEL_MACH depends on X86_INTEL_LPSS && GPIOLIB && I2C - depends on DMADEVICES depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT5677 help @@ -43,7 +41,6 @@ config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" depends on SND_SOC_INTEL_MACH depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DMADEVICES depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help @@ -56,7 +53,6 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on SND_SOC_INTEL_MACH depends on X86_INTEL_LPSS && I2C - depends on DMADEVICES depends on SND_SST_IPC_ACPI = n depends on SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 @@ -68,7 +64,6 @@ config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on SND_SOC_INTEL_MACH depends on X86_INTEL_LPSS && I2C - depends on DMADEVICES depends on SND_SST_IPC_ACPI = n depends on SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 -- 2.9.0

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Re: [alsa-devel] [PATCH 3/3] FIXUP: FROMLIST: ASoC: amd: Report accurate hw_ptr during dma
by Alex Deucher 08 Nov '17

08 Nov '17

On Wed, Nov 8, 2017 at 12:29 PM, Guenter Roeck <groeck(a)google.com> wrote: > On Wed, Nov 8, 2017 at 9:24 AM, Alex Deucher <alexdeucher(a)gmail.com> wrote: >> >> From: Guenter Roeck <groeck(a)chromium.org> >> >> ERROR: "__aeabi_uldivmod" [sound/soc/amd/snd-soc-acp-pcm.ko] undefined! >> >> 64-bit divides require special operations to avoid build errors on 32-bit >> systems. >> >> BUG=b:63121716 >> TEST="Build i386:allmodconfig" >> > > Is this an upstream submission ? Fine with me, but it should not include any > chromium specific tags, neither in the subject not in the description. I'm not familiar with which are chromium specific (TEST, BUG, FIXUP, FROMLIST I guess?). The info seems useful to have in the bug, but I can respin if it's a big deal. Alex > > Guenter > >> >> Signed-off-by: Guenter Roeck <groeck(a)chromium.org> >> Reviewed-on: https://chromium-review.googlesource.com/678919 >> Reviewed-by: Jason Clinton <jclinton(a)chromium.org> >> (cherry picked from commit 7ca726e80f21abdbaed9a5a70def1c33a26f8533) >> Reviewed-on: https://chromium-review.googlesource.com/681618 >> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> >> --- >> sound/soc/amd/acp-pcm-dma.c | 3 +-- >> 1 file changed, 1 insertion(+), 2 deletions(-) >> >> diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c >> index 13d040a4d26f..ef7e98ad960c 100644 >> --- a/sound/soc/amd/acp-pcm-dma.c >> +++ b/sound/soc/amd/acp-pcm-dma.c >> @@ -856,12 +856,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct >> snd_pcm_substream *substream) >> if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { >> if (bytescount > rtd->renderbytescount) >> bytescount = bytescount - rtd->renderbytescount; >> - pos = bytescount % buffersize; >> } else { >> if (bytescount > rtd->capturebytescount) >> bytescount = bytescount - rtd->capturebytescount; >> - pos = bytescount % buffersize; >> } >> + pos = do_div(bytescount, buffersize); >> return bytes_to_frames(runtime, pos); >> } >> >> -- >> 2.13.6 >> >

2 2

[alsa-devel] [PATCH 0/2] ASoC: stm32: spdifrx: DMA management fixes
by Olivier Moysan 08 Nov '17

08 Nov '17

This patch-set gathers DMA management fixes for STM32 SPDIFRX. Olivier Moysan (2): ASoC: stm32: spdifrx: fix 16 bits capture ASoC: stm32: spdifrx: fix control DMA error management sound/soc/stm/stm32_spdifrx.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: rt5514-spi: Let the buf_size to align with period_bytes
by Oder Chiou 08 Nov '17

08 Nov '17

The patch lets the buf_size to align with period_bytes to prevent the buffer reading over the real size of the DSP buffer and also avoid to calculate the wrong size of remaining data. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index c722f33..076116b 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -147,8 +147,13 @@ static void rt5514_spi_copy_work(struct work_struct *work) static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { + size_t period_bytes; u8 buf[8]; + if (!rt5514_dsp->substream) + return; + + period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); rt5514_dsp->get_size = 0; /** @@ -176,6 +181,10 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; + if (rt5514_dsp->buf_size % period_bytes) + rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * + period_bytes; + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH] conf/ucm: rt5651: Add missing cdev
by Carlo Caione 08 Nov '17

08 Nov '17

From: Carlo Caione <carlo(a)endlessm.com> Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- src/conf/ucm/bytcr-rt5651/HiFi.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index b9da30f7..8ded2d52 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -7,6 +7,7 @@ SectionVerb { ] DisableSequence [ + cdev "hw:bytcrrt5651" exec "echo HiFi disable sequence" ] -- 2.14.2

2 3

Re: [alsa-devel] ALC275 COEF stuff?
by Kailang 08 Nov '17

08 Nov '17

Hi Takashi, Attach patch for test. Thanks. BR, Kailang > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, November 08, 2017 3:11 PM > To: Kailang > Subject: Re: ALC275 COEF stuff? > > On Wed, 08 Nov 2017 07:50:27 +0100, > Kailang wrote: > > > > > > Hi Takashi, > > > > ALC275 had no EAPD control COEF register. > > So, BIOS verb table default will write COEF Index 0x0e bit 0 to 1. > > This is enable class-D power up. > > Windows restart will remove driver, it will write Index > 0x0e bit 0 to > > 0. (Power down class-D power) Maybe this will resolve the pop noise. > > > > So, I think add write COEF Index 0x0e bit 0 to 1 at EAPD > site will sovle this issue. > > I could make a patch for you. > > Great, thanks! > > BTW, there is another bug report about the missing right > speaker output on Huawei MateBook X with ALC298: > https://bugzilla.kernel.org/show_bug.cgi?id=197801 > > Can it be some missing COEF, too? > > > Takashi > > > > Thanks. > > > > BR, > > Kailang > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Tuesday, November 07, 2017 9:00 PM > > > To: Kailang > > > Subject: ALC275 COEF stuff? > > > > > > Hi Kailang, > > > > > > we got a bug report about the possibly missing COEF setup for > > > ALC275 on Sony VAIO. > > > https://bugzilla.kernel.org/show_bug.cgi?id=197737 > > > > > > Could you check whether the suggested verbs are correct? > > > > > > Thanks! > > > > > > > > > Takashi > > > > > > > > > ------Please consider the environment before printing this e-mail. > > > >

2 1

[alsa-devel] [alsa-lib][PATCH 0/2] ctl: deprecate APIs of dimension information
by Takashi Sakamoto 08 Nov '17

08 Nov '17

Hi, In ALSA control interface of asound.h, 'struct snd_ctl_elem_info' has 'dimen' member to deliver information for multi-dimensional array, however there's no common way to handle the member. As a result, drivers can force userspace applications to handle the information by inconsistent ways. A series of echoaudio drivers is an actual example of the inconsistent usage. This issue was addressed in a commit 51db452df07b ('Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"') to Linux kernel[1]. Fortunately, at present, this feature of interface is just used by the drivers, and there's a way to obsolete usage of the feature. We can obsolete it without large impacts to userland. As a result of discussion at Linux miniconference 2017, usage of 'dimen' member of 'struct snd_ctl_elem_info' is going to be deprecated for future removal. This patchset is for proposed tasks at a development period of Linux kernel v4.15[3]. Some APIs of dimension information are deprecated. [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/s… [2] [alsa-devel] [ANNOUNCE] Audio Mini Summit 2017 at Prague, Oct 27 http://mailman.alsa-project.org/pipermail/alsa-devel/2017-July/123110.html [3] https://github.com/takaswie/presentations/blob/master/20171027/contents.md Takashi Sakamoto (2): ctl: deprecate APIs of dimension information test: obsolete usage of APIs of dimension information src/control/control.c | 12 ++++++++++++ test/user-ctl-element-set.c | 31 ------------------------------- 2 files changed, 12 insertions(+), 31 deletions(-) -- 2.11.0

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Re: [alsa-devel] snd-bebob - Edirol FA66
by Takashi Sakamoto 08 Nov '17

08 Nov '17

(CCed to alsa-devel) On Nov 8 2017 05:50, Pawel wrote: > In start thanks for your great work. > I have such card: > > 1 [FA66 ]: BeBoB - EDIROL FA-66 > EDIROL EDIROL FA-66 (id:2, rev:1), GUID 0040ab0000c38ae5 at fw1.0, S400 > > > .. which ffado drivers , I can start jack with low parameters like; > jackd -P82 -p128 -t5000 -u -dfirewire -dhw:FA66 -r96000 -p128 -n2 > > .. but with alsa I need to use period value 3 and buffer size between 512 and 2048. > There are probably some hardcoded values in alsa driver. Edirol FA-66 uses DM1000 ASIC as communication engine. This ASIC is produced by BridgeCo. in its BeBoB solution. In this solution, IEC 61883-1/6 packet streaming is used to transfer audio/music data between units on IEEE 1394 bus and host driver. All of current implementation of drivers in ALSA firewire stack including BeBoB driver relies on ALSA packet streaming engine for IEC 61883-1/6. This engine runs with hardware interrupts from 1394OHCI host controllers. Interval of the interrupt is currently configured to every 16 interrupts. This mostly equals to 2 msec in actual time frame. In a view from ALSA PCM application, the 2 msec is corresponding to a period of PCM buffer. Thus, users can configure size of period over 2 msec in any PCM substream. Actual value of the size of period is decided depending on used sampling frequency. Furthermore, any interrupt-oriented driver requires 2 or more periods in PCM buffer. Therefore, users can configure size of buffer over multiples of 2 msec according to configured number of periods. That's current design of ALSA IEC 61883-1/6 engine and it's restriction. Regards Takashi Sakamoto

1 1

[alsa-devel] [PATCH v2 0/4] TDA1997x HDMI video receiver
by Tim Harvey 08 Nov '17

08 Nov '17

This is a v4l2 subdev driver supporting the TDA1997x HDMI video receiver. I've tested this on a Gateworks GW54xx with an IMX6Q which uses the TDA19971 with 16bits connected to the IMX6 CSI. For this configuration I've tested both 16bit YUV422 and 8bit BT656 mode. While the driver should support the TDA1993 I do not have one for testing. Further potential development efforts include: - CEC support - HDCP support - mbus format selection support for bus widths that support multiple formats - TDA19972 support (2 inputs) History: v2: - encorporate feedback into dt bindings - change audio dt bindings - implement dv timings enum/cap - remove deprecated g_mbus_config op - fix dv_query_timings - add EDID get/set handling - remove max-pixel-rate support - add audio codec DAI support - added media-ctl and v4l2-compliance details v1: - initial RFC Media device topology: # media-ctl -d /dev/media0 -p Media controller API version 4.13.0 Media device information ------------------------ driver imx-media model imx-media serial bus info hw revision 0x0 driver version 4.13.0 Device topology - entity 1: adv7180 2-0020 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 20004 device node name /dev/v4l-subdev0 pad0: Source [fmt:UYVY8_2X8/720x480 field:interlaced colorspace:smpte170m] -> "ipu2_csi1_mux":1 [] - entity 3: tda19971 2-0048 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev1 pad0: Source [fmt:UYVY8_1X16/0x0] [dv.caps:BT.656/1120 min:640x480@13000000 max:1920x1080@165000000 stds:CEA-861,DMT caps:progressive] [dv.query:no-link] [dv.current:BT.656/1120 0x0p0 (0x0) stds: flags:] -> "ipu1_csi0_mux":1 [ENABLED] - entity 5: ipu1_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev2 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] - entity 9: ipu2_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev3 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] - entity 13: ipu1_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev4 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_vdic":2 [] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf":0 [] - entity 17: ipu1_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev5 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc capture":0 [] - entity 20: ipu1_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video0 pad0: Sink <- "ipu1_ic_prpenc":1 [] - entity 26: ipu1_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev6 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf capture":0 [] - entity 29: ipu1_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video1 pad0: Sink <- "ipu1_ic_prpvf":1 [] - entity 35: ipu2_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev7 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_vdic":2 [] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf":0 [] - entity 39: ipu2_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev8 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc capture":0 [] - entity 42: ipu2_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video2 pad0: Sink <- "ipu2_ic_prpenc":1 [] - entity 48: ipu2_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev9 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf capture":0 [] - entity 51: ipu2_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video3 pad0: Sink <- "ipu2_ic_prpvf":1 [] - entity 57: ipu1_csi0 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev10 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu1_csi0_mux":2 [ENABLED] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi0 capture":0 [ENABLED] - entity 61: ipu1_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video4 pad0: Sink <- "ipu1_csi0":2 [ENABLED] - entity 67: ipu1_csi1 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev11 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi1 capture":0 [] - entity 71: ipu1_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video5 pad0: Sink <- "ipu1_csi1":2 [] - entity 77: ipu2_csi0 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev12 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi0 capture":0 [] - entity 81: ipu2_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video6 pad0: Sink <- "ipu2_csi0":2 [] - entity 87: ipu2_csi1 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev13 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu2_csi1_mux":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi1 capture":0 [] - entity 91: ipu2_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video7 pad0: Sink <- "ipu2_csi1":2 [] - entity 97: ipu1_csi0_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev14 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "tda19971 2-0048":0 [ENABLED] pad2: Source [fmt:unknown/0x0] -> "ipu1_csi0":0 [ENABLED] - entity 101: ipu2_csi1_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev15 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "adv7180 2-0020":0 [] pad2: Source [fmt:unknown/0x0] -> "ipu2_csi1":0 [] v4l2-compliance test results: root@ventana:~# v4l2-compliance -d /dev/video6 Driver Info: Driver name : imx-media-captu Card type : imx-media-capture Bus info : platform:ipu2_csi0 Driver version: 4.13.0 Capabilities : 0x84200001 Video Capture Streaming Extended Pix Format Device Capabilities Device Caps : 0x04200001 Video Capture Streaming Extended Pix Format Compliance test for device /dev/video6 (not using libv4l2): Required ioctls: test VIDIOC_QUERYCAP: OK Allow for multiple opens: test second video open: OK test VIDIOC_QUERYCAP: OK test VIDIOC_G/S_PRIORITY: OK Debug ioctls: test VIDIOC_DBG_G/S_REGISTER: OK (Not Supported) test VIDIOC_LOG_STATUS: OK (Not Supported) Input ioctls: test VIDIOC_G/S_TUNER/ENUM_FREQ_BANDS: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_S_HW_FREQ_SEEK: OK (Not Supported) test VIDIOC_ENUMAUDIO: OK (Not Supported) fail: v4l2-test-input-output.cpp(418): G_INPUT not supported for a capture device test VIDIOC_G/S/ENUMINPUT: FAIL test VIDIOC_G/S_AUDIO: OK (Not Supported) Inputs: 0 Audio Inputs: 0 Tuners: 0 Output ioctls: test VIDIOC_G/S_MODULATOR: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_ENUMAUDOUT: OK (Not Supported) test VIDIOC_G/S/ENUMOUTPUT: OK (Not Supported) test VIDIOC_G/S_AUDOUT: OK (Not Supported) Outputs: 0 Audio Outputs: 0 Modulators: 0 Input/Output configuration ioctls: test VIDIOC_ENUM/G/S/QUERY_STD: OK (Not Supported) test VIDIOC_ENUM/G/S/QUERY_DV_TIMINGS: OK (Not Supported) test VIDIOC_DV_TIMINGS_CAP: OK (Not Supported) test VIDIOC_G/S_EDID: OK (Not Supported) Control ioctls: test VIDIOC_QUERY_EXT_CTRL/QUERYMENU: OK test VIDIOC_QUERYCTRL: OK test VIDIOC_G/S_CTRL: OK fail: v4l2-test-controls.cpp(574): g_ext_ctrls does not support count == 0 test VIDIOC_G/S/TRY_EXT_CTRLS: FAIL test VIDIOC_(UN)SUBSCRIBE_EVENT/DQEVENT: OK (Not Supported) test VIDIOC_G/S_JPEGCOMP: OK (Not Supported) Standard Controls: 0 Private Controls: 0 Format ioctls: test VIDIOC_ENUM_FMT/FRAMESIZES/FRAMEINTERVALS: OK test VIDIOC_G/S_PARM: OK test VIDIOC_G_FBUF: OK (Not Supported) test VIDIOC_G_FMT: OK test VIDIOC_TRY_FMT: OK test VIDIOC_S_FMT: OK test VIDIOC_G_SLICED_VBI_CAP: OK (Not Supported) test Cropping: OK (Not Supported) test Composing: OK (Not Supported) test Scaling: OK (Not Supported) Codec ioctls: test VIDIOC_(TRY_)ENCODER_CMD: OK (Not Supported) test VIDIOC_G_ENC_INDEX: OK (Not Supported) test VIDIOC_(TRY_)DECODER_CMD: OK (Not Supported) Buffer ioctls: test VIDIOC_REQBUFS/CREATE_BUFS/QUERYBUF: OK test VIDIOC_EXPBUF: OK Test input 0: Total: 42, Succeeded: 40, Failed: 2, Warnings: 0 Tim Harvey (4): MAINTAINERS: add entry for NXP TDA1997x driver media: dt-bindings: Add bindings for TDA1997X media: i2c: Add TDA1997x HDMI receiver driver ARM: dts: imx: Add TDA19971 HDMI Receiver to GW54xx .../devicetree/bindings/media/i2c/tda1997x.txt | 179 ++ MAINTAINERS | 8 + arch/arm/boot/dts/imx6q-gw54xx.dts | 102 + arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 29 +- drivers/media/i2c/Kconfig | 9 + drivers/media/i2c/Makefile | 1 + drivers/media/i2c/tda1997x.c | 3336 ++++++++++++++++++++ include/dt-bindings/media/tda1997x.h | 78 + include/media/i2c/tda1997x.h | 53 + 9 files changed, 3792 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/media/i2c/tda1997x.txt create mode 100644 drivers/media/i2c/tda1997x.c create mode 100644 include/dt-bindings/media/tda1997x.h create mode 100644 include/media/i2c/tda1997x.h -- 2.7.4

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[alsa-devel] [PATCH v3] ASoC: Intel: kbl: Modify map for Headset Playback to fix pop-noise
by Naveen Manohar 07 Nov '17

07 Nov '17

Patch fixes wrong path in commit 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop noise. Current topology for Headset results in unwanted pop noise, while switching from spk->hs at the start of Headset Playback. Hence re-introduced mixin-mixout dsp module in topology for headset playback pipe to fix the regression. And the corresponding modification for headset route is updated here. Fixes: 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- v1->v2: updated commit message to suit the change more appropriately. v2->v3: Added Fixes Tag as suggested to explain a regression fix. sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bc..6dcad0a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164..271ae3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, -- 1.9.1

2 1

Re: [alsa-devel] BUG: looking up invalid subclass: 8
by Takashi Iwai 06 Nov '17

06 Nov '17

On Mon, 06 Nov 2017 19:36:14 +0100, syzbot wrote: > > Hello, > > syzkaller hit the following crash on > 5a3517e009e979f21977d362212b7729c5165d92 > git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > turning off the locking correctness validator. > CPU: 0 PID: 2988 Comm: syzkaller395259 Not tainted > 4.14.0-rc7-next-20171103+ #10 > Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:17 [inline] > dump_stack+0x194/0x257 lib/dump_stack.c:53 > look_up_lock_class kernel/locking/lockdep.c:686 [inline] > register_lock_class+0x5f2/0x2c70 kernel/locking/lockdep.c:769 Ah interesting, this is a result of the recent down_read_nested() usage because we allow more depth than the lock subclasses. Below is the quick fix to paper over it. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: seq: Avoid invalid lockdep class warning The recent fix for adding rwsem nesting annotation was using the given "hop" argument as the lock subclass key. Although the idea itself works, it may trigger a kernel warning like: BUG: looking up invalid subclass: 8 .... since the lockdep has a smaller number of subclasses (8) than we currently allow for the hops there (10). The current definition is merely a sanity check for avoiding the too deep delivery paths, and the 8 hops are already enough. So, as a quick fix, just follow the max hops as same as the max lockdep subclasses. Fixes: 1f20f9ff57ca ("ALSA: seq: Fix nested rwsem annotation for lockdep splat") Reported-by: syzbot <syzkaller(a)googlegroups.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/seq_kernel.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index feb58d455560..4b9ee3009aa0 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -49,7 +49,8 @@ typedef union snd_seq_timestamp snd_seq_timestamp_t; #define SNDRV_SEQ_DEFAULT_CLIENT_EVENTS 200 /* max delivery path length */ -#define SNDRV_SEQ_MAX_HOPS 10 +/* NOTE: this shouldn't be greater than MAX_LOCKDEP_SUBCLASSES */ +#define SNDRV_SEQ_MAX_HOPS 8 /* max size of event size */ #define SNDRV_SEQ_MAX_EVENT_LEN 0x3fffffff -- 2.14.3

1 0

[alsa-devel] [PATCH 1/1] ice1712: tiny patch defining i2c eeprom addr to header file
by Konstantinos Tsimpoukas 06 Nov '17

06 Nov '17

Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1712.h | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0e66afa..f1fe497 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2285,7 +2285,7 @@ static unsigned char snd_ice1712_read_i2c(struct snd_ice1712 *ice, static int snd_ice1712_read_eeprom(struct snd_ice1712 *ice, const char *modelname) { - int dev = 0xa0; /* EEPROM device address */ + int dev = ICE_I2C_EEPROM_ADDR; /* I2C EEPROM device address */ unsigned int i, size; struct snd_ice1712_card_info * const *tbl, *c; diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 5cfba09..8ae8742 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -218,8 +218,9 @@ /* - * + * I2C EEPROM Address */ +#define ICE_I2C_EEPROM_ADDR 0xA0 struct snd_ice1712; -- 2.7.4

2 1

[alsa-devel] usb/sound/usx2y: WARNING in usb_stream_start
by Andrey Konovalov 06 Nov '17

06 Nov '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit 3a99df9a3d14cd866b5516f8cba515a3bfd554ab (4.14-rc7+). Looks like there's no check for the actual endpoint types. usb 1-1: BOGUS urb xfer, pipe 0 != type 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:471 usb_submit_urb+0x113e/0x1400 Modules linked in: CPU: 0 PID: 24 Comm: kworker/0:1 Not tainted 4.14.0-rc7-44290-gf28444df2601-dirty #52 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff88006bef5c00 task.stack: ffff88006bf60000 RIP: 0010:usb_submit_urb+0x113e/0x1400 drivers/usb/core/urb.c:470 RSP: 0018:ffff88006bf67440 EFLAGS: 00010286 RAX: 0000000000000029 RBX: ffff880064698c80 RCX: ffffffff812495b5 RDX: 0000000000000000 RSI: ffffffff8124d76a RDI: 0000000000000005 RBP: ffff88006bf674b0 R08: ffff88006bef5c00 R09: 0000000000000006 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000003 R14: ffff880069a25a20 R15: ffff880064698d04 FS: 0000000000000000(0000) GS:ffff88006ca00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fe6fdba3000 CR3: 0000000068470000 CR4: 00000000000006f0 Call Trace: usb_stream_start+0x48a/0x9f0 sound/usb/usx2y/usb_stream.c:690 us122l_start+0x116/0x290 sound/usb/usx2y/us122l.c:365 us122l_create_card sound/usb/usx2y/us122l.c:502 us122l_usb_probe sound/usb/usx2y/us122l.c:588 snd_us122l_probe+0x7de/0x10a0 sound/usb/usx2y/us122l.c:623 usb_probe_interface+0x324/0x940 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x522/0x740 drivers/base/dd.c:557 __device_attach_driver+0x25d/0x2d0 drivers/base/dd.c:653 bus_for_each_drv+0xff/0x160 drivers/base/bus.c:463 __device_attach+0x1a8/0x2a0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1fc/0x2a0 drivers/base/bus.c:523 device_add+0xc27/0x15a0 drivers/base/core.c:1835 usb_set_configuration+0xd4f/0x17a0 drivers/usb/core/message.c:1932 generic_probe+0xbb/0x120 drivers/usb/core/generic.c:174 usb_probe_device+0xab/0x100 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x522/0x740 drivers/base/dd.c:557 __device_attach_driver+0x25d/0x2d0 drivers/base/dd.c:653 bus_for_each_drv+0xff/0x160 drivers/base/bus.c:463 __device_attach+0x1a8/0x2a0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1fc/0x2a0 drivers/base/bus.c:523 device_add+0xc27/0x15a0 drivers/base/core.c:1835 usb_new_device+0x7fa/0x1090 drivers/usb/core/hub.c:2538 hub_port_connect drivers/usb/core/hub.c:4987 hub_port_connect_change drivers/usb/core/hub.c:5093 port_event drivers/usb/core/hub.c:5199 hub_event_impl+0x17b8/0x3440 drivers/usb/core/hub.c:5311 hub_event+0x38/0x50 drivers/usb/core/hub.c:5209 process_one_work+0x925/0x15d0 kernel/workqueue.c:2113 worker_thread+0xef/0x10d0 kernel/workqueue.c:2247 kthread+0x346/0x410 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: 75 aa fd 48 8b 45 d0 48 8d b8 98 00 00 00 e8 4a cf f4 fe 45 89 e8 44 89 e1 4c 89 f2 48 89 c6 48 c7 c7 10 62 20 86 e8 8c 3e 99 fd <0f> ff e9 ce f4 ff ff e8 c6 75 aa fd 48 8b 45 d0 48 8d b8 e8 00 ---[ end trace bcf7836f85bb5e85 ]---

2 3

[alsa-devel] [PATCH] ASoC: rsnd: return -EIO if rsnd_dmaen_request_channel() failed
by Kuninori Morimoto 06 Nov '17

06 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> PTR_ERR(NULL) is success. Normally when a function returns both NULL and error pointers, it means that NULL is not a error. But, rsnd_dmaen_request_channel() returns NULL if requested resource was failed. Let's return -EIO if rsnd_dmaen_request_channel() was failed on rsnd_dmaen_nolock_start(). This patch fixes commit edce5c496c6a ("ASoC: rsnd: Request/Release DMA channel eachtime") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 5bc9ec1..fd557ab 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -219,11 +219,9 @@ static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod, dma->mod_from, dma->mod_to); if (IS_ERR_OR_NULL(dmaen->chan)) { - int ret = PTR_ERR(dmaen->chan); - dmaen->chan = NULL; dev_err(dev, "can't get dma channel\n"); - return ret; + return -EIO; } return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Delay and retry reading rt5663 ID register
by Oder Chiou 06 Nov '17

06 Nov '17

In the probe, the codec may not be ready for I2C reading. So if the value is not correct, it will read again after delay. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index fcd1643..5985d8e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3429,7 +3429,16 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret); return ret; } - regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + + ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + if (ret || (val != RT5663_DEVICE_ID_2 && val != RT5663_DEVICE_ID_1)) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663," + " retry one time.\n", val); + msleep(100); + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + } + switch (val) { case RT5663_DEVICE_ID_2: rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_v2_regmap); -- 2.7.4

2 1

Re: [alsa-devel] BUG: soft lockup
by Dmitry Vyukov 06 Nov '17

06 Nov '17

On Wed, Nov 1, 2017 at 10:54 PM, syzbot <bot+63583aefef5457348dcfa06b87d4fd1378b26b09(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzkaller hit the following crash on > 1d53d908b79d7870d89063062584eead4cf83448 > git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > watchdog: BUG: soft lockup - CPU#3 stuck for 210s! [syzkaller355698:2995] > Modules linked in: > irq event stamp: 23090 > hardirqs last enabled at (23089): [<ffffffff8155d77c>] > __raw_spin_lock_init+0x1c/0x100 kernel/locking/spinlock_debug.c:23 > hardirqs last disabled at (23090): [<ffffffff84d3eb58>] > apic_timer_interrupt+0x98/0xb0 arch/x86/entry/entry_64.S:577 > softirqs last enabled at (1838): [<ffffffff84d4496d>] > __do_softirq+0x74d/0xbd0 kernel/softirq.c:310 > softirqs last disabled at (1825): [<ffffffff81415033>] invoke_softirq > kernel/softirq.c:364 [inline] > softirqs last disabled at (1825): [<ffffffff81415033>] irq_exit+0x1d3/0x210 > kernel/softirq.c:405 > CPU: 3 PID: 2995 Comm: syzkaller355698 Not tainted 4.13.0-rc7-next-20170901+ > #13 > Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 > task: ffff88006b8fc580 task.stack: ffff880069df8000 > RIP: 0010:clear_page_rep+0x7/0x10 arch/x86/lib/clear_page_64.S:19 > RSP: 0000:ffff880069dff6f8 EFLAGS: 00010246 ORIG_RAX: ffffffffffffff10 > RAX: 0000000000000000 RBX: 0000000000000057 RCX: 0000000000000178 > RDX: 0000000000000000 RSI: ffff88006b8fc580 RDI: ffff880064657440 > RBP: ffff880069dff760 R08: 0000000000029190 R09: 0000000000000000 > R10: ffffffffffffffe8 R11: 0000000000000000 R12: dffffc0000000000 > R13: ffff88006b8fc580 R14: ffff88006b8fc580 R15: ffff88006b8fc580 > FS: 0000000000f65880(0000) GS:ffff88006df00000(0000) knlGS:0000000000000000 > CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > CR2: 0000000020c67ff1 CR3: 000000006741d000 CR4: 00000000000006e0 > DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 > DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 > Call Trace: > __do_huge_pmd_anonymous_page mm/huge_memory.c:570 [inline] > do_huge_pmd_anonymous_page+0x59c/0x1ba0 mm/huge_memory.c:728 > create_huge_pmd mm/memory.c:3802 [inline] > __handle_mm_fault+0x1827/0x39c0 mm/memory.c:4005 > handle_mm_fault+0x3bb/0x860 mm/memory.c:4071 > __do_page_fault+0x4f6/0xb60 arch/x86/mm/fault.c:1445 > do_page_fault+0xee/0x720 arch/x86/mm/fault.c:1520 > do_async_page_fault+0x72/0xc0 arch/x86/kernel/kvm.c:266 > async_page_fault+0x22/0x30 arch/x86/entry/entry_64.S:1069 > RIP: 0033:0x4012c8 > RSP: 002b:00007ffc12d93e10 EFLAGS: 00010286 > RAX: ffffffffffffffff RBX: ffffffffffffffff RCX: 0000000000439779 > RDX: 0000000000000000 RSI: 000000002076e000 RDI: ffffffffffffffff > RBP: 00327265636e6575 R08: 0000000000000000 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000286 R12: 7165732f7665642f > R13: 646e732f7665642f R14: 0030656c69662f2e R15: 0000000000000000 > Code: 60 39 d1 fc e9 8d fd ff ff be 08 00 00 00 4c 89 ff e8 2e 39 d1 fc e9 > 1c fd ff ff 90 90 90 90 90 90 90 90 90 b9 00 02 00 00 31 c0 <f3> 48 ab c3 0f > 1f 44 00 00 31 c0 b9 40 00 00 00 66 0f 1f 84 00 > Kernel panic - not syncing: softlockup: hung tasks > CPU: 3 PID: 2995 Comm: syzkaller355698 Tainted: G L > 4.13.0-rc7-next-20170901+ #13 > Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 > Call Trace: > <IRQ> > __dump_stack lib/dump_stack.c:16 [inline] > dump_stack+0x194/0x257 lib/dump_stack.c:52 > panic+0x1e4/0x417 kernel/panic.c:181 > watchdog_timer_fn+0x401/0x410 kernel/watchdog.c:433 > __run_hrtimer kernel/time/hrtimer.c:1213 [inline] > __hrtimer_run_queues+0x349/0xe10 kernel/time/hrtimer.c:1277 > hrtimer_interrupt+0x1d4/0x5f0 kernel/time/hrtimer.c:1311 > local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1021 [inline] > smp_apic_timer_interrupt+0x156/0x710 arch/x86/kernel/apic/apic.c:1046 > apic_timer_interrupt+0x9d/0xb0 arch/x86/entry/entry_64.S:577 > </IRQ> > RIP: 0010:clear_page_rep+0x7/0x10 arch/x86/lib/clear_page_64.S:19 > RSP: 0000:ffff880069dff6f8 EFLAGS: 00010246 ORIG_RAX: ffffffffffffff10 > RAX: 0000000000000000 RBX: 0000000000000057 RCX: 0000000000000178 > RDX: 0000000000000000 RSI: ffff88006b8fc580 RDI: ffff880064657440 > RBP: ffff880069dff760 R08: 0000000000029190 R09: 0000000000000000 > R10: ffffffffffffffe8 R11: 0000000000000000 R12: dffffc0000000000 > R13: ffff88006b8fc580 R14: ffff88006b8fc580 R15: ffff88006b8fc580 > __do_huge_pmd_anonymous_page mm/huge_memory.c:570 [inline] > do_huge_pmd_anonymous_page+0x59c/0x1ba0 mm/huge_memory.c:728 > create_huge_pmd mm/memory.c:3802 [inline] > __handle_mm_fault+0x1827/0x39c0 mm/memory.c:4005 > handle_mm_fault+0x3bb/0x860 mm/memory.c:4071 > __do_page_fault+0x4f6/0xb60 arch/x86/mm/fault.c:1445 > do_page_fault+0xee/0x720 arch/x86/mm/fault.c:1520 > do_async_page_fault+0x72/0xc0 arch/x86/kernel/kvm.c:266 > async_page_fault+0x22/0x30 arch/x86/entry/entry_64.S:1069 > RIP: 0033:0x4012c8 > RSP: 002b:00007ffc12d93e10 EFLAGS: 00010286 > RAX: ffffffffffffffff RBX: ffffffffffffffff RCX: 0000000000439779 > RDX: 0000000000000000 RSI: 000000002076e000 RDI: ffffffffffffffff > RBP: 00327265636e6575 R08: 0000000000000000 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000286 R12: 7165732f7665642f > R13: 646e732f7665642f R14: 0030656c69662f2e R15: 0000000000000000 > Dumping ftrace buffer: > (ftrace buffer empty) > Kernel Offset: disabled > Rebooting in 86400 seconds.. Looking at the reproducer it looks more sound-related, so +alsa maintainers. This reproducer on linux-next 36ef71cae353f88fd6e095e2aaa3e5953af1685d (Oct 20) as well. INFO: rcu_sched detected stalls on CPUs/tasks: 3-...0: (1 GPs behind) idle=d3e/1/4611686018427387906 softirq=4046/4046 fqs=31231 (detected by 2, t=125002 jiffies, g=2422, c=2421, q=5593) Sending NMI from CPU 2 to CPUs 3: NMI backtrace for cpu 3 CPU: 3 PID: 3082 Comm: a.out Not tainted 4.14.0-rc5-next-20171018 #15 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff880063ed4140 task.stack: ffff880063a98000 RIP: 0010:snd_timer_user_interrupt+0x68/0x3f0 sound/core/timer.c:1220 RSP: 0000:ffff88006cb87730 EFLAGS: 00000802 RAX: 0000000000000000 RBX: ffff8800646727c0 RCX: 0000000000000000 RDX: 1ffff1000c8ce4fd RSI: 0000000000000001 RDI: ffff8800646727e8 RBP: ffff88006cb87768 R08: 0000000000000008 R09: 00000000000022df R10: ffff880063ed49d8 R11: ffff880063ed4140 R12: dffffc0000000000 R13: ffff88006b842410 R14: ffff880064672808 R15: 0000000000000001 FS: 0000000001def880(0000) GS:ffff88006cb80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000203a2000 CR3: 000000006b29f006 CR4: 00000000001606e0 Call Trace: <IRQ> snd_timer_interrupt+0xae5/0x1510 sound/core/timer.c:808 snd_hrtimer_callback+0x211/0x3b0 sound/core/hrtimer.c:64 __run_hrtimer kernel/time/hrtimer.c:1213 [inline] __hrtimer_run_queues+0x38f/0x1000 kernel/time/hrtimer.c:1277 hrtimer_interrupt+0x1e2/0x650 kernel/time/hrtimer.c:1311 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] smp_apic_timer_interrupt+0x153/0x6d0 arch/x86/kernel/apic/apic.c:1050 apic_timer_interrupt+0x9d/0xb0 arch/x86/entry/entry_64.S:577 </IRQ> RIP: 0010:clear_page_erms+0x7/0x10 arch/x86/lib/clear_page_64.S:49 RSP: 0000:ffff880063a9f680 EFLAGS: 00010246 ORIG_RAX: ffffffffffffff11 RAX: 0000000000000000 RBX: 0000000000000034 RCX: 0000000000000040 RDX: ffff880063ed4140 RSI: 0000160000000000 RDI: ffff88005bd78fc0 RBP: ffff880063a9f6d0 R08: ffff880063ed4140 R09: 0000000000000000 R10: ffffffffffffffe2 R11: 0000000000000000 R12: dffffc0000000000 R13: ffffea00016f0000 R14: 0000000000000144 R15: 000000000000005e __do_huge_pmd_anonymous_page mm/huge_memory.c:570 [inline] do_huge_pmd_anonymous_page+0x8ad/0x1e70 mm/huge_memory.c:728 create_huge_pmd mm/memory.c:3828 [inline] __handle_mm_fault+0x261d/0x3eb0 mm/memory.c:4031 handle_mm_fault+0x395/0xa10 mm/memory.c:4097 __do_page_fault+0x63f/0xff0 arch/x86/mm/fault.c:1438 do_page_fault+0xee/0x880 arch/x86/mm/fault.c:1514 do_async_page_fault+0x37/0x80 arch/x86/kernel/kvm.c:273 async_page_fault+0x22/0x30 arch/x86/entry/entry_64.S:1069 RIP: 0033:0x427449 RSP: 002b:00007ffd9a2f3568 EFLAGS: 00010202 RAX: 00000000203a2000 RBX: 00000000004002c8 RCX: 0030656c69662f2e RDX: 0000000000000008 RSI: 000000000049813d RDI: 00000000203a2000 RBP: 00007ffd9a2f3580 R08: ffffffffffffffff R09: 0000000000000000 R10: 0000000000000032 R11: 0000000000000246 R12: 0000000000000000 R13: 00000000004068d0 R14: 0000000000406960 R15: 0000000000000000 Code: 00 00 48 8b 5b 40 4c 8d 73 48 4c 89 f7 e8 41 32 1b 01 48 8d 7b 28 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 0f b6 04 02 <84> c0 74 08 3c 03 0f 8e ed 02 00 00 44 8b 63 28 31 ff 44 89 e6 I've sampled cpu stacks before stall message and it seems sound interrupts happen too frequently: Sending NMI from CPU 2 to CPUs 0-1,3: NMI backtrace for cpu 3 CPU: 3 PID: 3082 Comm: a.out Not tainted 4.14.0-rc5-next-20171018 #15 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff880063ed4140 task.stack: ffff880063a98000 RIP: 0010:__lock_acquire+0xaa8/0x4480 kernel/locking/lockdep.c:3373 RSP: 0000:ffff88006cb87110 EFLAGS: 00000046 RAX: 0000000000000003 RBX: 0000000000000001 RCX: 0000000000000000 RDX: 0000000000000004 RSI: 0000000000000000 RDI: ffff88006b125120 RBP: ffff88006cb875e0 R08: 0000000000000001 R09: 0000000000000001 R10: 0000000000000000 R11: ffff880063ed4140 R12: 0000000000000000 R13: ffff88006b125120 R14: 0000000000000000 R15: 0000000000000001 FS: 0000000001def880(0000) GS:ffff88006cb80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000203a2000 CR3: 000000006b29f006 CR4: 00000000001606e0 Call Trace: <IRQ> lock_acquire+0x1d3/0x520 kernel/locking/lockdep.c:3991 __raw_spin_lock include/linux/spinlock_api_smp.h:142 [inline] _raw_spin_lock+0x2a/0x40 kernel/locking/spinlock.c:149 spin_lock include/linux/spinlock.h:314 [inline] snd_timer_user_interrupt+0x4f/0x3f0 sound/core/timer.c:1219 snd_timer_interrupt+0xae5/0x1510 sound/core/timer.c:808 snd_hrtimer_callback+0x211/0x3b0 sound/core/hrtimer.c:64 __run_hrtimer kernel/time/hrtimer.c:1213 [inline] __hrtimer_run_queues+0x38f/0x1000 kernel/time/hrtimer.c:1277 hrtimer_interrupt+0x1e2/0x650 kernel/time/hrtimer.c:1311 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1025 [inline] smp_apic_timer_interrupt+0x153/0x6d0 arch/x86/kernel/apic/apic.c:1050 apic_timer_interrupt+0x9d/0xb0 arch/x86/entry/entry_64.S:577 </IRQ> RIP: 0010:clear_page_erms+0x7/0x10 arch/x86/lib/clear_page_64.S:49 RSP: 0000:ffff880063a9f680 EFLAGS: 00010246 ORIG_RAX: ffffffffffffff11 RAX: 0000000000000000 RBX: 0000000000000034 RCX: 0000000000000040 RDX: ffff880063ed4140 RSI: 0000160000000000 RDI: ffff88005bd78fc0 RBP: ffff880063a9f6d0 R08: ffff880063ed4140 R09: 0000000000000000 R10: ffffffffffffffe2 R11: 0000000000000000 R12: dffffc0000000000 R13: ffffea00016f0000 R14: 0000000000000144 R15: 000000000000005e __do_huge_pmd_anonymous_page mm/huge_memory.c:570 [inline] do_huge_pmd_anonymous_page+0x8ad/0x1e70 mm/huge_memory.c:728 create_huge_pmd mm/memory.c:3828 [inline] __handle_mm_fault+0x261d/0x3eb0 mm/memory.c:4031 handle_mm_fault+0x395/0xa10 mm/memory.c:4097 __do_page_fault+0x63f/0xff0 arch/x86/mm/fault.c:1438 do_page_fault+0xee/0x880 arch/x86/mm/fault.c:1514 do_async_page_fault+0x37/0x80 arch/x86/kernel/kvm.c:273 async_page_fault+0x22/0x30 arch/x86/entry/entry_64.S:1069 RIP: 0033:0x427449 RSP: 002b:00007ffd9a2f3568 EFLAGS: 00010202 RAX: 00000000203a2000 RBX: 00000000004002c8 RCX: 0030656c69662f2e RDX: 0000000000000008 RSI: 000000000049813d RDI: 00000000203a2000 RBP: 00007ffd9a2f3580 R08: ffffffffffffffff R09: 0000000000000000 R10: 0000000000000032 R11: 0000000000000246 R12: 0000000000000000 R13: 00000000004068d0 R14: 0000000000406960 R15: 0000000000000000 Code: ff df 49 8d 7c dd 08 48 89 fa 48 c1 ea 03 80 3c 02 00 0f 85 61 35 00 00 49 8b 44 dd 08 48 85 c0 0f 85 77 f7 ff ff e9 27 f7 ff ff <48> c7 c7 40 c4 0c 86 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 NMI backtrace for cpu 0 CPU: 0 PID: 3086 Comm: a.out Not tainted 4.14.0-rc5-next-20171018 #15 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88006391a7c0 task.stack: ffff880062800000 RIP: 0010:__lock_is_held+0x37/0x140 kernel/locking/lockdep.c:3808 RSP: 0000:ffff880062807578 EFLAGS: 00000806 RAX: dffffc0000000000 RBX: 0000000000000286 RCX: ffffffff81a04fa8 RDX: 1ffff1000c723605 RSI: 00000000ffffffff RDI: ffffffff8613d500 RBP: ffff8800628075b8 R08: ffff88006391a7c0 R09: 0000000000000000 R10: ffffffffffffffe2 R11: 0000000000000000 R12: ffff88006391a7c0 R13: ffffffff8613d500 R14: 00000000000011e5 R15: 000000000000005d FS: 0000000001def880(0000) GS:ffff88006ca00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000205a3fcc CR3: 000000006ae36002 CR4: 00000000001606f0 Call Trace: lock_is_held_type+0x118/0x210 kernel/locking/lockdep.c:4029 lock_is_held include/linux/lockdep.h:436 [inline] rcu_preempt_sleep_check include/linux/rcupdate.h:301 [inline] ___might_sleep+0x295/0x370 kernel/sched/core.c:6012 clear_huge_page+0x3b5/0x750 mm/memory.c:4581 __do_huge_pmd_anonymous_page mm/huge_memory.c:570 [inline] do_huge_pmd_anonymous_page+0x8ad/0x1e70 mm/huge_memory.c:728 create_huge_pmd mm/memory.c:3828 [inline] __handle_mm_fault+0x261d/0x3eb0 mm/memory.c:4031 handle_mm_fault+0x395/0xa10 mm/memory.c:4097 __do_page_fault+0x63f/0xff0 arch/x86/mm/fault.c:1438 do_page_fault+0xee/0x880 arch/x86/mm/fault.c:1514 do_async_page_fault+0x37/0x80 arch/x86/kernel/kvm.c:273 async_page_fault+0x22/0x30 arch/x86/entry/entry_64.S:1069 RIP: 0033:0x401355 RSP: 002b:00007ffd9a2f3570 EFLAGS: 00010202 RAX: 00000000205a3fcc RBX: 00000000004002c8 RCX: 0000000000401620 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00007ffd9a2f3140 RBP: 00007ffd9a2f3580 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000010 R11: 0000000000000246 R12: 0000000000000000 R13: 00000000004068d0 R14: 0000000000406960 R15: 0000000000000000 Code: 65 4c 8b 24 25 c0 64 01 00 49 8d 84 24 68 08 00 00 53 48 89 c2 48 83 ec 18 48 89 45 c8 48 c1 ea 03 48 b8 00 00 00 00 00 fc ff df <48> 89 7d d0 0f b6 04 02 89 75 c4 84 c0 74 08 3c 03 0f 8e c7 00 NMI backtrace for cpu 1 CPU: 1 PID: 3084 Comm: a.out Not tainted 4.14.0-rc5-next-20171018 #15 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff880063f44780 task.stack: ffff880063380000 RIP: 0010:clear_page_erms+0x9/0x10 arch/x86/lib/clear_page_64.S:50 RSP: 0000:ffff880063387680 EFLAGS: 00000246 RAX: 0000000000000000 RBX: 0000000000000047 RCX: 0000000000000000 RDX: ffff880063f44780 RSI: 0000160000000000 RDI: ffff88005e448000 RBP: ffff8800633876d0 R08: ffff880063f44780 R09: 0000000000000000 R10: ffffffffffffffe2 R11: 0000000000000000 R12: 0000000000000000 R13: ffffea0001790000 R14: dffffc0000000000 R15: 0000000000000000 FS: 0000000001def880(0000) GS:ffff88006ca80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000020000fe0 CR3: 000000006b4ef005 CR4: 00000000001606e0 Call Trace: __do_huge_pmd_anonymous_page mm/huge_memory.c:570 [inline] do_huge_pmd_anonymous_page+0x8ad/0x1e70 mm/huge_memory.c:728 create_huge_pmd mm/memory.c:3828 [inline] __handle_mm_fault+0x261d/0x3eb0 mm/memory.c:4031 handle_mm_fault+0x395/0xa10 mm/memory.c:4097 __do_page_fault+0x63f/0xff0 arch/x86/mm/fault.c:1438 do_page_fault+0xee/0x880 arch/x86/mm/fault.c:1514 do_async_page_fault+0x37/0x80 arch/x86/kernel/kvm.c:273 async_page_fault+0x22/0x30 arch/x86/entry/entry_64.S:1069 RIP: 0033:0x4012af RSP: 002b:00007ffd9a2f3570 EFLAGS: 00010246 RAX: 0000000020000fe0 RBX: 00000000004002c8 RCX: 0030656c69662f2e RDX: 0000000000000000 RSI: 0000000000498145 RDI: 00000000203a2008 RBP: 00007ffd9a2f3580 R08: ffffffffffffffff R09: 0000000000000000 R10: 0000000000000032 R11: 0000000000000246 R12: 0000000000000000 R13: 00000000004068d0 R14: 0000000000406960 R15: 0000000000000000 Code: 89 47 18 48 89 47 20 48 89 47 28 48 89 47 30 48 89 47 38 48 8d 7f 40 75 d9 90 c3 0f 1f 80 00 00 00 00 b9 00 10 00 00 31 c0 f3 aa <c3> 90 90 90 90 90 90 55 48 89 e5 41 57 41 56 41 55 41 54 53 48 > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > Please credit me with: Reported-by: syzbot <syzkaller(a)googlegroups.com> > > syzbot will keep track of this bug report. > Once a fix for this bug is committed, please reply to this email with: > #syz fix: exact-commit-title > To mark this as a duplicate of another syzbot report, please reply with: > #syz dup: exact-subject-of-another-report > If it's a one-off invalid bug report, please reply with: > #syz invalid > Note: if the crash happens again, it will cause creation of a new bug > report. > Note: all commands must start from beginning of the line. > > -- > You received this message because you are subscribed to the Google Groups > "syzkaller-bugs" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to syzkaller-bugs+unsubscribe(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/syzkaller-bugs/94eb2c19df188b1926055cf13c…. > For more options, visit https://groups.google.com/d/optout.

3 4

[alsa-devel] ALSA trigger soft lockup since "ALSA: seq: Fix race at creating a queue"
by Jerome Glisse 06 Nov '17

06 Nov '17

On Wed, Nov 01, 2017 at 01:02:27PM -0700, Andrew Morton wrote: > Begin forwarded message: > > Date: Wed, 01 Nov 2017 12:54:16 -0700 > From: syzbot <bot+63583aefef5457348dcfa06b87d4fd1378b26b09(a)syzkaller.appspotmail.com> > To: aaron.lu(a)intel.com, akpm(a)linux-foundation.org, aneesh.kumar(a)linux.vnet.ibm.com, jack(a)suse.cz, kirill.shutemov(a)linux.intel.com, linux-kernel(a)vger.kernel.org, linux-mm(a)kvack.org, mhocko(a)suse.com, minchan(a)kernel.org, peterz(a)infradead.org, rientjes(a)google.com, sfr(a)canb.auug.org.au, shli(a)fb.com, syzkaller-bugs(a)googlegroups.com, willy(a)linux.intel.com, ying.huang(a)intel.com, zi.yan(a)cs.rutgers.edu > Subject: BUG: soft lockup > > > Hello, > > syzkaller hit the following crash on > 1d53d908b79d7870d89063062584eead4cf83448 > git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers Sorry to be the bringer of bad news but after: 4842e98f26dd80be3623c4714a244ba52ea096a8 and 7e1d90f60a0d501c8503e636942ca704a454d910 The attached syzkaller repro trigger a softlockup in workqueue. I am unfamiliar with that code so some familiar in that domain is more likely to find a proper solution. Note with git revert 7e1d90f60a0d501c8503e636942ca704a454d910 git revert 4842e98f26dd80be3623c4714a244ba52ea096a8 fix the issue Cheers, Jérôme

2 6

[alsa-devel] HDA HDMI audio doesn't work on Intel Geminilake machines
by Hui Wang 06 Nov '17

06 Nov '17

We have several Geminilake laptop and desktop machines, on these machines, the HW configs are same and are as shown below: the HDA controller: 00:0e.0 Audio device [0403]: Intel Corporation Device [8086:3198] (rev 01) the HDMI codec: Vendor Id: 0x8086280d VGA controller: 00:02.0 VGA compatible controller [0300]: Intel Corporation Device [8086:3184] (rev 01) (prog-if 00 [VGA controller]) The problem is if we boot the system without plugging the HDMI monitor, the Linux kernel can't find HDMI codec at all. If we plug the HDMI monitor before booting up, the kernel can find the HDMI codec, but there are many errors (as shown below) and the HDMI audio can't play any sound. Codec: Intel Geminilake HDMI Address: 2 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x8086280d Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: N/A Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power: setting=UNKNOWN, actual=UNKNOWN, Error, Clock-stop-OK, Setting-reset Invalid AFG subtree --endcollapse-- [ 25.504069] snd_hda_intel 0000:00:0e.0: azx_get_response timeout, switching to polling mode: last cmd=0x20bf8100 [ 26.516125] snd_hda_intel 0000:00:0e.0: No response from codec, disabling MSI: last cmd=0x20bf8100 [ 27.528058] snd_hda_intel 0000:00:0e.0: azx_get_response timeout, switching to single_cmd mode: last cmd=0x20bf8100 [ 27.528259] azx_single_wait_for_response: 12 callbacks suppressed [ 27.547927] snd_hda_codec_hdmi hdaudioC0D2: Unable to sync register 0x2f0d00. -5 We have tested the latest mainline kernel, but the testing result is the same. So does the existing kernel driver support the HDMI audio well for Geminilake? If it does, what is wrong with those machines, HW issue or BIOS issue? Thanks in advance.

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[alsa-devel] [PATCH 0/4] ASoC: prepare for Component conversion - step [2/6]
by Kuninori Morimoto 06 Nov '17

06 Nov '17

Hi Mark One month passed. I will re-post prepare patches of "replace component" which is not yet accepted. Because it is big patch-set, to avoid sending patch bomb, I will repost it as small amount of patch set. To reviewing from big picture, please check below patch-set. Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 I will continue to resend remainings if this small patch-sets were accepted. Total 6steps are needed Kuninori Morimoto (4): ASoC: add snd_soc_component_read32 ASoC: add snd_soc_component_xxx_bias_level() ASoC: add snd_soc_component_cache_sync() ASoC: add snd_soc_dapm_kcontrol_component() include/sound/soc.h | 78 ++++++++ sound/soc/soc-compress.c | 461 +++++++++++++++++++++++++++++++++++++++++++---- sound/soc/soc-core.c | 83 +++++---- sound/soc/soc-io.c | 14 ++ sound/soc/soc-pcm.c | 378 ++++++++++++++++++++++++++++++++++++-- 5 files changed, 929 insertions(+), 85 deletions(-) -- 1.9.1

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[alsa-devel] [bug report] ASoC: rsnd: Request/Release DMA channel each time
by Dan Carpenter 06 Nov '17

06 Nov '17

Hello Kuninori Morimoto, The patch edce5c496c6a: "ASoC: rsnd: Request/Release DMA channel each time" from Nov 14, 2016, leads to the following static checker warning: sound/soc/sh/rcar/dma.c:214 rsnd_dmaen_nolock_start() warn: passing zero to 'PTR_ERR' sound/soc/sh/rcar/dma.c 192 static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod, 193 struct rsnd_dai_stream *io, 194 struct rsnd_priv *priv) 195 { 196 struct rsnd_dma *dma = rsnd_mod_to_dma(mod); 197 struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); 198 struct device *dev = rsnd_priv_to_dev(priv); 199 200 if (dmaen->chan) { 201 dev_err(dev, "it already has dma channel\n"); 202 return -EIO; 203 } 204 205 /* 206 * DMAEngine request uses mutex lock. 207 * Thus, it shouldn't be called under spinlock. 208 * Let's call it under nolock_start 209 */ 210 dmaen->chan = rsnd_dmaen_request_channel(io, 211 dma->mod_from, 212 dma->mod_to); 213 if (IS_ERR_OR_NULL(dmaen->chan)) { 214 int ret = PTR_ERR(dmaen->chan); 215 216 dmaen->chan = NULL; 217 dev_err(dev, "can't get dma channel\n"); 218 return ret; PTR_ERR(NULL) is success. Normally when a function returns both NULL and error pointers, it means that NULL is not a error. For example, you call get_some_resource_pointer() and it returns NULL because your computer doesn't have one of those resources plugged in, but it returns an error pointer if it's defective or you're out of memory or whatever. In this case it really feels like NULL is returned on error so the right fix is to make rsnd_dmaen_request_channel() never return NULL and never be checked for NULL. 219 } 220 221 return 0; 222 } regards, dan carpenter

2 1

[alsa-devel] [alsa-tools][PATCH] echomixer: obsolete usage of dimension information
by Takashi Sakamoto 05 Nov '17

05 Nov '17

In current implementation for a series of echoaudio drivers, some control element set delivers information to use element values for multi-dimensional array. Essentially, ALSA control interface of asound.h has no common way to parse the information. As a result, the drivers forces 'echomixer' application to parse the information by several ways. This issue was reported by a commit 51db452df07b ('Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"') to Linux kernel. As a result of discussion at Linux miniconference 2017, usage of 'dimen' member of 'struct snd_ctl_elem_info' is going to be deprecated for future removal. According to the above decision, this commit obsoletes usage of 'dimen' information in 'echomixer' application. In a series of echoaudio drivers, a control element set named as 'Channels info' includes duplicated information of the dimension information on value array. Additionally, some of dimension information have fixed values. This commit utilizes them to obsolete usage of dimension information. This patch is tested with Echo Gina 3G. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- echomixer/echomixer.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/echomixer/echomixer.c b/echomixer/echomixer.c index 54d1520..ac9b51a 100644 --- a/echomixer/echomixer.c +++ b/echomixer/echomixer.c @@ -525,6 +525,25 @@ void GetChannels(void) { nLOut=snd_ctl_elem_value_get_integer(control, 2); // Number of output channels fdOut=snd_ctl_elem_value_get_integer(control, 3); // First digital out nPOut=snd_ctl_elem_value_get_integer(control, 4); // Number of virtual output channels (==nLOut on non-vmixer cards) + + mixerControl.outputs = nLOut; + mixerControl.inputs = nIn; + + if (vmixerId) { + vmixerControl.outputs = nLOut; + vmixerControl.inputs = nPOut; + + /* For outputs and inputs. */ + metersStreams = 2; + } else { + /* For outputs, inputs and system outputs. */ + metersStreams = 3; + } + + /* For the number of channels. */ + metersNumber = 16; + /* For each of levels and peaks. */ + metersTypes = 2; } ClockMask=snd_ctl_elem_value_get_integer(control, 5); // Bitmask of available input clocks } @@ -1712,15 +1731,11 @@ int OpenControls(const char *card, const char *cardname) { if (!mixerId) { mixerId=numid; CTLID_DEBUG(("First Mixer id=%d\n", mixerId)); - mixerControl.outputs=snd_ctl_elem_info_get_dimension(info, 0); - mixerControl.inputs=snd_ctl_elem_info_get_dimension(info, 1); } } else if (!strcmp("VMixer Volume", snd_ctl_elem_id_get_name(id))) { if (!vmixerId) { vmixerId=vmixerControl.id=numid; CTLID_DEBUG(("First Vmixer id=%d\n", vmixerId)); - vmixerControl.outputs=snd_ctl_elem_info_get_dimension(info, 0); - vmixerControl.inputs=snd_ctl_elem_info_get_dimension(info, 1); } } else if (!strcmp("PCM Playback Volume", snd_ctl_elem_id_get_name(id))) { pcmoutId=pcmoutControl.id=numid; @@ -1789,9 +1804,6 @@ int OpenControls(const char *card, const char *cardname) { CTLID_DEBUG(("VU-meter switch id=%d\n", numid)); } else if (!strcmp("VU-meters", snd_ctl_elem_id_get_name(id))) { vumetersId=numid; - metersStreams=snd_ctl_elem_info_get_dimension(info, 0); // 2 or 3: output, input and (vmixer cards only) pcm - metersNumber=snd_ctl_elem_info_get_dimension(info, 1); // Number of channels - metersTypes=snd_ctl_elem_info_get_dimension(info, 2); // 2: level and peak CTLID_DEBUG(("VU-meters id=%d\n", numid)); } else if (!strcmp("Channels info", snd_ctl_elem_id_get_name(id))) { channelsId=numid; -- 2.11.0

2 1

[alsa-devel] How to handle jack detection through gpios
by Hans de Goede 03 Nov '17

03 Nov '17

Hi All, Umberto Izzo (in the Cc) mailed me to see if I could help with implementing headphone detection through GPIOS as I've been doing all kinds of Bay / Cherry Trail related work recently. Umberto found this commit: https://github.com/me176c-dev/kernel_asus_me176c/commit/498116d7e7b4718fa06… Which looks interesting, but is specific to the RT5640 driver. Looking at my own test devices I found a device with an ES8316 using a gpio for headphone detection. What Umberto and my device have in common is that the GPIO to use is listed in the ACPI tables under an AMCR0F28 device and that is also what the above commit binds too. So I was wondering if there is a way to have a headphone switch without tying it to a card, while userspace will still know what to do ? ### Another (non audio) problem is that the AMCR0F28 ACPI device seems to be present in the DSTD for almost all Bay / Cherry Trail devices, but its _STA method always returns 0 even on devices where we do need to use gpios for headphone-detection. I'm thinking about solving this by adding the AMCR0F28 device to the list of always present ACPI devices, so that the kernel ignores _STA and creates a platform device for it. And then in the platform driver for the AMCR0F28 platform device use a DMI whitelist, this is not the prettiest solution but it is the best I can come up with atm. As said not really audio related, Regards, Hans

3 6

[alsa-devel] [PATCH] ASoC: wm8741: Fix setting BCLK and LRCLK polarity
by Sergej Sawazki 03 Nov '17

03 Nov '17

After checking the code and the datasheet, it seems like we are handling the clock inversion (SND_SOC_DAIFMT_NB_IF and SND_SOC_DAIFMT_IB_IF) not correctly. >From the datasheet (Table 58): R5 Format Control, BITS[5:4], [BCP:LRP]: (0) 00 = normal BCLK, normal LRCLK (1) 01 = normal BCLK, inverted LRCLK <-- Fix this (2) 10 = inverted BCLK, normal LRCLK (3) 11 = inverted BCLK, inverted LRCLK <-- Fix this Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- sound/soc/codecs/wm8741.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3c96af0..a394dbe 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -333,13 +333,13 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; - case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_NB_IF: iface |= 0x10; break; case SND_SOC_DAIFMT_IB_NF: iface |= 0x20; break; - case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: iface |= 0x30; break; default: -- 2.7.4

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[alsa-devel] [PATCH RESEND] ASoC: wm8741: Fix setting BCLK and LRCLK polarity
by Sergej Sawazki 03 Nov '17

03 Nov '17

After checking the code and the datasheet, it seems like we are handling the clock inversion (SND_SOC_DAIFMT_NB_IF and SND_SOC_DAIFMT_IB_IF) not correctly. >From the datasheet (Table 58): R5 Format Control, BITS[5:4], [BCP:LRP]: (0) 00 = normal BCLK, normal LRCLK (1) 01 = normal BCLK, inverted LRCLK <-- Fix this (2) 10 = inverted BCLK, normal LRCLK (3) 11 = inverted BCLK, inverted LRCLK <-- Fix this Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Based on 'for-next' (e678524) sound/soc/codecs/wm8741.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3c96af0..a394dbe 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -333,13 +333,13 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; - case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_NB_IF: iface |= 0x10; break; case SND_SOC_DAIFMT_IB_NF: iface |= 0x20; break; - case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: iface |= 0x30; break; default: -- 2.7.4

2 5

[alsa-devel] [PATCH v2] ASoC: Intel: kbl: Modify map for Headset Playback to fix pop-noise
by Naveen Manohar 03 Nov '17

03 Nov '17

Patch fixes wrong path in commit 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop noise. Current topology for Headset results in unwanted pop noise, while switching from spk->hs at the start of Headset Playback. Hence re-introduced mixin-mixout dsp module in topology for headset playback pipe to fix the regression. And the corresponding modification for headset route is updated here. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- v1->v2: updated commit message to suit the change more appropriately. sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bc..6dcad0a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164..271ae3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5514: The ACPI also should use the function rt5514_parse_dp()
by Oder Chiou 03 Nov '17

03 Nov '17

The patch fixed that the ACPI cannot access the device property from the function rt5514_parse_dp(). Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index d7956ab..2a5b5d7 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1143,7 +1143,7 @@ static const struct acpi_device_id rt5514_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match); #endif -static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) +static int rt5514_parse_dp(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", &rt5514->pdata.dmic_init_delay); @@ -1183,8 +1183,8 @@ static int rt5514_i2c_probe(struct i2c_client *i2c, if (pdata) rt5514->pdata = *pdata; - else if (i2c->dev.of_node) - rt5514_parse_dt(rt5514, &i2c->dev); + else + rt5514_parse_dp(rt5514, &i2c->dev); rt5514->i2c_regmap = devm_regmap_init_i2c(i2c, &rt5514_i2c_regmap); if (IS_ERR(rt5514->i2c_regmap)) { -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: Intel: kbl: Modify Headset map for Multi-Playback
by Vinod Koul 03 Nov '17

03 Nov '17

On Fri, Nov 03, 2017 at 02:18:34PM +0530, Naveen M wrote: > On Fri, Nov 03, 2017 at 01:51:00PM +0530, Vinod Koul wrote: > > On Fri, Nov 03, 2017 at 08:58:17AM +0530, Naveen Manohar wrote: > > > Current topology for Headset resulted in unwanted pop noise, > > > while switching from spk->hs at the start of Headset Playback. > > > > what is the reason for the pop ? > > > There is an artifact observed at the start of the stream, which is not > seen earlier with mixin-mixout dsp module, through which data was > passing-through. > > > > Modified topology to fix the same. > > > > how does that fix it, sorry am quite unclear about that part > > > Only for headset playback pipe, we re-introduced the mixin-mixout which > was there earlier. This does not fix the problem but removes regression > (pop-noise in headset playback) introduced by earlier > commit-ID:0b06122fc8d0949a0aa7b817d98e031104802611 Then please mention that in the log. IIUC you should say it fixes the wrong path in commit 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop noise. > > > > This modification also requires headset route to be updated. > > > > that seems to be the only change here, so not sure I follow the changes dont > > for this > > > As the topology was modified to earlier path, earlier route change was > reverted for headset playback. > > > > > > > Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> > > > Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> > > > --- > > > sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- > > > sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- > > > 2 files changed, 2 insertions(+), 2 deletions(-) > > > > > > diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c > > > index 6f9a8bc..6dcad0a 100644 > > > --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c > > > +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c > > > @@ -101,7 +101,7 @@ enum { > > > { "ssp0 Tx", NULL, "spk_out" }, > > > > > > { "AIF Playback", NULL, "ssp1 Tx" }, > > > - { "ssp1 Tx", NULL, "hs_out" }, > > > + { "ssp1 Tx", NULL, "codec1_out" }, > > > > > > { "hs_in", NULL, "ssp1 Rx" }, > > > { "ssp1 Rx", NULL, "AIF Capture" }, > > > diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c > > > index 6072164..271ae3c 100644 > > > --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c > > > +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c > > > @@ -109,7 +109,7 @@ enum { > > > { "ssp0 Tx", NULL, "spk_out" }, > > > > > > { "AIF Playback", NULL, "ssp1 Tx" }, > > > - { "ssp1 Tx", NULL, "hs_out" }, > > > + { "ssp1 Tx", NULL, "codec1_out" }, > > > > > > { "hs_in", NULL, "ssp1 Rx" }, > > > { "ssp1 Rx", NULL, "AIF Capture" }, > > > -- > > > 1.9.1 > > > > > > > -- > > ~Vinod > > -- -- ~Vinod

1 0

[alsa-devel] [PATCH] ASoC: Intel: fix Kconfig dependencies
by Arnd Bergmann 02 Nov '17

02 Nov '17

I ran into multiple problems during randconfig builds of the recently changed Kconfig logic for Intel ASoC drivers: - Building without DMADEVICES doesn't work in general - With that dependency added, we can relax the 'depends on X86' again and allow compile-testing, except for SND_SST_ATOM_HIFI2_PLATFORM, which depends on X86 for asm/platform_sst_audio.h - Skylake requires SND_SOC_INTEL_SST_ACPI, so we have to depend on ACPI in turn - Haswell needs SND_DMA_SGBUF for snd_sgbuf_aligned_pages() With the new set of dependencies, I no longer get any build failures. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a59dda39007e..e18118209b75 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -31,13 +31,14 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_INTEL_SST_TOPLEVEL tristate "Intel ASoC SST drivers" - depends on X86 + depends on X86 || COMPILE_TEST + depends on DMADEVICES select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON config SND_SOC_INTEL_HASWELL tristate "Intel ASoC SST driver for Haswell/Broadwell" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL && SND_DMA_SGBUF select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_FIRMWARE @@ -49,12 +50,12 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SST_ATOM_HIFI2_PLATFORM tristate "Intel ASoC SST driver for HiFi2 platforms (*field, *trail)" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL && X86 select SND_SOC_COMPRESS config SND_SOC_INTEL_SKYLAKE tristate "Intel ASoC SST driver for SKL/BXT/KBL/GLK/CNL" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL && PCI && ACPI select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY -- 2.9.0

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[alsa-devel] [INTERNAL][PATCH 0/7] ASoC: stm32: Add SAI synchronization support
by Olivier Moysan 02 Nov '17

02 Nov '17

This patch-set adds support of synchronization features for SAI interface. It also adds minor fixes and improvements. Olivier Moysan (7): ASoC: stm32: Add synchronization to SAI bindings ASoC: stm32: sai: Move static settings to DAI init ASoC: stm32: sai: Fix DMA burst size ASoC: stm32: sai: fix stop management in isr ASoC: stm32: sai: Remove spurious IRQs on stop ASoC: stm32: sai: Fix get reset controller ASoC: stm32: sai: Add synchronization support .../devicetree/bindings/sound/st,stm32-sai.txt | 14 +- sound/soc/stm/stm32_sai.c | 162 ++++++++++++++++++++- sound/soc/stm/stm32_sai.h | 22 ++- sound/soc/stm/stm32_sai_sub.c | 156 +++++++++++++++++--- 4 files changed, 318 insertions(+), 36 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm8741: Use snd_soc_update_bits rather than hard coding
by Charles Keepax 02 Nov '17

02 Nov '17

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Only build tested but Sergej should be able to test on actually hardware for us. Thanks, Charles sound/soc/codecs/wm8741.c | 35 +++++++++++++++++++++-------------- 1 file changed, 21 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b8c1940f2243..3c96af057a3e 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -196,7 +196,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; + unsigned int iface; int i; /* The set of sample rates that can be supported depends on the @@ -223,15 +223,16 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_width(params)) { case 16: + iface = 0x0; break; case 20: - iface |= 0x0001; + iface = 0x1; break; case 24: - iface |= 0x0002; + iface = 0x2; break; case 32: - iface |= 0x0003; + iface = 0x3; break; default: dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d", @@ -242,7 +243,9 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", params_width(params), params_rate(params)); - snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + snd_soc_update_bits(codec, WM8741_FORMAT_CONTROL, WM8741_IWL_MASK, + iface); + return 0; } @@ -295,7 +298,7 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3; + unsigned int iface; /* check master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -308,18 +311,19 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - iface |= 0x0008; + iface = 0x08; break; case SND_SOC_DAIFMT_RIGHT_J: + iface = 0x00; break; case SND_SOC_DAIFMT_LEFT_J: - iface |= 0x0004; + iface = 0x04; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x000C; + iface = 0x0C; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x001C; + iface = 0x1C; break; default: return -EINVAL; @@ -330,13 +334,13 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_IF: - iface |= 0x0010; + iface |= 0x10; break; case SND_SOC_DAIFMT_IB_NF: - iface |= 0x0020; + iface |= 0x20; break; case SND_SOC_DAIFMT_NB_IF: - iface |= 0x0030; + iface |= 0x30; break; default: return -EINVAL; @@ -347,7 +351,10 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, fmt & SND_SOC_DAIFMT_FORMAT_MASK, ((fmt & SND_SOC_DAIFMT_INV_MASK))); - snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + snd_soc_update_bits(codec, WM8741_FORMAT_CONTROL, + WM8741_BCP_MASK | WM8741_LRP_MASK | WM8741_FMT_MASK, + iface); + return 0; } -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: intel: include linux/module.h as needed
by Arnd Bergmann 02 Nov '17

02 Nov '17

The MODULE_DESCRIPTION() macro is only available when including the linux/module.h header. Apparently this is included indirectly from sst-firmware.c in some configurations, but not in others: sound/soc/intel/common/sst-firmware.c:1278:20: error: expected declaration specifiers or '...' before string constant MODULE_DESCRIPTION("Intel SST Firmware Loader"); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/intel/common/sst-firmware.c:1279:16: error: expected declaration specifiers or '...' before string constant This adds the missing include line. Fixes: a395bdd6b24b ("ASoC: intel: Fix sst-dsp dependency on dw stuff") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/common/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 79a9fdf94d38..657afc02f1c4 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -19,6 +19,7 @@ #include <linux/sched.h> #include <linux/firmware.h> #include <linux/export.h> +#include <linux/module.h> #include <linux/platform_device.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: stm32: sai: attempt to fix uninitialized variable
by Arnd Bergmann 02 Nov '17

02 Nov '17

After commit 61fb4ff70377 ("ASoC: stm32: sai: Move static settings to DAI init"), we get a valid warning about the use of an uninitialized variable: sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_hw_params': sound/soc/stm/stm32_sai_sub.c:485:7: error: 'cr1' may be used uninitialized in this function [-Werror=maybe-uninitialized] I could not immediately figure out what the code should do instead, this is the hack I applied locally to shut up the warning. It may or may not be correct, please take a look. Fixes: 61fb4ff70377 ("ASoC: stm32: sai: Move static settings to DAI init") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/stm/stm32_sai_sub.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 150ad546d8b9..d8a51fce368a 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -480,6 +480,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, /* Mode, data format and channel config */ cr1_mask = SAI_XCR1_DS_MASK; + cr1 = 0; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: fix build warning in soc-core.c
by Randy Dunlap 02 Nov '17

02 Nov '17

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc build error. A symbol that ends with an underscore character ('_') has special meaning in reST (reStructuredText), so add a '*' to prevent this error and to indicate that there are several of these values to choose from. ../sound/soc/soc-core.c:2799: ERROR: Unknown target name: "snd_soc_daifmt". Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-414-rc7.orig/sound/soc/soc-core.c +++ lnx-414-rc7/sound/soc/soc-core.c @@ -2796,7 +2796,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_r /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @fmt: SND_SOC_DAIFMT_ format value. + * @fmt: SND_SOC_DAIFMT_* format value. * * Configures the DAI hardware format and clocking. */

2 1

[alsa-devel] Edirol M-16DX broken after fw upgrade
by András Murányi 02 Nov '17

02 Nov '17

Hi List, The Edirol M-16DX has been working out of the box with ALSA (using firmware v1.21), capable of recording 18 channels, until upgraded to firmware v2.0. symptom: When connected via USB, the device's lcd displays "USB ON LINE" which never disappears. Sound still goes through the device but the interface (knobs, buttons) don't react any more. On the computer, the sound device gets listed, but it cannot be opened. alsa-info: http://www.alsa-project.org/db/?f=d094a5abda1b562c6ba572e4dd9425aa30c8a42d dmesg: [18641.439112] usb 2-4: new high-speed USB device number 5 using ehci-pci [18641.554781] usb 2-4: New USB device found, idVendor=0582, idProduct=00c4 [18641.554794] usb 2-4: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [18641.554802] usb 2-4: Product: M-16DX [18641.554807] usb 2-4: Manufacturer: EDIROL [18647.744008] usb 2-4: 1:1: usb_set_interface failed (-110) [18652.743929] usb 2-4: 1:1: usb_set_interface failed (-110) [18657.744786] usb 2-4: 1:1: usb_set_interface failed (-110) [18662.745466] usb 2-4: 1:1: usb_set_interface failed (-110) [18667.745220] usb 2-4: 1:1: usb_set_interface failed (-110) [18672.746071] usb 2-4: 1:1: usb_set_interface failed (-110) [18677.747007] usb 2-4: 1:1: usb_set_interface failed (-110) [18682.747709] usb 2-4: 1:1: usb_set_interface failed (-110) [18687.748416] usb 2-4: 1:1: usb_set_interface failed (-110) [18692.748261] usb 2-4: 1:1: usb_set_interface failed (-110) [18697.749174] usb 2-4: 1:1: usb_set_interface failed (-110) [18702.749893] usb 2-4: 1:1: usb_set_interface failed (-110) I'll be more than happy to provide more info and to test possible solutions. This is a wonderful device and it was working like a charm until the firmware change. Please, if possible, cc your reply to my personal address too. Thanks, András

2 2

[alsa-devel] [PATCH 1/1] ASoC: stm32: sai: fix warning in stm32_sai_set_config()
by Olivier Moysan 02 Nov '17

02 Nov '17

Fix uninitialized warning introduced by "Move static settings to DAI init" commit in stm32_sai_set_config() function. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 150ad54..08583b9 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -482,13 +482,13 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, cr1_mask = SAI_XCR1_DS_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: - cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); + cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_8); break; case SNDRV_PCM_FORMAT_S16_LE: - cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_16); + cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_16); break; case SNDRV_PCM_FORMAT_S32_LE: - cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_32); + cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32); break; default: dev_err(cpu_dai->dev, "Data format not supported"); -- 1.9.1

2 1

[alsa-devel] [PATCH RESEND1 00/12] ALSA: vsnd: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 02 Nov '17

02 Nov '17

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Thank you, Oleksandr Resending because of rebase onto [2] + added missing patch [1] https://xenproject.org/ [2] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/log/?h=for-… Oleksandr Andrushchenko (12): ALSA: vsnd: Introduce Xen para-virtualized sound frontend driver ALSA: vsnd: Implement driver's probe/remove ALSA: vsnd: Implement Xen bus state handling ALSA: vsnd: Read sound driver configuration from Xen store ALSA: vsnd: Implement Xen event channel handling ALSA: vsnd: Implement handling of shared buffers ALSA: vsnd: Introduce ALSA virtual sound driver ALSA: vsnd: Initialize virtul sound card ALSA: vsnd: Add timer for period interrupt emulation ALSA: vsnd: Implement ALSA PCM operations ALSA: vsnd: Implement communication with backend ALSA: vsnd: Introduce Kconfig option to enable Xen PV sound sound/drivers/Kconfig | 12 + sound/drivers/Makefile | 2 + sound/drivers/xen-front.c | 2107 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 2121 insertions(+) create mode 100644 sound/drivers/xen-front.c -- 2.7.4

5 33

[alsa-devel] [PATCH] ALSA: usb-audio: support new Amanero Combo384 firmware version
by Jussi Laako 02 Nov '17

02 Nov '17

Support DSD_U32_BE sample format on new Amanero Combo384 firmware version on older VID/PID. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5a040e3..77eecaa 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1369,6 +1369,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x199: return SNDRV_PCM_FMTBIT_DSD_U32_LE; case 0x19b: + case 0x203: return SNDRV_PCM_FMTBIT_DSD_U32_BE; default: break; -- 2.7.4

2 1

[alsa-devel] [Patch v6 0/7] Introduce framework for SLIMbus device drivers
by srinivas.kandagatla＠linaro.org 02 Nov '17

02 Nov '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> It's been very long time since there was any activity on the slimbus patches, Am currently working on getting Qualcomm DSP based audio working on top of mainline. Slimbus is one of the major component for getting any analog audio on QCOM SoC's. So am taking intiative to address the review comments on the older patchset and send it. I have tested this patch on IFC6410 board with wcd9310 codec. SLIMbus (Serial Low Power Interchip Media Bus) is a specification developed by MIPI (Mobile Industry Processor Interface) alliance. SLIMbus is a 2-wire implementation, which is used to communicate with peripheral components like audio-codec. SLIMbus uses Time-Division-Multiplexing to accommodate multiple data channels, and control channel. Control channel has messages to do device-enumeration, messages to send/receive control-data to/from slimbus devices, messages for port/channel management, and messages to do bandwidth allocation. Framework is introduced to support multiple instances of the bus (1 controller per bus), and multiple slave devices per controller. SPI and I2C frameworks, and comments from last time when I submitted the patches were referred-to while working on this framework. These patchsets introduce device-management, OF helpers, and messaging APIs, controller driver for Qualcomm's slimbus controller, and clock-pause feature for entering/exiting low-power mode for SLIMbus. Framework patches to do channel, port and bandwidth management are work-in-progress and will be sent out once these initial patches are accepted. These patchsets were tested on Qualcomm Snapdragon processor board using the controller driver, and a test slave device. Changes from V5 to V6: * aligned slim_driver_register more like other buses, suggested by Arnd. * removed boardinfo and add_device apis for now, suggested by Arnd * Few namespace cleanups suggested by Masami * merged of apis in to first patch as suggested by Arnd. * slimbus clients "compatible" name space made much inline with USB and PCIE, suggested by Rob and Arnd. * Removed memory allocations to controller drivers, as suggested by Arnd. * Various bindings comments addressed as suggested by Mark and others. * Added regmap interface so that codecs can write more generic code. * Added MAINTAINER file. Sagar Dharia (5): slimbus: Device management on SLIMbus slimbus: Add messaging APIs to slimbus framework slimbus: qcom: Add Qualcomm Slimbus controller driver slimbus: Add support for 'clock-pause' feature slimbus: qcom: Add runtime-pm support using clock-pause feature Srinivas Kandagatla (2): regmap: add SLIMBUS support MAINTAINERS: Add SLIMbus maintainer Documentation/devicetree/bindings/slimbus/bus.txt | 57 ++ .../devicetree/bindings/slimbus/slim-qcom-ctrl.txt | 43 ++ Documentation/slimbus/summary | 109 ++++ MAINTAINERS | 8 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-slimbus.c | 89 +++ drivers/slimbus/Kconfig | 20 + drivers/slimbus/Makefile | 8 + drivers/slimbus/slim-core.c | 725 +++++++++++++++++++++ drivers/slimbus/slim-messaging.c | 509 +++++++++++++++ drivers/slimbus/slim-qcom-ctrl.c | 714 ++++++++++++++++++++ drivers/slimbus/slim-qcom.h | 64 ++ drivers/slimbus/slim-sched.c | 126 ++++ include/linux/mod_devicetable.h | 13 + include/linux/regmap.h | 18 + include/linux/slimbus.h | 512 +++++++++++++++ 19 files changed, 3023 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/bus.txt create mode 100644 Documentation/devicetree/bindings/slimbus/slim-qcom-ctrl.txt create mode 100644 Documentation/slimbus/summary create mode 100644 drivers/base/regmap/regmap-slimbus.c create mode 100644 drivers/slimbus/Kconfig create mode 100644 drivers/slimbus/Makefile create mode 100644 drivers/slimbus/slim-core.c create mode 100644 drivers/slimbus/slim-messaging.c create mode 100644 drivers/slimbus/slim-qcom-ctrl.c create mode 100644 drivers/slimbus/slim-qcom.h create mode 100644 drivers/slimbus/slim-sched.c create mode 100644 include/linux/slimbus.h -- 2.9.3

11 57

[alsa-devel] High CPU usage on the default device, low CPU usage on a specific device
by Andoru 01 Nov '17

01 Nov '17

As the title says, I'm having a bit of an issue with ALSA, and the high CPU (~15-20%) usage in some cases. So far on the default device, I've only noticed this with VLC and foobar2000 (through Wine), but there are other apps/games that might have the same problem. It's just easy to tell if an app/ALSA uses too much CPU if you just play some audio files and you don't have anything else that might need the extra CPU power, like rendering the video stream or render the graphics. I can fix the high CPU usage in VLC if I switch the audio sub-device from the Default to a specific one (in my case: Analog Front Speakers). There is no change in the audio quality, but the resources used instantly decrease. The only problem with this is that I can no longer play/hear any audio from other apps/games (so basically VLC monopolises the card). Also, this is not possible under Wine, so I can't achieve the same thing for foobar2000, or other Windows apps/games. I haven't made any custom config files for ALSA, except to disable the HDMI audio output (so that the normal analogue output is used by default in all apps/games), and I've done that in /etc/modprobe.d/intel-disable-hdmi.conf with this as it's content: _______________________________________________________________________ options snd_hda_intel enable=0,1 _______________________________________________________________________ The soundcard that I'm using is an integrated one, and it's a Realtek ALC887-VD. Does anyone know what I could do to fix this issue, while still being able to play multiple streams of audio at the same time?

1 0

[alsa-devel] Changing snd_hda_intel.power_save default value ?
by Hans de Goede 01 Nov '17

01 Nov '17

Hi All, I'm working on trying to improve the OOTB power-consumption of Linux (Fedora Workstation) on laptops. One of the easy wins here is setting snd_hda_intel.power_save=1 I see that the default is configurable through a Kconfig option, currently Fedora is using the Kconfig default value of 0. So I'm wondering if there any known issues with changing the default to 1 ? I was also thinking that maybe an automatic value (SND_HDA_POWER_SAVE_DEFAULT == -1) might be a good idea, which would mean 0 normally except for PCI-ids which are markes with a MOBILE flag in their driver_data in the azx_ids pci_device_id table ? Regards, Hans

2 2

[alsa-devel] ASoC fixes for v4.14
by Mark Brown 01 Nov '17

01 Nov '17

The following changes since commit 0b07194bb55ed836c2cc7c22e866b87a14681984: Linux 4.14-rc7 (2017-10-29 13:58:38 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.14-rc7 for you to fetch changes up to b817d936248b9bcee8282e97fb1dda1b03c903fe: Merge remote-tracking branches 'asoc/fix/topology', 'asoc/fix/adau17x1', 'asoc/fix/rt5514', 'asoc/fix/rt5616', 'asoc/fix/rt5659' and 'asoc/fix/rt5663' into tmp (2017-11-01 13:36:16 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.14 A bunch of fixes here, mostly device specific ones (the biggest one being the revert of the hotword support for rt5514), with a couple of core fixes for potential issues with corrupted or otherwise invalid topology files. ---------------------------------------------------------------- Bard Liao (3): ASoC: rt5659: register power bit of LOUT Amp ASoC: rt5659: connect LOUT Amp with Charge Pump ASoC: rt5616: fix 0x91 default value Christophe JAILLET (2): ASoC: topology: Fix a potential NULL pointer dereference in 'soc_tplg_dapm_widget_denum_create()' ASoC: topology: Fix a potential memory leak in 'soc_tplg_dapm_widget_denum_create()' Hsin-Yu Chao (2): ASoC: rt5514-spi: check irq status to schedule data copy ASoC: rt5514: Revert Hotword Model control Mark Brown (1): Merge remote-tracking branches 'asoc/fix/topology', 'asoc/fix/adau17x1', 'asoc/fix/rt5514', 'asoc/fix/rt5616', 'asoc/fix/rt5659' and 'asoc/fix/rt5663' into tmp Ricard Wanderlof (1): ASoC: adau17x1: Workaround for noise bug in ADC oder_chiou(a)realtek.com (1): ASoC: rt5663: Change the dev getting function in rt5663_irq sound/soc/codecs/adau17x1.c | 24 ++++++++++++++++- sound/soc/codecs/adau17x1.h | 2 ++ sound/soc/codecs/rt5514-spi.c | 17 ++++++++++-- sound/soc/codecs/rt5514-spi.h | 3 +++ sound/soc/codecs/rt5514.c | 63 ------------------------------------------- sound/soc/codecs/rt5514.h | 3 --- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5659.c | 4 ++- sound/soc/codecs/rt5663.c | 3 ++- sound/soc/soc-topology.c | 5 +++- 10 files changed, 53 insertions(+), 73 deletions(-)

2 1

[alsa-devel] [PATCH] alsa-lib: topology: fix DSP_B mode string
by Liam Girdwood 01 Nov '17

01 Nov '17

Use the correct string for DSP_B mode. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 0f4deb4f..c5e8d465 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -997,7 +997,7 @@ static int get_audio_hw_format(const char *val) if (!strcmp(val, "DSP_A")) return SND_SOC_DAI_FORMAT_DSP_A; - if (!strcmp(val, "LEFT_B")) + if (!strcmp(val, "DSP_B")) return SND_SOC_DAI_FORMAT_DSP_B; if (!strcmp(val, "AC97")) -- 2.11.0

2 1

[alsa-devel] [PATCH 0/2] ASoC: rsnd: remove NULL check from rsnd_mod_name()/rsnd_mod_id()
by Kuninori Morimoto 01 Nov '17

01 Nov '17

Hi Mark These are not important from technical point of view, but measures against statically code checker. Kuninori Morimoto (2): ASoC: rsnd: rsnd_ssi_run_mods() needs to care ssi_parent_mod ASoC: rsnd: remove NULL check from rsnd_mod_name()/rsnd_mod_id() sound/soc/sh/rcar/core.c | 8 -------- sound/soc/sh/rcar/dma.c | 24 ++++++++++++++++++------ sound/soc/sh/rcar/rsnd.h | 6 +++--- sound/soc/sh/rcar/ssi.c | 11 ++++++++--- 4 files changed, 29 insertions(+), 20 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] dmaengine: imx-sdma: advertise all supported slave bus widths
by Lucas Stach 31 Oct '17

31 Oct '17

The commit [1] which changed imx-pcm-dma to derive the pcm_config from the attached dmaengine introduced a regression on the supported sample formats, as imx-sdma doesn't properly advertise all the supported slave bus widths. Fix this to restore full functionality of the imx sound subsystem. [1] 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config") Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- drivers/dma/imx-sdma.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index a67ec1bdc4e0..e374183f19d0 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -1851,8 +1851,12 @@ static int sdma_probe(struct platform_device *pdev) sdma->dma_device.device_prep_dma_cyclic = sdma_prep_dma_cyclic; sdma->dma_device.device_config = sdma_config; sdma->dma_device.device_terminate_all = sdma_disable_channel_with_delay; - sdma->dma_device.src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - sdma->dma_device.dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + sdma->dma_device.src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_1_BYTE); + sdma->dma_device.dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_1_BYTE); sdma->dma_device.directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); sdma->dma_device.residue_granularity = DMA_RESIDUE_GRANULARITY_SEGMENT; sdma->dma_device.device_issue_pending = sdma_issue_pending; -- 2.11.0

4 11

[alsa-devel] [PATCH] ASoC: rt5645: remove unexisting route on new rt5645
by Bard Liao 31 Oct '17

31 Oct '17

"SPOL MIX DAC R1 Switch" and "SPOL MIX SPKVOL R Switch" are only exist in the early version of rt5645. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e34f278..b7eef6a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -432,6 +432,7 @@ struct rt5645_priv { int jack_type; bool en_button_func; bool hp_on; + int v_id; }; static int rt5645_reset(struct snd_soc_codec *codec) @@ -2516,9 +2517,7 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPKVOL L", "Switch", "SPK MIXL" }, { "SPKVOL R", "Switch", "SPK MIXR" }, - { "SPOL MIX", "DAC R1 Switch", "DAC R1" }, { "SPOL MIX", "DAC L1 Switch", "DAC L1" }, - { "SPOL MIX", "SPKVOL R Switch", "SPKVOL R" }, { "SPOL MIX", "SPKVOL L Switch", "SPKVOL L" }, { "SPOR MIX", "DAC R1 Switch", "DAC R1" }, { "SPOR MIX", "SPKVOL R Switch", "SPKVOL R" }, @@ -2707,6 +2706,11 @@ static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { { "DAC R2 Mux", "IF1 DAC", "RT5645 IF1 DAC2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5645_old_dapm_routes[] = { + { "SPOL MIX", "DAC R1 Switch", "DAC R1" }, + { "SPOL MIX", "SPKVOL R Switch", "SPKVOL R" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -3363,6 +3367,11 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); + if (rt5645->v_id < 3) { + snd_soc_dapm_add_routes(dapm, + rt5645_old_dapm_routes, + ARRAY_SIZE(rt5645_old_dapm_routes)); + } break; case CODEC_TYPE_RT5650: snd_soc_dapm_new_controls(dapm, @@ -3803,6 +3812,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_write(rt5645->regmap, RT5645_RESET, 0); + regmap_read(regmap, RT5645_VENDOR_ID, &val); + rt5645->v_id = val & 0xff; + ret = regmap_register_patch(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/5] ASoC: rsnd: small tidyups for Renesas sound
by Kuninori Morimoto 31 Oct '17

31 Oct '17

Hi Mark These are small tidyups for Renesas sound driver. Kuninori Morimoto (5): ASoC: rsnd: don't use io->mod[] directly ASoC: rsnd: tidyup rsnd_mod_next() for loop method ASoC: rsnd: NULL check is not needed for clk_unprepare() ASoC: rsnd: use snd_pcm_running() in rsnd_io_is_working() ASoC: rsnd: Don't check SSISR::DIRQ when Capture sound/soc/sh/rcar/core.c | 16 ++++++++-------- sound/soc/sh/rcar/ssi.c | 15 +++++++++------ 2 files changed, 17 insertions(+), 14 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH 0/2] Enable TDM slot and fix SSP params for kabylake machine driver
by Harsha Priya 31 Oct '17

31 Oct '17

This patch series includes 2 patches for kabylake machine driver using max98927 and rt5663 codecs. The changes are: 1. Enable 4 tdm slots for IV feedback and 2 tdm slots for playback on max98927 2. Fix setting of SSP parameters to be based on FE and BE dai links. Naveen M (2): ASoC: Intel: Enable tdm slots for max98927 ASoC: Intel: Fix setting of SSP parameters in Kabylake machine driver sound/soc/intel/boards/kbl_rt5663_max98927.c | 55 ++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 3 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: Intel: kbl: fix jack name
by Vinod Koul 31 Oct '17

31 Oct '17

Commit d1c4cb447a7e ("ASoC: Intel: Skylake: Fix jack name format substitution") added Jack name but erroneously added a space as well, so remove the space in Jack name. Fixes: d1c4cb447a7e ("ASoC: Intel: Skylake: Fix jack name format substitution") Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 9cd0769ccd34..69ab55956492 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -605,7 +605,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { codec = pcm->codec_dai->codec; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &ctx->kabylake_hdmi[i], NULL, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH] sound: Retire OSS
by Takashi Iwai 31 Oct '17

31 Oct '17

Hi, below is a snippet of the patch I'm going to apply for finally dropping the OSS code. It's been already disabled for a few kernel versions, and we didn't get any complains. So, as discussed in the last audio mini-summit, we go forward now. Since the whole patch just removing a bunch of files is too big to post (and useless), I attach only a part of it: namely, the changelog, the diffstat, and the files that are modified (but not deleted). thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] sound: Retire OSS Since no complaints have been raised after disabling the build of OSS (Open Sound System) by the commit 31cbee6a5611 ("sound: Disable the build of OSS drivers"), let's finally drop the whole code and documentation. Some glue codes are still left intact since sound/oss/dmasound stuff remains -- which is an independent implementation solely for m68k, and it's not covered by ALSA yet. Also, a couple of API header files (linux/sound.h and linux/soundcard.h) are kept remaining as well, since the OSS API itself is still supported by ALSA OSS emulation, and applications can refer to these. Where we're at it, some help texts in the top-level Kconfig are adjusted, too (who still needs to specify I/O port in kbuild nowadays?). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/oss/ALS | 66 - Documentation/sound/oss/AudioExcelDSP16 | 101 - Documentation/sound/oss/CMI8330 | 152 -- Documentation/sound/oss/ESS | 34 - Documentation/sound/oss/ESS1868 | 55 - Documentation/sound/oss/Introduction | 459 ----- Documentation/sound/oss/MultiSound | 1137 ----------- Documentation/sound/oss/OPL3 | 6 - Documentation/sound/oss/Opti | 218 -- Documentation/sound/oss/PAS16 | 162 -- Documentation/sound/oss/PSS | 41 - Documentation/sound/oss/PSS-updates | 88 - Documentation/sound/oss/README.OSS | 1455 ------------- Documentation/sound/oss/README.modules | 106 - Documentation/sound/oss/README.ymfsb | 107 - Documentation/sound/oss/SoundPro | 105 - Documentation/sound/oss/Soundblaster | 53 - Documentation/sound/oss/Tropez+ | 26 - Documentation/sound/oss/VIBRA16 | 80 - Documentation/sound/oss/WaveArtist | 170 -- Documentation/sound/oss/btaudio | 92 - Documentation/sound/oss/mwave | 185 -- Documentation/sound/oss/oss-parameters.txt | 51 - Documentation/sound/oss/ultrasound | 30 - MAINTAINERS | 11 - sound/Kconfig | 33 +- sound/oss/CHANGELOG | 369 ---- sound/oss/Kconfig | 533 ----- sound/oss/Makefile | 107 - sound/oss/README.FIRST | 6 - sound/oss/ad1848.c | 3062 ---------------------------- sound/oss/ad1848.h | 24 - sound/oss/ad1848_mixer.h | 253 --- sound/oss/aedsp16.c | 1373 ------------- sound/oss/audio.c | 985 --------- sound/oss/bin2hex.c | 39 - sound/oss/coproc.h | 12 - sound/oss/dev_table.c | 256 --- sound/oss/dev_table.h | 390 ---- sound/oss/dmabuf.c | 1268 ------------ sound/oss/hex2hex.c | 101 - sound/oss/kahlua.c | 229 --- sound/oss/midi_ctrl.h | 22 - sound/oss/midi_synth.c | 712 ------- sound/oss/midi_synth.h | 47 - sound/oss/midibuf.c | 427 ---- sound/oss/mpu401.c | 1804 ---------------- sound/oss/mpu401.h | 11 - sound/oss/msnd.c | 413 ---- sound/oss/msnd.h | 278 --- sound/oss/msnd_classic.c | 3 - sound/oss/msnd_classic.h | 185 -- sound/oss/msnd_pinnacle.c | 1941 ------------------ sound/oss/msnd_pinnacle.h | 246 --- sound/oss/opl3.c | 1255 ------------ sound/oss/opl3_hw.h | 246 --- sound/oss/os.h | 45 - sound/oss/pas2.h | 20 - sound/oss/pas2_card.c | 458 ----- sound/oss/pas2_midi.c | 262 --- sound/oss/pas2_mixer.c | 327 --- sound/oss/pas2_pcm.c | 419 ---- sound/oss/pss.c | 1270 ------------ sound/oss/sb.h | 185 -- sound/oss/sb_audio.c | 1097 ---------- sound/oss/sb_card.c | 354 ---- sound/oss/sb_card.h | 149 -- sound/oss/sb_common.c | 1287 ------------ sound/oss/sb_ess.c | 1822 ----------------- sound/oss/sb_ess.h | 34 - sound/oss/sb_midi.c | 206 -- sound/oss/sb_mixer.c | 770 ------- sound/oss/sb_mixer.h | 105 - sound/oss/sequencer.c | 1661 --------------- sound/oss/sleep.h | 18 - sound/oss/sound_calls.h | 87 - sound/oss/sound_config.h | 144 -- sound/oss/sound_firmware.h | 29 - sound/oss/sound_timer.c | 327 --- sound/oss/soundcard.c | 733 ------- sound/oss/soundvers.h | 2 - sound/oss/swarm_cs4297a.c | 2781 ------------------------- sound/oss/sys_timer.c | 280 --- sound/oss/trix.c | 525 ----- sound/oss/tuning.h | 23 - sound/oss/uart401.c | 477 ----- sound/oss/uart6850.c | 361 ---- sound/oss/ulaw.h | 69 - sound/oss/v_midi.c | 290 --- sound/oss/v_midi.h | 14 - sound/oss/vidc.c | 557 ----- sound/oss/vidc.h | 63 - sound/oss/vidc_fill.S | 218 -- sound/oss/waveartist.c | 2043 ------------------- sound/oss/waveartist.h | 92 - 95 files changed, 1 insertion(+), 41223 deletions(-) delete mode 100644 Documentation/sound/oss/ALS delete mode 100644 Documentation/sound/oss/AudioExcelDSP16 delete mode 100644 Documentation/sound/oss/CMI8330 delete mode 100644 Documentation/sound/oss/ESS delete mode 100644 Documentation/sound/oss/ESS1868 delete mode 100644 Documentation/sound/oss/Introduction delete mode 100644 Documentation/sound/oss/MultiSound delete mode 100644 Documentation/sound/oss/OPL3 delete mode 100644 Documentation/sound/oss/Opti delete mode 100644 Documentation/sound/oss/PAS16 delete mode 100644 Documentation/sound/oss/PSS delete mode 100644 Documentation/sound/oss/PSS-updates delete mode 100644 Documentation/sound/oss/README.OSS delete mode 100644 Documentation/sound/oss/README.modules delete mode 100644 Documentation/sound/oss/README.ymfsb delete mode 100644 Documentation/sound/oss/SoundPro delete mode 100644 Documentation/sound/oss/Soundblaster delete mode 100644 Documentation/sound/oss/Tropez+ delete mode 100644 Documentation/sound/oss/VIBRA16 delete mode 100644 Documentation/sound/oss/WaveArtist delete mode 100644 Documentation/sound/oss/btaudio delete mode 100644 Documentation/sound/oss/mwave delete mode 100644 Documentation/sound/oss/oss-parameters.txt delete mode 100644 Documentation/sound/oss/ultrasound delete mode 100644 sound/oss/CHANGELOG delete mode 100644 sound/oss/Kconfig delete mode 100644 sound/oss/Makefile delete mode 100644 sound/oss/README.FIRST delete mode 100644 sound/oss/ad1848.c delete mode 100644 sound/oss/ad1848.h delete mode 100644 sound/oss/ad1848_mixer.h delete mode 100644 sound/oss/aedsp16.c delete mode 100644 sound/oss/audio.c delete mode 100644 sound/oss/bin2hex.c delete mode 100644 sound/oss/coproc.h delete mode 100644 sound/oss/dev_table.c delete mode 100644 sound/oss/dev_table.h delete mode 100644 sound/oss/dmabuf.c delete mode 100644 sound/oss/hex2hex.c delete mode 100644 sound/oss/kahlua.c delete mode 100644 sound/oss/midi_ctrl.h delete mode 100644 sound/oss/midi_synth.c delete mode 100644 sound/oss/midi_synth.h delete mode 100644 sound/oss/midibuf.c delete mode 100644 sound/oss/mpu401.c delete mode 100644 sound/oss/mpu401.h delete mode 100644 sound/oss/msnd.c delete mode 100644 sound/oss/msnd.h delete mode 100644 sound/oss/msnd_classic.c delete mode 100644 sound/oss/msnd_classic.h delete mode 100644 sound/oss/msnd_pinnacle.c delete mode 100644 sound/oss/msnd_pinnacle.h delete mode 100644 sound/oss/opl3.c delete mode 100644 sound/oss/opl3_hw.h delete mode 100644 sound/oss/os.h delete mode 100644 sound/oss/pas2.h delete mode 100644 sound/oss/pas2_card.c delete mode 100644 sound/oss/pas2_midi.c delete mode 100644 sound/oss/pas2_mixer.c delete mode 100644 sound/oss/pas2_pcm.c delete mode 100644 sound/oss/pss.c delete mode 100644 sound/oss/sb.h delete mode 100644 sound/oss/sb_audio.c delete mode 100644 sound/oss/sb_card.c delete mode 100644 sound/oss/sb_card.h delete mode 100644 sound/oss/sb_common.c delete mode 100644 sound/oss/sb_ess.c delete mode 100644 sound/oss/sb_ess.h delete mode 100644 sound/oss/sb_midi.c delete mode 100644 sound/oss/sb_mixer.c delete mode 100644 sound/oss/sb_mixer.h delete mode 100644 sound/oss/sequencer.c delete mode 100644 sound/oss/sleep.h delete mode 100644 sound/oss/sound_calls.h delete mode 100644 sound/oss/sound_config.h delete mode 100644 sound/oss/sound_firmware.h delete mode 100644 sound/oss/sound_timer.c delete mode 100644 sound/oss/soundcard.c delete mode 100644 sound/oss/soundvers.h delete mode 100644 sound/oss/swarm_cs4297a.c delete mode 100644 sound/oss/sys_timer.c delete mode 100644 sound/oss/trix.c delete mode 100644 sound/oss/tuning.h delete mode 100644 sound/oss/uart401.c delete mode 100644 sound/oss/uart6850.c delete mode 100644 sound/oss/ulaw.h delete mode 100644 sound/oss/v_midi.c delete mode 100644 sound/oss/v_midi.h delete mode 100644 sound/oss/vidc.c delete mode 100644 sound/oss/vidc.h delete mode 100644 sound/oss/vidc_fill.S delete mode 100644 sound/oss/waveartist.c delete mode 100644 sound/oss/waveartist.h diff --git a/MAINTAINERS b/MAINTAINERS index 2d3d750b19c0..01f0b76dcc06 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -527,11 +527,6 @@ W: http://ez.analog.com/community/linux-device-drivers S: Supported F: drivers/input/misc/adxl34x.c -AEDSP16 DRIVER -M: Riccardo Facchetti <fizban(a)tin.it> -S: Maintained -F: sound/oss/aedsp16.c - AF9013 MEDIA DRIVER M: Antti Palosaari <crope(a)iki.fi> L: linux-media(a)vger.kernel.org @@ -9199,12 +9194,6 @@ F: include/linux/dt-bindings/mux/ F: include/linux/mux/ F: drivers/mux/ -MULTISOUND SOUND DRIVER -M: Andrew Veliath <andrewtv(a)usa.net> -S: Maintained -F: Documentation/sound/oss/MultiSound -F: sound/oss/msnd* - MULTITECH MULTIPORT CARD (ISICOM) S: Orphan F: drivers/tty/isicom.c diff --git a/sound/Kconfig b/sound/Kconfig index d7d2aac9542e..b6d047985905 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -3,25 +3,7 @@ menuconfig SOUND depends on HAS_IOMEM help If you have a sound card in your computer, i.e. if it can say more - than an occasional beep, say Y. Be sure to have all the information - about your sound card and its configuration down (I/O port, - interrupt and DMA channel), because you will be asked for it. - - You want to read the Sound-HOWTO, available from - <http://www.tldp.org/docs.html#howto>. General information about - the modular sound system is contained in the files - <file:Documentation/sound/oss/Introduction>. The file - <file:Documentation/sound/oss/README.OSS> contains some slightly - outdated but still useful information as well. Newer sound - driver documentation is found in <file:Documentation/sound/alsa/*>. - - If you have a PnP sound card and you want to configure it at boot - time using the ISA PnP tools (read - <http://www.roestock.demon.co.uk/isapnptools/>), then you need to - compile the sound card support as a module and load that module - after the PnP configuration is finished. To do this, choose M here - and read <file:Documentation/sound/oss/README.modules>; the module - will be called soundcore. + than an occasional beep, say Y. if SOUND @@ -114,19 +96,6 @@ source "sound/synth/Kconfig" endif # SND -menuconfig SOUND_PRIME - tristate "Open Sound System (DEPRECATED)" - select SOUND_OSS_CORE - depends on BROKEN - help - Say 'Y' or 'M' to enable Open Sound System drivers. - -if SOUND_PRIME - -source "sound/oss/Kconfig" - -endif # SOUND_PRIME - endif # !UML endif # SOUND -- 2.14.2

2 4

Re: [alsa-devel] possible deadlock in snd_seq_deliver_event
by Dmitry Vyukov 31 Oct '17

31 Oct '17

On Fri, Oct 27, 2017 at 10:09 AM, syzbot <bot+7feb8de6b4d6bf810cf098bef942cc387e79d0ad(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzkaller hit the following crash on > 2bd6bf03f4c1c59381d62c61d03f6cc3fe71f66e > git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-next.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > ============================================ > WARNING: possible recursive locking detected > 4.14.0-rc1+ #88 Not tainted > -------------------------------------------- > syzkaller883997/2981 is trying to acquire lock: > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] deliver_to_subscribers > sound/core/seq/seq_clientmgr.c:666 [inline] > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > > but task is already holding lock: > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] deliver_to_subscribers > sound/core/seq/seq_clientmgr.c:666 [inline] > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > > other info that might help us debug this: > Possible unsafe locking scenario: > > CPU0 > ---- > lock(&grp->list_mutex); > lock(&grp->list_mutex); > > *** DEADLOCK *** > > May be due to missing lock nesting notation > > 2 locks held by syzkaller883997/2981: > #0: (register_mutex#4){+.+.}, at: [<ffffffff83d60ada>] > odev_release+0x4a/0x70 sound/core/seq/oss/seq_oss.c:152 > #1: (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] > #1: (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > > stack backtrace: > CPU: 1 PID: 2981 Comm: syzkaller883997 Not tainted 4.14.0-rc1+ #88 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS > Google 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:16 [inline] > dump_stack+0x194/0x257 lib/dump_stack.c:52 > print_deadlock_bug kernel/locking/lockdep.c:1797 [inline] > check_deadlock kernel/locking/lockdep.c:1844 [inline] > validate_chain kernel/locking/lockdep.c:2453 [inline] > __lock_acquire+0x1232/0x4620 kernel/locking/lockdep.c:3498 > lock_acquire+0x1d5/0x580 kernel/locking/lockdep.c:4002 > down_read+0x96/0x150 kernel/locking/rwsem.c:23 > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 > snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 > sound/core/seq/seq_clientmgr.c:621 > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] > snd_seq_deliver_event+0x318/0x790 sound/core/seq/seq_clientmgr.c:807 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 > snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 > sound/core/seq/seq_clientmgr.c:621 > snd_seq_deliver_event+0x12c/0x790 sound/core/seq/seq_clientmgr.c:818 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > snd_seq_oss_dispatch sound/core/seq/oss/seq_oss_device.h:150 [inline] > snd_seq_oss_midi_reset+0x44b/0x700 sound/core/seq/oss/seq_oss_midi.c:481 > snd_seq_oss_synth_reset+0x398/0x980 sound/core/seq/oss/seq_oss_synth.c:416 > snd_seq_oss_reset+0x6c/0x260 sound/core/seq/oss/seq_oss_init.c:448 > snd_seq_oss_release+0x71/0x120 sound/core/seq/oss/seq_oss_init.c:425 > odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:153 > __fput+0x333/0x7f0 fs/file_table.c:210 > ____fput+0x15/0x20 fs/file_table.c:244 > task_work_run+0x199/0x270 kernel/task_work.c:112 > exit_task_work include/linux/task_work.h:21 [inline] > do_exit+0xa52/0x1b40 kernel/exit.c:865 > do_group_exit+0x149/0x400 kernel/exit.c:968 > SYSC_exit_group kernel/exit.c:979 [inline] > SyS_exit_group+0x1d/0x20 kernel/exit.c:977 > entry_SYSCALL_64_fastpath+0x1f/0xbe > RIP: 0033:0x442c58 > RSP: 002b:00007ffd15d4f8d8 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 > RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000442c58 > RDX: 0000000000000000 RSI: 000000000000003c RDI: 0000000000000000 > RBP: 0000000000000082 R08: 00000000000000e7 R09: ffffffffffffffd0 > R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000401ca0 > R13: 0000000000401d30 R14 I've just re-reproduced this on upstream 15f859ae5c43c7f0a064ed92d33f7a5bc5de6de0 (Oct 26): ============================================ WARNING: possible recursive locking detected 4.14.0-rc6+ #10 Not tainted -------------------------------------------- a.out/3062 is trying to acquire lock: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 but task is already holding lock: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(&grp->list_mutex); lock(&grp->list_mutex); *** DEADLOCK *** May be due to missing lock nesting notation 2 locks held by a.out/3062: #0: (register_mutex#4){+.+.}, at: [<ffffffff83d3b5da>] odev_release+0x4a/0x70 sound/core/seq/oss/seq_oss.c:152 #1: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] #1: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 stack backtrace: CPU: 0 PID: 3062 Comm: a.out Not tainted 4.14.0-rc6+ #10 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:16 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:52 print_deadlock_bug kernel/locking/lockdep.c:1797 [inline] check_deadlock kernel/locking/lockdep.c:1844 [inline] validate_chain kernel/locking/lockdep.c:2445 [inline] __lock_acquire+0xed5/0x3d50 kernel/locking/lockdep.c:3490 lock_acquire+0x1d5/0x580 kernel/locking/lockdep.c:3994 down_read+0x96/0x150 kernel/locking/rwsem.c:23 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] snd_seq_deliver_event+0x318/0x790 sound/core/seq/seq_clientmgr.c:807 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 snd_seq_deliver_event+0x12c/0x790 sound/core/seq/seq_clientmgr.c:818 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 snd_seq_oss_dispatch sound/core/seq/oss/seq_oss_device.h:150 [inline] snd_seq_oss_midi_reset+0x44b/0x700 sound/core/seq/oss/seq_oss_midi.c:481 snd_seq_oss_synth_reset+0x398/0x980 sound/core/seq/oss/seq_oss_synth.c:416 snd_seq_oss_reset+0x6c/0x260 sound/core/seq/oss/seq_oss_init.c:448 snd_seq_oss_release+0x71/0x120 sound/core/seq/oss/seq_oss_init.c:425 odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:153 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:112 exit_task_work include/linux/task_work.h:21 [inline] do_exit+0x9b5/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 SYSC_exit_group kernel/exit.c:979 [inline] SyS_exit_group+0x1d/0x20 kernel/exit.c:977 entry_SYSCALL_64_fastpath+0x1f/0xbe RIP: 0033:0x437b99 RSP: 002b:00007ffe1b782328 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000437b99 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000082 R08: 000000000000003c R09: 00000000000000e7 R10: ffffffffffffffc0 R11: 0000000000000246 R12: 0000000000000000 R13: 0000000000401c00 R14: 0000000000401c90 R15: 0000000000000000 > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > > syzbot will keep track of this bug report. > Once a fix for this bug is committed, please reply to this email with: > #syz fix: exact-commit-title > To mark this as a duplicate of another syzbot report, please reply with: > #syz dup: exact-subject-of-another-report > If it's a one-off invalid bug report, please reply with: > #syz invalid > Note: if the crash happens again, it will cause creation of a new bug > report. > > -- > You received this message because you are subscribed to the Google Groups > "syzkaller-bugs" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to syzkaller-bugs+unsubscribe(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/syzkaller-bugs/001a114089d614a0f4055c82d0…. > For more options, visit https://groups.google.com/d/optout.

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[alsa-devel] sound tree breaks make clean
by Daniel Vetter 31 Oct '17

31 Oct '17

Hi all, The following commit commit 60d4981ccf3603669c8de304619048ad5da707bd Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Oct 24 09:15:23 2017 +0200 sound: Retire OSS breaks $ make defconfig && make clean (just make clean in an already clean tree seems to work). We've taken the sound tree out of drm-tip and our CI for now to work around this. Thanks, Daniel -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

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[alsa-devel] [PATCH v2] ASoC: wm8741: Add digital mute callback
by Sergej Sawazki 31 Oct '17

31 Oct '17

Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Changes in v2: * Refactored debug message. sound/soc/codecs/wm8741.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b8c1940..5571882 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -351,6 +351,17 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + dev_dbg(codec->dev, "wm8741_mute: mute = %d\n", mute); + + snd_soc_update_bits(codec, WM8741_VOLUME_CONTROL, + WM8741_SOFT_MASK, !!mute << WM8741_SOFT_SHIFT); + return 0; +} + #define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ @@ -364,6 +375,7 @@ static const struct snd_soc_dai_ops wm8741_dai_ops = { .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, .set_fmt = wm8741_set_dai_fmt, + .digital_mute = wm8741_mute, }; static struct snd_soc_dai_driver wm8741_dai = { -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC fixes for v4.14
by Mark Brown 31 Oct '17

31 Oct '17

The following changes since commit bb176f67090ca54869fc1262c913aa69d2ede070: Linux 4.14-rc6 (2017-10-23 06:49:47 -0400) are available in the git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.14-rc6 for you to fetch changes up to 0bebd2f1bfcbf954a93b65d4b2848488fd049c78: Merge remote-tracking branches 'asoc/fix/topology' and 'asoc/fix/wm8998' into asoc-linus (2017-10-26 09:45:52 +0200) ---------------------------------------------------------------- ASoC: Fixes for v4.14 I've been quite lax in sending these due to conference season but here's a fairly large collection of ASoC updates. The one thing that's not device specific is Takashi's fix for races between delayed work and PCM destruction, otherwise everything is specific to an individual device. ---------------------------------------------------------------- Bard Liao (3): ASoC: rt5659: register power bit of LOUT Amp ASoC: rt5659: connect LOUT Amp with Charge Pump ASoC: rt5616: fix 0x91 default value Charles Keepax (1): ASoC: wm8998: Correct handling of input muxes Chintan Patel (1): ASoC: Intel: Skylake: Fix jack name format substitution Christophe JAILLET (2): ASoC: topology: Fix a potential NULL pointer dereference in 'soc_tplg_dapm_widget_denum_create()' ASoC: topology: Fix a potential memory leak in 'soc_tplg_dapm_widget_denum_create()' Christophe Jaillet (2): ASoC: samsung: i2s: Fix error handling path in i2s_set_sysclk() ASoC: davinci-mcasp: Fix an error handling path in 'davinci_mcasp_probe()' Colin Ian King (1): ASoC: Intel: Skylake: fix swapped order of function arguments dir and pin_index Hsin-Yu Chao (2): ASoC: rt5514-spi: check irq status to schedule data copy ASoC: rt5514: Revert Hotword Model control Jean-François Têtu (2): ASoC: msm8916-wcd-digital: fix RX2 MIX1 and RX3 MIX1 ASoC: codecs: msm8916-wcd-analog: fix micbias level John Keeping (1): ASoC: rockchip: i2s: fix unbalanced clk_disable Kuninori Morimoto (2): ASoC: rsnd: fix ADG flags ASoC: rcar: skip disabled-SSI nodes Marek Szyprowski (1): ASoC: samsung: i2s: disable secondary DAI until it gets fixed Mark Brown (8): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/msm8916' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/rockchip' into asoc-linus Merge remote-tracking branches 'asoc/fix/adau17x1', 'asoc/fix/davinci' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5514', 'asoc/fix/rt5616', 'asoc/fix/rt5659', 'asoc/fix/rt5663', 'asoc/fix/samsung' and 'asoc/fix/stm32' into asoc-linus Merge remote-tracking branches 'asoc/fix/topology' and 'asoc/fix/wm8998' into asoc-linus Olivier Moysan (3): ASoC: stm32: sai: fix stop management in isr ASoC: stm32: sai: Fix DMA burst size ASoC: stm32: sai: Fix get reset controller Pierre-Louis Bossart (2): ASoC: max98090: reduce verbosity on PLL unlock ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code Ricard Wanderlof (1): ASoC: adau17x1: Workaround for noise bug in ADC Takashi Iwai (1): ASoC: pcm: Sync delayed work before releasing resources oder_chiou(a)realtek.com (1): ASoC: rt5663: Change the dev getting function in rt5663_irq sound/soc/codecs/adau17x1.c | 24 ++++++- sound/soc/codecs/adau17x1.h | 2 + sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 7 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 +- sound/soc/codecs/rt5514-spi.c | 17 ++++- sound/soc/codecs/rt5514-spi.h | 3 + sound/soc/codecs/rt5514.c | 63 ---------------- sound/soc/codecs/rt5514.h | 3 - sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5659.c | 4 +- sound/soc/codecs/rt5663.c | 3 +- sound/soc/codecs/wm8998.c | 84 +++++----------------- sound/soc/davinci/davinci-mcasp.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 + sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/samsung/i2s.c | 6 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/ssi.c | 5 +- sound/soc/soc-pcm.c | 13 +++- sound/soc/soc-topology.c | 5 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 33 +++++---- 25 files changed, 134 insertions(+), 169 deletions(-)

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[alsa-devel] "PM / QoS: Fix device resume latency PM QoS" breaks sound
by Chris Clayton 31 Oct '17

31 Oct '17

Hi, I pulled the latestchanges from Linus' tree this evening and have found that with the new kernel, sound is not working on my laptop. More precisely, the built-in speakers don't produce any sound. Sound does work when I use ear-plugs in the headphone socket. It also works via a bluetooth speaker. I've bisected the problem and ended up at: 0cc2b4e5a020fc7f4d1795741c116c983e9467d7 is the first bad commit commit 0cc2b4e5a020fc7f4d1795741c116c983e9467d7 Author: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Date: Tue Oct 24 15:20:45 2017 +0200 PM / QoS: Fix device resume latency PM QoS The special value of 0 for device resume latency PM QoS means "no restriction", but there are two problems with that. First, device resume latency PM QoS requests with 0 as the value are always put in front of requests with positive values in the priority lists used internally by the PM QoS framework, causing 0 to be chosen as an effective constraint value. However, that 0 is then interpreted as "no restriction" effectively overriding the other requests with specific restrictions which is incorrect. Second, the users of device resume latency PM QoS have no way to specify that *any* resume latency at all should be avoided, which is an artificial limitation in general. To address these issues, modify device resume latency PM QoS to use S32_MAX as the "no constraint" value and 0 as the "no latency at all" one and rework its users (the cpuidle menu governor, the genpd QoS governor and the runtime PM framework) to follow these changes. Also add a special "n/a" value to the corresponding user space I/F to allow user space to indicate that it cannot accept any resume latencies at all for the given device. Fixes: 85dc0b8a4019 (PM / QoS: Make it possible to expose PM QoS latency constraints) Link: https://bugzilla.kernel.org/show_bug.cgi?id=197323 Reported-by: Reinette Chatre <reinette.chatre(a)intel.com> Tested-by: Reinette Chatre <reinette.chatre(a)intel.com> Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Acked-by: Alex Shi <alex.shi(a)linaro.org> Cc: All applicable <stable(a)vger.kernel.org> :040000 040000 f0c128ec799bb9894cfc5c341f88ad7bdfb15bac 9a2e8171ca47f864bd534cd9c160cce58449a889 M Documentation :040000 040000 0028ffec81675e686bdd621c0445d3e814d7980c 29db53c6356a6fed9c8bdbc2d6bc7bd56a96e529 M drivers :040000 040000 2e66b79bd2ffb4fcb00f04a69a0afe5c80d1d3f3 dd6d8e90b59389cd2bd8a0c92716d79d2eeb8268 M include With that change reverted, the speakers emit sound again. The audio devices identified by "lspci -vv" are as follows: 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) Subsystem: CLEVO/KAPOK Computer Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at f7f14000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit- Address: 00000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) Subsystem: CLEVO/KAPOK Computer 8 Series/C220 Series Chipset High Definition Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f7f10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel Let me know if I can provide any additional diagnostics, but please cc me because I'm not subscribed. Thanks Chris

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[alsa-devel] [asoc:topic/intel 41/42] sound/soc/intel/boards/bytcr_rt5651.c:325:2: error: implicit declaration of function 'rt5651_set_jack_detect'
by kbuild test robot 30 Oct '17

30 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 2fe30129b0a6a8ddf71c46c8fb4524cb89739bc0 commit: d9f8f9b2f3e2b95875cd59ae5c3276e4662e7513 [41/42] ASoC: intel: byt: Add headset jack config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout d9f8f9b2f3e2b95875cd59ae5c3276e4662e7513 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/boards/bytcr_rt5651.c: In function 'byt_rt5651_init': >> sound/soc/intel/boards/bytcr_rt5651.c:325:2: error: implicit declaration of function 'rt5651_set_jack_detect' [-Werror=implicit-function-declaration] rt5651_set_jack_detect(codec, &priv->jack); ^~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/rt5651_set_jack_detect +325 sound/soc/intel/boards/bytcr_rt5651.c 255 256 static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) 257 { 258 struct snd_soc_card *card = runtime->card; 259 struct snd_soc_codec *codec = runtime->codec; 260 struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); 261 const struct snd_soc_dapm_route *custom_map; 262 int num_routes; 263 int ret; 264 265 card->dapm.idle_bias_off = true; 266 267 switch (BYT_RT5651_MAP(byt_rt5651_quirk)) { 268 case BYT_RT5651_IN1_MAP: 269 custom_map = byt_rt5651_intmic_in1_map; 270 num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); 271 break; 272 case BYT_RT5651_IN2_MAP: 273 custom_map = byt_rt5651_intmic_in2_map; 274 num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); 275 break; 276 default: 277 custom_map = byt_rt5651_intmic_dmic_map; 278 num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); 279 } 280 ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); 281 if (ret) 282 return ret; 283 284 ret = snd_soc_add_card_controls(card, byt_rt5651_controls, 285 ARRAY_SIZE(byt_rt5651_controls)); 286 if (ret) { 287 dev_err(card->dev, "unable to add card controls\n"); 288 return ret; 289 } 290 snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); 291 snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); 292 293 if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) { 294 /* 295 * The firmware might enable the clock at 296 * boot (this information may or may not 297 * be reflected in the enable clock register). 298 * To change the rate we must disable the clock 299 * first to cover these cases. Due to common 300 * clock framework restrictions that do not allow 301 * to disable a clock that has not been enabled, 302 * we need to enable the clock first. 303 */ 304 ret = clk_prepare_enable(priv->mclk); 305 if (!ret) 306 clk_disable_unprepare(priv->mclk); 307 308 if (byt_rt5651_quirk & BYT_RT5651_MCLK_25MHZ) 309 ret = clk_set_rate(priv->mclk, 25000000); 310 else 311 ret = clk_set_rate(priv->mclk, 19200000); 312 313 if (ret) 314 dev_err(card->dev, "unable to set MCLK rate\n"); 315 } 316 317 ret = snd_soc_card_jack_new(runtime->card, "Headset", 318 SND_JACK_HEADSET, &priv->jack, 319 bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); 320 if (ret) { 321 dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); 322 return ret; 323 } 324 > 325 rt5651_set_jack_detect(codec, &priv->jack); 326 327 return ret; 328 } 329 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] timestamp update when no hw delta
by Henrik Eriksson 30 Oct '17

30 Oct '17

Hi I am looking into commit 3179f62001880e588e229db3006a59ad87b7792a ("ALSA: core: add .get_time_info"), which, if I understand the code correctly, changed the behaviour of timestamps. Before this change, snd_pcm_update_hw_ptr0() just returned if the no_delta_check showed that the hw_ptr had not changed. With this change it will update the timestamps before returning. Was this intended, and if so does anyone have any insights to why? The original behaviour to not update the timestamps when the hw position has not changed seems more correct to me, but I am new to this code. Regards, /henrik

2 2

[alsa-devel] [PATCH 0/3] ASoC: wm8741: Varios enhancements
by Sergej Sawazki 30 Oct '17

30 Oct '17

This patch series contains the following 3 patches for the WM8741 codec driver. 1) Adds digital mute callback. 2) Sets oversampling rate mode (OSR), required for sampling rates > 48kHz. 3) Sets MCLK/LRCLK sampling rate ratio (SR), avoids relying on auto-detection. Sergej Sawazki (3): ASoC: wm8741: Add digital mute callback ASoC: wm8741: Set OSR mode in hw_params() ASoC: wm8741: Set SR mode in hw_params() sound/soc/codecs/wm8741.c | 49 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 8 deletions(-) -- 2.7.4

3 15

[alsa-devel] [PATCH linux-next 1/1] ASoC: rsnd: ssiu: reset SSI_MODE register
by jiada_wang＠mentor.com 30 Oct '17

30 Oct '17

From: Jiada Wang <jiada_wang(a)mentor.com> register SSI_MODE is set when SSI works in TDM Extended or TDM (Ex-)Split mode, but it isn't reset after SSI stops. this will cause issue, if SSI starts to work in other modes which requie SSI_MODE to have different value. This patch resets SSI_MODE register in .stop callback, when last stream calls .stop. Fixes: 186fadc132f0 ("ASoC: rsnd: add TDM Extend Mode support") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d94875..eff9f3b 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -217,6 +217,8 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0); + rsnd_mod_write(mod, SSI_MODE, 0); + return 0; } -- 1.7.9.5

3 3

[alsa-devel] [PATCH] sound: fix kernel-doc build warning
by Randy Dunlap 30 Oct '17

30 Oct '17

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc build error. A symbol that ends with an underscore character ('_') has special meaning in reST (reStructuredText), so add a '*' to prevent this error and to indicate that there are several of these values to choose from. ../sound/core/jack.c:312: ERROR: Unknown target name: "snd_jack_btn". Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/core/jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-414-rc7.orig/sound/core/jack.c +++ lnx-414-rc7/sound/core/jack.c @@ -310,7 +310,7 @@ EXPORT_SYMBOL(snd_jack_set_parent); * @type: Jack report type for this key * @keytype: Input layer key type to be reported * - * Map a SND_JACK_BTN_ button type to an input layer key, allowing + * Map a SND_JACK_BTN_* button type to an input layer key, allowing * reporting of keys on accessories via the jack abstraction. If no * mapping is provided but keys are enabled in the jack type then * BTN_n numeric buttons will be reported.

2 1

[alsa-devel] message size limit
by Jaroslav Kysela 29 Oct '17

29 Oct '17

Hello all, I increased the maximal message size for this alsa-devel list to 200kB (as we discussed with Takashi to allow bigger patches for review). Anyway, if you like to send something big, it's better to upload the data to a public place and provide only URL. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 1

[alsa-devel] [PATCH RESEND 0/2] Enable TDM slot and fix SSP params for kabylake machine driver
by Harsha Priya 27 Oct '17

27 Oct '17

Hi Mark, These are resend of an already acked patch series. I posted it a few weeks ago but there has been update on the base branch. I want to resend these. Kindly review and merge. This patch series includes 2 patches for kabylake machine driver using max98927 and rt5663 codecs. The changes are: 1. Enable 4 tdm slots for IV feedback and 2 tdm slots for playback on max98927 2. Fix setting of SSP parameters to be based on FE and BE dai links. Harsha Priya (2): ASoC: Intel: Enable tdm slots for max98927 ASoC: Intel: Fix setting of SSP parameters in Kabylake machine driver sound/soc/intel/boards/kbl_rt5663_max98927.c | 67 +++++++++++++++++++++++++--- 1 file changed, 60 insertions(+), 7 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH] conf/ucm: rt5651: Fix HiFi file name
by Carlo Caione 26 Oct '17

26 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> The conf HiFi file name is HiFi.conf, fix the name in the main configuration file. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf index 2dcc72cc..60798ce7 100644 --- a/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf +++ b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf @@ -1,7 +1,7 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 SectionUseCase."HiFi" { - File "HiFi" + File "HiFi.conf" Comment "Play HiFi quality Music" } -- 2.14.2

2 1

[alsa-devel] [asoc:for-next 16/17] sound/soc/codecs/rt5651.c:1874:20: error: redefinition of 'rt5651_irq'
by kbuild test robot 26 Oct '17

26 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 2708a8339e928912307fe5cd1ae66d959caa7279 commit: b16351eba5e00bd7957fb16b98a152e5425ceb74 [16/17] Merge remote-tracking branches 'asoc/topic/sgtl5000', 'asoc/topic/stm32', 'asoc/topic/tas571x', 'asoc/topic/tfa9879' and 'asoc/topic/ts3a277e' into asoc-next config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout b16351eba5e00bd7957fb16b98a152e5425ceb74 # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings (new ones prefixed by >>): >> sound/soc/codecs/rt5651.c:1874:20: error: redefinition of 'rt5651_irq' static irqreturn_t rt5651_irq(int irq, void *data) ^ sound/soc/codecs/rt5651.c:1787:20: note: previous definition of 'rt5651_irq' was here static irqreturn_t rt5651_irq(int irq, void *data) ^ >> sound/soc/codecs/rt5651.c:1884:12: error: redefinition of 'rt5651_jack_detect' static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ sound/soc/codecs/rt5651.c:1797:12: note: previous definition of 'rt5651_jack_detect' was here static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ >> sound/soc/codecs/rt5651.c:1920:13: error: redefinition of 'rt5651_jack_detect_work' static void rt5651_jack_detect_work(struct work_struct *work) ^ sound/soc/codecs/rt5651.c:1833:13: note: previous definition of 'rt5651_jack_detect_work' was here static void rt5651_jack_detect_work(struct work_struct *work) ^ >> sound/soc/codecs/rt5651.c:1949:5: error: redefinition of 'rt5651_set_jack_detect' int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ sound/soc/codecs/rt5651.c:1862:5: note: previous definition of 'rt5651_set_jack_detect' was here int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ In file included from include/linux/linkage.h:6:0, from include/linux/kernel.h:6, from include/linux/list.h:8, from include/linux/module.h:9, from sound/soc/codecs/rt5651.c:12: >> include/linux/export.h:67:20: error: redefinition of '__kstrtab_rt5651_set_jack_detect' static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ >> sound/soc/codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:67:20: note: previous definition of '__kstrtab_rt5651_set_jack_detect' was here static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc/codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: error: redefinition of '__ksymtab_rt5651_set_jack_detect' static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ >> sound/soc/codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: note: previous definition of '__ksymtab_rt5651_set_jack_detect' was here static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc/codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ sound/soc/codecs/rt5651.c:1833:13: warning: 'rt5651_jack_detect_work' defined but not used [-Wunused-function] static void rt5651_jack_detect_work(struct work_struct *work) ^ -- sound/soc//codecs/rt5651.c:1874:20: error: redefinition of 'rt5651_irq' static irqreturn_t rt5651_irq(int irq, void *data) ^ sound/soc//codecs/rt5651.c:1787:20: note: previous definition of 'rt5651_irq' was here static irqreturn_t rt5651_irq(int irq, void *data) ^ sound/soc//codecs/rt5651.c:1884:12: error: redefinition of 'rt5651_jack_detect' static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ sound/soc//codecs/rt5651.c:1797:12: note: previous definition of 'rt5651_jack_detect' was here static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ sound/soc//codecs/rt5651.c:1920:13: error: redefinition of 'rt5651_jack_detect_work' static void rt5651_jack_detect_work(struct work_struct *work) ^ sound/soc//codecs/rt5651.c:1833:13: note: previous definition of 'rt5651_jack_detect_work' was here static void rt5651_jack_detect_work(struct work_struct *work) ^ sound/soc//codecs/rt5651.c:1949:5: error: redefinition of 'rt5651_set_jack_detect' int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ sound/soc//codecs/rt5651.c:1862:5: note: previous definition of 'rt5651_set_jack_detect' was here int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ In file included from include/linux/linkage.h:6:0, from include/linux/kernel.h:6, from include/linux/list.h:8, from include/linux/module.h:9, from sound/soc//codecs/rt5651.c:12: >> include/linux/export.h:67:20: error: redefinition of '__kstrtab_rt5651_set_jack_detect' static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:67:20: note: previous definition of '__kstrtab_rt5651_set_jack_detect' was here static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: error: redefinition of '__ksymtab_rt5651_set_jack_detect' static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: note: previous definition of '__ksymtab_rt5651_set_jack_detect' was here static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ sound/soc//codecs/rt5651.c:1833:13: warning: 'rt5651_jack_detect_work' defined but not used [-Wunused-function] static void rt5651_jack_detect_work(struct work_struct *work) ^ vim +/rt5651_irq +1874 sound/soc/codecs/rt5651.c b4435130 Carlo Caione 2017-10-20 1786 80bbe4a3 Carlo Caione 2017-10-20 @1787 static irqreturn_t rt5651_irq(int irq, void *data) 80bbe4a3 Carlo Caione 2017-10-20 1788 { 80bbe4a3 Carlo Caione 2017-10-20 1789 struct rt5651_priv *rt5651 = data; 80bbe4a3 Carlo Caione 2017-10-20 1790 80bbe4a3 Carlo Caione 2017-10-20 1791 queue_delayed_work(system_power_efficient_wq, 80bbe4a3 Carlo Caione 2017-10-20 1792 &rt5651->jack_detect_work, msecs_to_jiffies(250)); 80bbe4a3 Carlo Caione 2017-10-20 1793 80bbe4a3 Carlo Caione 2017-10-20 1794 return IRQ_HANDLED; 80bbe4a3 Carlo Caione 2017-10-20 1795 } 80bbe4a3 Carlo Caione 2017-10-20 1796 80bbe4a3 Carlo Caione 2017-10-20 @1797 static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) 80bbe4a3 Carlo Caione 2017-10-20 1798 { 80bbe4a3 Carlo Caione 2017-10-20 1799 struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); 80bbe4a3 Carlo Caione 2017-10-20 1800 int jack_type; 80bbe4a3 Carlo Caione 2017-10-20 1801 80bbe4a3 Carlo Caione 2017-10-20 1802 if (jack_insert) { 80bbe4a3 Carlo Caione 2017-10-20 1803 snd_soc_dapm_force_enable_pin(dapm, "LDO"); 80bbe4a3 Carlo Caione 2017-10-20 1804 snd_soc_dapm_sync(dapm); 80bbe4a3 Carlo Caione 2017-10-20 1805 80bbe4a3 Carlo Caione 2017-10-20 1806 snd_soc_update_bits(codec, RT5651_MICBIAS, 80bbe4a3 Carlo Caione 2017-10-20 1807 RT5651_MIC1_OVCD_MASK | 80bbe4a3 Carlo Caione 2017-10-20 1808 RT5651_MIC1_OVTH_MASK | 80bbe4a3 Carlo Caione 2017-10-20 1809 RT5651_PWR_CLK12M_MASK | 80bbe4a3 Carlo Caione 2017-10-20 1810 RT5651_PWR_MB_MASK, 80bbe4a3 Carlo Caione 2017-10-20 1811 RT5651_MIC1_OVCD_EN | 80bbe4a3 Carlo Caione 2017-10-20 1812 RT5651_MIC1_OVTH_600UA | b16351eb Mark Brown 2017-10-26 1813 RT5651_PWR_MB_PU | b16351eb Mark Brown 2017-10-26 1814 RT5651_PWR_CLK12M_PU); b16351eb Mark Brown 2017-10-26 1815 msleep(100); b16351eb Mark Brown 2017-10-26 1816 if (snd_soc_read(codec, RT5651_IRQ_CTRL2) & RT5651_MB1_OC_CLR) b16351eb Mark Brown 2017-10-26 1817 jack_type = SND_JACK_HEADPHONE; b16351eb Mark Brown 2017-10-26 1818 else b16351eb Mark Brown 2017-10-26 1819 jack_type = SND_JACK_HEADSET; b16351eb Mark Brown 2017-10-26 1820 snd_soc_update_bits(codec, RT5651_IRQ_CTRL2, b16351eb Mark Brown 2017-10-26 1821 RT5651_MB1_OC_CLR, 0); b16351eb Mark Brown 2017-10-26 1822 } else { /* jack out */ b16351eb Mark Brown 2017-10-26 1823 jack_type = 0; b16351eb Mark Brown 2017-10-26 1824 b16351eb Mark Brown 2017-10-26 1825 snd_soc_update_bits(codec, RT5651_MICBIAS, b16351eb Mark Brown 2017-10-26 1826 RT5651_MIC1_OVCD_MASK, b16351eb Mark Brown 2017-10-26 1827 RT5651_MIC1_OVCD_DIS); b16351eb Mark Brown 2017-10-26 1828 } b16351eb Mark Brown 2017-10-26 1829 b16351eb Mark Brown 2017-10-26 1830 return jack_type; b16351eb Mark Brown 2017-10-26 1831 } b16351eb Mark Brown 2017-10-26 1832 b16351eb Mark Brown 2017-10-26 1833 static void rt5651_jack_detect_work(struct work_struct *work) b16351eb Mark Brown 2017-10-26 1834 { b16351eb Mark Brown 2017-10-26 1835 struct rt5651_priv *rt5651 = b16351eb Mark Brown 2017-10-26 1836 container_of(work, struct rt5651_priv, jack_detect_work.work); b16351eb Mark Brown 2017-10-26 1837 b16351eb Mark Brown 2017-10-26 1838 int report, val = 0; b16351eb Mark Brown 2017-10-26 1839 b16351eb Mark Brown 2017-10-26 1840 if (!rt5651->codec) b16351eb Mark Brown 2017-10-26 1841 return; b16351eb Mark Brown 2017-10-26 1842 b16351eb Mark Brown 2017-10-26 1843 switch (rt5651->pdata.jd_src) { b16351eb Mark Brown 2017-10-26 1844 case RT5651_JD1_1: b16351eb Mark Brown 2017-10-26 1845 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x1000; b16351eb Mark Brown 2017-10-26 1846 break; b16351eb Mark Brown 2017-10-26 1847 case RT5651_JD1_2: b16351eb Mark Brown 2017-10-26 1848 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x2000; b16351eb Mark Brown 2017-10-26 1849 break; b16351eb Mark Brown 2017-10-26 1850 case RT5651_JD2: b16351eb Mark Brown 2017-10-26 1851 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x4000; b16351eb Mark Brown 2017-10-26 1852 break; b16351eb Mark Brown 2017-10-26 1853 default: b16351eb Mark Brown 2017-10-26 1854 break; b16351eb Mark Brown 2017-10-26 1855 } b16351eb Mark Brown 2017-10-26 1856 b16351eb Mark Brown 2017-10-26 1857 report = rt5651_jack_detect(rt5651->codec, !val); b16351eb Mark Brown 2017-10-26 1858 b16351eb Mark Brown 2017-10-26 1859 snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); b16351eb Mark Brown 2017-10-26 1860 } b16351eb Mark Brown 2017-10-26 1861 b16351eb Mark Brown 2017-10-26 1862 int rt5651_set_jack_detect(struct snd_soc_codec *codec, b16351eb Mark Brown 2017-10-26 1863 struct snd_soc_jack *hp_jack) b16351eb Mark Brown 2017-10-26 1864 { b16351eb Mark Brown 2017-10-26 1865 struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); b16351eb Mark Brown 2017-10-26 1866 b16351eb Mark Brown 2017-10-26 1867 rt5651->hp_jack = hp_jack; b16351eb Mark Brown 2017-10-26 1868 rt5651_irq(0, rt5651); 3ae08dc0 Bard Liao 2015-12-23 1869 3ae08dc0 Bard Liao 2015-12-23 1870 return 0; 3ae08dc0 Bard Liao 2015-12-23 1871 } b16351eb Mark Brown 2017-10-26 1872 EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); 3ae08dc0 Bard Liao 2015-12-23 1873 60d5a1a4 Carlo Caione 2017-10-20 @1874 static irqreturn_t rt5651_irq(int irq, void *data) 60d5a1a4 Carlo Caione 2017-10-20 1875 { 60d5a1a4 Carlo Caione 2017-10-20 1876 struct rt5651_priv *rt5651 = data; 60d5a1a4 Carlo Caione 2017-10-20 1877 60d5a1a4 Carlo Caione 2017-10-20 1878 queue_delayed_work(system_power_efficient_wq, 60d5a1a4 Carlo Caione 2017-10-20 1879 &rt5651->jack_detect_work, msecs_to_jiffies(250)); 60d5a1a4 Carlo Caione 2017-10-20 1880 60d5a1a4 Carlo Caione 2017-10-20 1881 return IRQ_HANDLED; 60d5a1a4 Carlo Caione 2017-10-20 1882 } 60d5a1a4 Carlo Caione 2017-10-20 1883 60d5a1a4 Carlo Caione 2017-10-20 @1884 static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) 60d5a1a4 Carlo Caione 2017-10-20 1885 { 60d5a1a4 Carlo Caione 2017-10-20 1886 struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); 60d5a1a4 Carlo Caione 2017-10-20 1887 int jack_type; 60d5a1a4 Carlo Caione 2017-10-20 1888 60d5a1a4 Carlo Caione 2017-10-20 1889 if (jack_insert) { 60d5a1a4 Carlo Caione 2017-10-20 1890 snd_soc_dapm_force_enable_pin(dapm, "LDO"); 60d5a1a4 Carlo Caione 2017-10-20 1891 snd_soc_dapm_sync(dapm); 60d5a1a4 Carlo Caione 2017-10-20 1892 60d5a1a4 Carlo Caione 2017-10-20 1893 snd_soc_update_bits(codec, RT5651_MICBIAS, 60d5a1a4 Carlo Caione 2017-10-20 1894 RT5651_MIC1_OVCD_MASK | 60d5a1a4 Carlo Caione 2017-10-20 1895 RT5651_MIC1_OVTH_MASK | 60d5a1a4 Carlo Caione 2017-10-20 1896 RT5651_PWR_CLK12M_MASK | 60d5a1a4 Carlo Caione 2017-10-20 1897 RT5651_PWR_MB_MASK, 60d5a1a4 Carlo Caione 2017-10-20 1898 RT5651_MIC1_OVCD_EN | 60d5a1a4 Carlo Caione 2017-10-20 1899 RT5651_MIC1_OVTH_600UA | 80bbe4a3 Carlo Caione 2017-10-20 1900 RT5651_PWR_MB_PU | 80bbe4a3 Carlo Caione 2017-10-20 1901 RT5651_PWR_CLK12M_PU); 80bbe4a3 Carlo Caione 2017-10-20 1902 msleep(100); 80bbe4a3 Carlo Caione 2017-10-20 1903 if (snd_soc_read(codec, RT5651_IRQ_CTRL2) & RT5651_MB1_OC_CLR) 80bbe4a3 Carlo Caione 2017-10-20 1904 jack_type = SND_JACK_HEADPHONE; 80bbe4a3 Carlo Caione 2017-10-20 1905 else 80bbe4a3 Carlo Caione 2017-10-20 1906 jack_type = SND_JACK_HEADSET; 80bbe4a3 Carlo Caione 2017-10-20 1907 snd_soc_update_bits(codec, RT5651_IRQ_CTRL2, 80bbe4a3 Carlo Caione 2017-10-20 1908 RT5651_MB1_OC_CLR, 0); 80bbe4a3 Carlo Caione 2017-10-20 1909 } else { /* jack out */ 80bbe4a3 Carlo Caione 2017-10-20 1910 jack_type = 0; 80bbe4a3 Carlo Caione 2017-10-20 1911 80bbe4a3 Carlo Caione 2017-10-20 1912 snd_soc_update_bits(codec, RT5651_MICBIAS, 80bbe4a3 Carlo Caione 2017-10-20 1913 RT5651_MIC1_OVCD_MASK, 80bbe4a3 Carlo Caione 2017-10-20 1914 RT5651_MIC1_OVCD_DIS); 80bbe4a3 Carlo Caione 2017-10-20 1915 } 80bbe4a3 Carlo Caione 2017-10-20 1916 80bbe4a3 Carlo Caione 2017-10-20 1917 return jack_type; 80bbe4a3 Carlo Caione 2017-10-20 1918 } 80bbe4a3 Carlo Caione 2017-10-20 1919 80bbe4a3 Carlo Caione 2017-10-20 @1920 static void rt5651_jack_detect_work(struct work_struct *work) 80bbe4a3 Carlo Caione 2017-10-20 1921 { 80bbe4a3 Carlo Caione 2017-10-20 1922 struct rt5651_priv *rt5651 = 80bbe4a3 Carlo Caione 2017-10-20 1923 container_of(work, struct rt5651_priv, jack_detect_work.work); 80bbe4a3 Carlo Caione 2017-10-20 1924 80bbe4a3 Carlo Caione 2017-10-20 1925 int report, val = 0; 80bbe4a3 Carlo Caione 2017-10-20 1926 80bbe4a3 Carlo Caione 2017-10-20 1927 if (!rt5651->codec) 80bbe4a3 Carlo Caione 2017-10-20 1928 return; 80bbe4a3 Carlo Caione 2017-10-20 1929 80bbe4a3 Carlo Caione 2017-10-20 1930 switch (rt5651->pdata.jd_src) { 80bbe4a3 Carlo Caione 2017-10-20 1931 case RT5651_JD1_1: 80bbe4a3 Carlo Caione 2017-10-20 1932 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x1000; 80bbe4a3 Carlo Caione 2017-10-20 1933 break; 80bbe4a3 Carlo Caione 2017-10-20 1934 case RT5651_JD1_2: 80bbe4a3 Carlo Caione 2017-10-20 1935 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x2000; 80bbe4a3 Carlo Caione 2017-10-20 1936 break; 80bbe4a3 Carlo Caione 2017-10-20 1937 case RT5651_JD2: 80bbe4a3 Carlo Caione 2017-10-20 1938 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x4000; 80bbe4a3 Carlo Caione 2017-10-20 1939 break; 80bbe4a3 Carlo Caione 2017-10-20 1940 default: 80bbe4a3 Carlo Caione 2017-10-20 1941 break; 80bbe4a3 Carlo Caione 2017-10-20 1942 } 80bbe4a3 Carlo Caione 2017-10-20 1943 80bbe4a3 Carlo Caione 2017-10-20 1944 report = rt5651_jack_detect(rt5651->codec, !val); 80bbe4a3 Carlo Caione 2017-10-20 1945 80bbe4a3 Carlo Caione 2017-10-20 1946 snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); 80bbe4a3 Carlo Caione 2017-10-20 1947 } 80bbe4a3 Carlo Caione 2017-10-20 1948 80bbe4a3 Carlo Caione 2017-10-20 @1949 int rt5651_set_jack_detect(struct snd_soc_codec *codec, 80bbe4a3 Carlo Caione 2017-10-20 1950 struct snd_soc_jack *hp_jack) 80bbe4a3 Carlo Caione 2017-10-20 1951 { 80bbe4a3 Carlo Caione 2017-10-20 1952 struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); 80bbe4a3 Carlo Caione 2017-10-20 1953 80bbe4a3 Carlo Caione 2017-10-20 1954 rt5651->hp_jack = hp_jack; 80bbe4a3 Carlo Caione 2017-10-20 1955 rt5651_irq(0, rt5651); 80bbe4a3 Carlo Caione 2017-10-20 1956 80bbe4a3 Carlo Caione 2017-10-20 1957 return 0; 80bbe4a3 Carlo Caione 2017-10-20 1958 } 80bbe4a3 Carlo Caione 2017-10-20 @1959 EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); 80bbe4a3 Carlo Caione 2017-10-20 1960 :::::: The code at line 1874 was first introduced by commit :::::: 60d5a1a47b9a8381c08d2263b11ac9c757c87746 ASoC: rt5651: Enable jack detection on JD* pins :::::: TO: Carlo Caione <carlo(a)endlessm.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2 0/5] rt5651: Enable jack detection
by Carlo Caione 26 Oct '17

26 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> Patchset to support jack detection for the rt5651 codec on the JD* pins and enable it on the KIANO cherry-trail laptop. Changelog: v2: - Added quirks for the KIANO laptop - Extended support to JD1_2 and JD2 pins - Reworked quirk logic in the codec driver - Codec driver can now differentiate between headphone and headset (thank you Bard Liao) Carlo Caione (5): ASoC: rt5651: Enable jack detection on JD* pins ASoC: intel: byt: Add headset jack ASoC: intel: byt: Enable IN2 map quirk for a KIANO laptop ASoC: rt5651: Rework quirk logic ASoC: rt5651: Enable JD1_1 quirk for KIANO laptopt include/sound/rt5651.h | 8 ++ sound/soc/codecs/rt5651.c | 215 +++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5651.h | 4 + sound/soc/intel/boards/bytcr_rt5651.c | 31 +++++ 4 files changed, 252 insertions(+), 6 deletions(-) -- 2.14.2

2 11

[alsa-devel] [PATCH] dt-bindings: sgtl5000: Remove leading zero from '@0a' notation
by Marco Franchi 26 Oct '17

26 Oct '17

Improve the binding example by removing the leading 0 from '@0a' notation, which fixes the following build warning: Warning (unit_address_format): Node /sgtl5000@0a unit name should not have leading 0s Signed-off-by: Marco Franchi <marco.franchi(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 7a73a9d..060cb4a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -37,7 +37,7 @@ VDDIO 1.8V 2.5V 3.3V Example: -codec: sgtl5000@0a { +codec: sgtl5000@a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; -- 2.7.4

2 2

Re: [alsa-devel] ALC700 in the Intel chipset
by Pshou 26 Oct '17

26 Oct '17

Hi alsa-devel ML: Could you merge this 000-alc700-intel.patch file to your ALSA driver? I changed "Intel debug" to "Intel Reference board". Best regards Pshou -----Original Message----- From: Takashi Iwai [mailto:tiwai@suse.de] Sent: Friday, September 01, 2017 4:13 PM To: Pshou Cc: CHLin; Kailang; Singh, Guneshwor O Subject: Re: ALC700 in the Intel chipset On Fri, 01 Sep 2017 10:06:13 +0200, Pshou wrote: > > Hi Takashi: > > This is 0000-alc700-intel.patch file. Thanks, it looks good now. One thing to be changed is the name "Intel debug". Maybe "Intel Reference board" or such would fit better. In anyway, could you submit the patch to alsa-devel ML? Once after it's submitted, I'll merge it. thanks, Takashi > > Best regards > pshou > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Tuesday, August 22, 2017 5:04 PM > To: CHLin > Cc: Pshou; Kailang; Singh, Guneshwor O > Subject: Re: ALC700 in the Intel chipset > > On Tue, 22 Aug 2017 10:05:18 +0200, > CHLin wrote: > > > > Hi Takashi > > > > ALC 700 needs this patch for jack detection function. > > Because ALC700's jack detect function defaults is disable. > > So alc700 needs pathc to enable jack detection function. > > OK, then could you update the patch description for giving more such information, prepare the proper patch in a form described in Documentation/SubmittingPatches with signed-off-by tags? > This is mandatory for merging to upstream. > > > thanks, > > Takashi > > > Thanks. > > > > ======================= > > Realtek Semiconductor Corp. > > 林家賢 CH_Lin > > System Design Dept II PCBG > > Tel: 886-3-578-0211 ext:13162 > > ======================= > > > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Monday, August 21, 2017 6:23 PM > > To: Pshou > > Cc: Kailang; Singh, Guneshwor O; CHLin > > Subject: Re: ALC700 in the Intel chipset > > > > On Mon, 21 Aug 2017 10:05:25 +0200, > > Pshou wrote: > > > > > > Hi Tiwai: > > > > > > Nice to meet you. > > > > > > This is the case of intel. > > > Fixing Jack detection on ALC700 > > > > Could you give more information about this patch? > > For which platform and for what problem? At best, please add more information in the patch changelog itself. > > > > > > thanks, > > > > Takashi > > > > ------Please consider the environment before printing this e-mail.

2 5

[alsa-devel] [PATCH] sound: Convert timers to use timer_setup()
by Kees Cook 26 Oct '17

26 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. These are all the "mechanical" changes remaining in the sound subsystem. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/i2c/other/ak4117.c | 8 ++++---- sound/pci/ctxfi/cttimer.c | 7 +++---- sound/pci/echoaudio/midi.c | 10 +++++----- sound/pci/korg1212/korg1212.c | 7 +++---- sound/pci/rme9652/hdsp.c | 8 ++++---- sound/pci/rme9652/hdspm.c | 8 ++++---- sound/soc/codecs/rt5645.c | 7 +++---- sound/soc/omap/ams-delta.c | 4 ++-- 8 files changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 3ab099fb8c15..b923342cadf4 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -35,7 +35,7 @@ MODULE_LICENSE("GPL"); #define AK4117_ADDR 0x00 /* fixed address */ -static void snd_ak4117_timer(unsigned long data); +static void snd_ak4117_timer(struct timer_list *t); static void reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char val) { @@ -91,7 +91,7 @@ int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t chip->read = read; chip->write = write; chip->private_data = private_data; - setup_timer(&chip->timer, snd_ak4117_timer, (unsigned long)chip); + timer_setup(&chip->timer, snd_ak4117_timer, 0); for (reg = 0; reg < 5; reg++) chip->regmap[reg] = pgm[reg]; @@ -529,9 +529,9 @@ int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags) return res; } -static void snd_ak4117_timer(unsigned long data) +static void snd_ak4117_timer(struct timer_list *t) { - struct ak4117 *chip = (struct ak4117 *)data; + struct ak4117 *chip = from_timer(chip, t, timer); if (chip->init) return; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 8f945341720b..08e874e9a7f6 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -63,9 +63,9 @@ struct ct_timer { * system-timer-based updates */ -static void ct_systimer_callback(unsigned long data) +static void ct_systimer_callback(struct timer_list *t) { - struct ct_timer_instance *ti = (struct ct_timer_instance *)data; + struct ct_timer_instance *ti = from_timer(ti, t, timer); struct snd_pcm_substream *substream = ti->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct ct_atc_pcm *apcm = ti->apcm; @@ -93,8 +93,7 @@ static void ct_systimer_callback(unsigned long data) static void ct_systimer_init(struct ct_timer_instance *ti) { - setup_timer(&ti->timer, ct_systimer_callback, - (unsigned long)ti); + timer_setup(&ti->timer, ct_systimer_callback, 0); } static void ct_systimer_start(struct ct_timer_instance *ti) diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 8c685ddb1a41..6045a115cffe 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -199,9 +199,9 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) -static void snd_echo_midi_output_write(unsigned long data) +static void snd_echo_midi_output_write(struct timer_list *t) { - struct echoaudio *chip = (struct echoaudio *)data; + struct echoaudio *chip = from_timer(chip, t, timer); unsigned long flags; int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; @@ -257,8 +257,8 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream spin_lock_irq(&chip->lock); if (up) { if (!chip->tinuse) { - setup_timer(&chip->timer, snd_echo_midi_output_write, - (unsigned long)chip); + timer_setup(&chip->timer, snd_echo_midi_output_write, + 0); chip->tinuse = 1; } } else { @@ -273,7 +273,7 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream spin_unlock_irq(&chip->lock); if (up && !chip->midi_full) - snd_echo_midi_output_write((unsigned long)chip); + snd_echo_midi_output_write(&chip->timer); } diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 04cd71c74e5c..c7b007164c99 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -599,9 +599,9 @@ static void snd_korg1212_SendStopAndWait(struct snd_korg1212 *korg1212) } /* timer callback for checking the ack of stop request */ -static void snd_korg1212_timer_func(unsigned long data) +static void snd_korg1212_timer_func(struct timer_list *t) { - struct snd_korg1212 *korg1212 = (struct snd_korg1212 *) data; + struct snd_korg1212 *korg1212 = from_timer(korg1212, t, timer); unsigned long flags; spin_lock_irqsave(&korg1212->lock, flags); @@ -2189,8 +2189,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, init_waitqueue_head(&korg1212->wait); spin_lock_init(&korg1212->lock); mutex_init(&korg1212->open_mutex); - setup_timer(&korg1212->timer, snd_korg1212_timer_func, - (unsigned long)korg1212); + timer_setup(&korg1212->timer, snd_korg1212_timer_func, 0); korg1212->irq = -1; korg1212->clkSource = K1212_CLKIDX_Local; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 9f0f73875f01..1bff4b1b39cd 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1410,9 +1410,9 @@ static void snd_hdsp_midi_input_trigger(struct snd_rawmidi_substream *substream, spin_unlock_irqrestore (&hdsp->lock, flags); } -static void snd_hdsp_midi_output_timer(unsigned long data) +static void snd_hdsp_midi_output_timer(struct timer_list *t) { - struct hdsp_midi *hmidi = (struct hdsp_midi *) data; + struct hdsp_midi *hmidi = from_timer(hmidi, t, timer); unsigned long flags; snd_hdsp_midi_output_write(hmidi); @@ -1439,8 +1439,8 @@ static void snd_hdsp_midi_output_trigger(struct snd_rawmidi_substream *substream spin_lock_irqsave (&hmidi->lock, flags); if (up) { if (!hmidi->istimer) { - setup_timer(&hmidi->timer, snd_hdsp_midi_output_timer, - (unsigned long) hmidi); + timer_setup(&hmidi->timer, snd_hdsp_midi_output_timer, + 0); mod_timer(&hmidi->timer, 1 + jiffies); hmidi->istimer++; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f20d42714e4d..4c59983158e0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1946,9 +1946,9 @@ snd_hdspm_midi_input_trigger(struct snd_rawmidi_substream *substream, int up) spin_unlock_irqrestore (&hdspm->lock, flags); } -static void snd_hdspm_midi_output_timer(unsigned long data) +static void snd_hdspm_midi_output_timer(struct timer_list *t) { - struct hdspm_midi *hmidi = (struct hdspm_midi *) data; + struct hdspm_midi *hmidi = from_timer(hmidi, t, timer); unsigned long flags; snd_hdspm_midi_output_write(hmidi); @@ -1976,8 +1976,8 @@ snd_hdspm_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) spin_lock_irqsave (&hmidi->lock, flags); if (up) { if (!hmidi->istimer) { - setup_timer(&hmidi->timer, snd_hdspm_midi_output_timer, - (unsigned long) hmidi); + timer_setup(&hmidi->timer, + snd_hdspm_midi_output_timer, 0); mod_timer(&hmidi->timer, 1 + jiffies); hmidi->istimer++; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a98647ac497c..735f2d0033dd 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3340,9 +3340,9 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } -static void rt5645_btn_check_callback(unsigned long data) +static void rt5645_btn_check_callback(struct timer_list *t) { - struct rt5645_priv *rt5645 = (struct rt5645_priv *)data; + struct rt5645_priv *rt5645 = from_timer(rt5645, t, btn_check_timer); queue_delayed_work(system_power_efficient_wq, &rt5645->jack_detect_work, msecs_to_jiffies(5)); @@ -3934,8 +3934,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); + timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 6c49f3d6fd96..d40219678700 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -260,7 +260,7 @@ static bool cx81801_cmd_pending; static bool ams_delta_muted; static DEFINE_SPINLOCK(ams_delta_lock); -static void cx81801_timeout(unsigned long data) +static void cx81801_timeout(struct timer_list *unused) { int muted; @@ -349,7 +349,7 @@ static void cx81801_receive(struct tty_struct *tty, /* First modem response, complete setup procedure */ /* Initialize timer used for config pulse generation */ - setup_timer(&cx81801_timer, cx81801_timeout, 0); + timer_setup(&cx81801_timer, cx81801_timeout, 0); v253_ops.receive_buf(tty, cp, fp, count); -- 2.7.4 -- Kees Cook Pixel Security

3 3

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Decrease loglevel for topology loading
by Chintan Patel 25 Oct '17

25 Oct '17

In case of failure in loading customize topology firmware, dfw_sst.bin gets loaded. However, current log provides this message as error even after successfully falling back to default topology "dfw_sst.bin". Hence to convey proper message, changing log level and message. Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..12f383fd7c39 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3404,7 +3404,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_err(bus->dev, "tplg fw %s load failed with %d\n", + dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", skl->tplg_name, ret); ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/7] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Subhransu S. Prusty 25 Oct '17

25 Oct '17

For certain platforms clocks (mclk/sclk/fs) are required to be up before the stream start. Example: some codecs needs the mclk/sclk/fs to be enabled early for a successful clock synchronization. Some patforms require clock to be enabled at boot and be always ON By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Skylake driver creates a platform driver with the clock information and register the clk ops callback. kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. set_dma_control API can be used with different payload configuration. Modify the arguments to take configurable parameters. v1 -> v2 - Register parent clocks with skylake device. With the patch "clk: Add support for runtime PM" soon to be merged will help DSP to stay active on call to clock enable. Reference: (https://patchwork.kernel.org/patch/9911741/) - Fix the machine driver to enable clocks early for headphone playback path as well to fix a pop noise issue - Include the eve machine driver changes as well Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Jaikrishna Nemallapudi (5): ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments ASoC: Intel: Skylake: Parse nhlt to populate clock information ASoC: Intel: Skylake: Prepare DMA control IPC to enable/disable clock ASoC: Intel: Skylake: Register clock device and ops ASoC: Intel: Skylake: Add ssp clock driver Naveen M (1): ASoC: Intel: eve: Enable mclk and ssp sclk early sound/soc/intel/Kconfig | 9 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 98 ++++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 97 +++++++ sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 56 ++++ sound/soc/intel/skylake/skl-messages.c | 109 +++++++- sound/soc/intel/skylake/skl-nhlt.c | 148 +++++++++++ sound/soc/intel/skylake/skl-ssp-clk.c | 288 +++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 124 +++++++++ sound/soc/intel/skylake/skl-topology.h | 4 +- sound/soc/intel/skylake/skl.c | 235 +++++++++++++++++ sound/soc/intel/skylake/skl.h | 12 + 12 files changed, 1173 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-i2s.h create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h -- 1.9.1

3 10

[alsa-devel] [PATCH] sound: oss: waveartist: Convert timers to use timer_setup()
by Kees Cook 25 Oct '17

25 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: David Howells <dhowells(a)redhat.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/oss/waveartist.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 4f0c3a232e41..c051c8e6023e 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -151,6 +151,7 @@ static DEFINE_SPINLOCK(waveartist_lock); #define machine_is_netwinder() 0 #else static struct timer_list vnc_timer; +static int timer_nr_waveartist_devs; static void vnc_configure_mixer(struct wavnc_info *devc, unsigned int input_mask); static int vnc_private_ioctl(int dev, unsigned int cmd, int __user *arg); @@ -1414,8 +1415,8 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m else { #ifdef CONFIG_ARCH_NETWINDER if (machine_is_netwinder()) { - setup_timer(&vnc_timer, vnc_slider_tick, - nr_waveartist_devs); + timer_nr_waveartist_devs = nr_waveartist_devs; + timer_setup(&vnc_timer, vnc_slider_tick, 0); mod_timer(&vnc_timer, jiffies); vnc_configure_mixer(devc, 0); @@ -1799,11 +1800,11 @@ vnc_slider(struct wavnc_info *devc) } static void -vnc_slider_tick(unsigned long data) +vnc_slider_tick(struct timer_list *unused) { int next_timeout; - if (vnc_slider(adev_info + data)) + if (vnc_slider(adev_info + timer_nr_waveartist_devs)) next_timeout = 5; // mixer reported change else next_timeout = VNC_TIMER_PERIOD; -- 2.7.4 -- Kees Cook Pixel Security

2 5

[alsa-devel] [PATCH] conf/ucm: rt5651: Add UCM config for bytcr-rt5651
by Carlo Caione 25 Oct '17

25 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> Add a UCM configuration for the rt5651 codec on Intel's Cherry-Trail platform. Adapted from [0]. [0] https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/bytcr-rt5651/HiFi.conf | 126 +++++++++++++++++++++++++ src/conf/ucm/bytcr-rt5651/Makefile.am | 4 + src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf | 138 ++++++++++++++++++++++++++++ 5 files changed, 270 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/bytcr-rt5651/HiFi.conf create mode 100644 src/conf/ucm/bytcr-rt5651/Makefile.am create mode 100644 src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf diff --git a/configure.ac b/configure.ac index fbcfa829..c16f2461 100644 --- a/configure.ac +++ b/configure.ac @@ -703,6 +703,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/bytcr-rt5651/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index ea9b3850..2bd4ebe7 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c broxton-rt298 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c broxton-rt298 bytcr-rt5651 diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf new file mode 100644 index 00000000..b9da30f7 --- /dev/null +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -0,0 +1,126 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 + +SectionVerb { + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo HiFi enable sequence" + ] + + DisableSequence [ + exec "echo HiFi disable sequence" + ] + + Value { + PlaybackPCM "hw:bytcrrt5651" + CapturePCM "hw:bytcrrt5651" + } +} + +SectionDevice."Headphones" { + Comment "Headphones playback" + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Headphones playback enable sequence" + cset "name='Headphone Switch' on" + cset "name='HPO L Playback Switch' on" + cset "name='HPO R Playback Switch' on" + cset "name='HP Playback Volume' 100" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Headphones playback disable sequence" + cset "name='Headphone Switch' off" + cset "name='HPO L Playback Switch' off" + cset "name='HPO R Playback Switch' off" + cset "name='HP Playback Volume' 0" + ] + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + } +} + +SectionDevice."Speaker" { + Comment "Speaker playback" + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Speaker enable sequence" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Speaker disable sequence" + cset "name='Speaker Switch' off" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Handset" { + Comment "Handset Microphone" + + ConflictingDevice [ + "Mainmic" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Handset Microphone enable sequence" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Handset Microphone disable sequence" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + ] + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + } +} + + +SectionDevice."Mainmic" { + Comment "Main Microphone" + + ConflictingDevice [ + "Handset" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Main Microphone enable sequence" + cset "name='RECMIXL BST2 Switch' on" + cset "name='RECMIXR BST2 Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Main Microphone disable sequence" + cset "name='RECMIXL BST2 Switch' off" + cset "name='RECMIXR BST2 Switch' off" + ] + + Value { + CaptureChannels "2" + } +} diff --git a/src/conf/ucm/bytcr-rt5651/Makefile.am b/src/conf/ucm/bytcr-rt5651/Makefile.am new file mode 100644 index 00000000..95aec978 --- /dev/null +++ b/src/conf/ucm/bytcr-rt5651/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/bytcr-rt5651 +ucm_DATA = bytcr-rt5651.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf new file mode 100644 index 00000000..2dcc72cc --- /dev/null +++ b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf @@ -0,0 +1,138 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 + +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music" +} + +SectionDefaults [ + cdev "hw:bytcrrt5651" + + # media mixer settings + # compress + cset "name='media0_in Gain 0 Switch' on" + cset "name='media0_in Gain 0 Volume' 0" + # normal + cset "name='media1_in Gain 0 Switch' on" + cset "name='media1_in Gain 0 Volume' 0" + # swm loopback + cset "name='media2_in Gain 0 Switch' off" + cset "name='media2_in Gain 0 Volume' 0%" + # deep buffer + cset "name='media3_in Gain 0 Switch' on" + cset "name='media3_in Gain 0 Volume' 0" + cset "name='media0_out mix 0 media0_in Switch' on" + cset "name='media0_out mix 0 media1_in Switch' on" + cset "name='media0_out mix 0 media2_in Switch' off" + cset "name='media0_out mix 0 media3_in Switch' on" + cset "name='media1_out mix 0 media0_in Switch' off" + cset "name='media1_out mix 0 media1_in Switch' off" + cset "name='media1_out mix 0 media2_in Switch' off" + cset "name='media1_out mix 0 media3_in Switch' off" + cset "name='pcm0_in Gain 0 Switch' on" + cset "name='pcm0_in Gain 0 Volume' 0" + cset "name='pcm1_in Gain 0 Switch' off" + cset "name='pcm1_in Gain 0 Volume' 0%" + + # codec0_out settings + cset "name='codec_out0 mix 0 codec_in0 Switch' off" + cset "name='codec_out0 mix 0 codec_in1 Switch' off" + cset "name='codec_out0 mix 0 media_loop1_in Switch' off" + cset "name='codec_out0 mix 0 media_loop2_in Switch' off" + cset "name='codec_out0 mix 0 pcm0_in Switch' on" + cset "name='codec_out0 mix 0 pcm1_in Switch' off" + cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" + cset "name='codec_out0 Gain 0 Switch' on" + cset "name='codec_out0 Gain 0 Volume' 0" + + # input settings + # pcm1_out settings + cset "name='codec_in0 Gain 0 Switch' on" + cset "name='codec_in0 Gain 0 Volume' 0" + cset "name='pcm1_out mix 0 codec_in0 Switch' on" + cset "name='pcm1_out mix 0 codec_in1 Switch' off" + cset "name='pcm1_out mix 0 media_loop1_in Switch' off" + cset "name='pcm1_out mix 0 media_loop2_in Switch' off" + cset "name='pcm1_out mix 0 pcm0_in Switch' off" + cset "name='pcm1_out mix 0 pcm1_in Switch' off" + cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" + cset "name='pcm1_out Gain 0 Switch' on" + cset "name='pcm1_out Gain 0 Volume' 0" + + # disable codec_out1 + cset "name='codec_out1 mix 0 codec_in0 Switch' off" + cset "name='codec_out1 mix 0 codec_in1 Switch' off" + cset "name='codec_out1 mix 0 media_loop1_in Switch' off" + cset "name='codec_out1 mix 0 media_loop2_in Switch' off" + cset "name='codec_out1 mix 0 pcm0_in Switch' off" + cset "name='codec_out1 mix 0 pcm1_in Switch' off" + cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" + cset "name='codec_out1 Gain 0 Switch' off" + cset "name='codec_out1 Gain 0 Volume' 0%" + + # disable codec_in1 + cset "name='codec_in1 Gain 0 Switch' off" + cset "name='codec_in1 Gain 0 Volume' 0%" + + # disable all loops + cset "name='media_loop1_out mix 0 codec_in0 Switch' off" + cset "name='media_loop1_out mix 0 codec_in1 Switch' off" + cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" + cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop1_out mix 0 pcm0_in Switch' off" + cset "name='media_loop1_out mix 0 pcm1_in Switch' off" + cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" + cset "name='media_loop2_out mix 0 codec_in0 Switch' off" + cset "name='media_loop2_out mix 0 codec_in1 Switch' off" + cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" + cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop2_out mix 0 pcm0_in Switch' off" + cset "name='media_loop2_out mix 0 pcm1_in Switch' off" + cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" + cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" + cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" + cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" + cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" + cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" + cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" + cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" + + # default state + cset "name='ADC Capture Volume' 126" + cset "name='HP Playback Volume' 29" + cset "name='HPO L Playback Switch' on" + cset "name='HPO MIX DAC1 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='HPO R Playback Switch' on" + cset "name='HPOVOL L Switch' on" + cset "name='HPOVOL R Switch' on" + cset "name='IF1 ASRC Switch' on" + cset "name='IN Capture Volume' 27" + cset "name='IN2 Boost' 1" + cset "name='IN1 Boost' 1" + cset "name='LOUT L Playback Switch' on" + cset "name='LOUT MIX DAC L1 Switch' on" + cset "name='LOUT MIX DAC R1 Switch' on" + cset "name='LOUT R Playback Switch' on" + cset "name='OUT Playback Volume' 33" + cset "name='Stereo DAC MIXL DAC L1 Switch' on" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC2 Switch' on" + cset "name='Stereo2 ADC MIXL ADC1 Switch' on" + cset "name='Stereo2 ADC MIXL ADC2 Switch' on" + cset "name='Stereo2 ADC MIXR ADC1 Switch' on" + cset "name='Stereo2 ADC MIXR ADC2 Switch' on" + + # input + cset "name='Stereo1 ADC L1 Mux' ADC" + cset "name='Stereo1 ADC R1 Mux' ADC" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='RECMIXL BST3 Switch' off" + cset "name='RECMIXR BST3 Switch' off" + cset "name='RECMIXL BST2 Switch' off" + cset "name='RECMIXR BST2 Switch' off" +] -- 2.14.2

3 5

[alsa-devel] [PATCH] ALSA: sb: Minor optimization / fix of timer usage in sb8_midi.c
by Takashi Iwai 25 Oct '17

25 Oct '17

Currently the SB8 MIDI code sets up the timer object at each time before scheduling it at trigger callback, but basically this is superfluous once after set up. Also, the code misses the del_timer_sync() call that may leave a race condition for use-after-free. This patch addresses these issues, moving timer_setup() to snd_sb8dsp_midi(), and adding the del_timer_sync() call at snd_sb8dsp_midi_output_trigger() to make sure. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/sb/sb8_midi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index 05ba2c6b1a8c..4affdcb78f72 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -138,6 +138,7 @@ static int snd_sb8dsp_midi_output_close(struct snd_rawmidi_substream *substream) struct snd_sb *chip; chip = substream->rmidi->private_data; + del_timer_sync(&chip->midi_timer); spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~(SB_OPEN_MIDI_OUTPUT | SB_OPEN_MIDI_OUTPUT_TRIGGER); chip->midi_substream_output = NULL; @@ -230,8 +231,6 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre spin_lock_irqsave(&chip->open_lock, flags); if (up) { if (!(chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER)) { - timer_setup(&chip->midi_timer, - snd_sb8dsp_midi_output_timer, 0); mod_timer(&chip->midi_timer, 1 + jiffies); chip->open |= SB_OPEN_MIDI_OUTPUT_TRIGGER; } @@ -274,6 +273,7 @@ int snd_sb8dsp_midi(struct snd_sb *chip, int device) if (chip->hardware >= SB_HW_20) rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = chip; + timer_setup(&chip->midi_timer, snd_sb8dsp_midi_output_timer, 0); chip->rmidi = rmidi; return 0; } -- 2.14.2

1 0

[alsa-devel] [RFCv3][PATCH 00/39] axfer: rewrite aplay, adding 'timer-based scheduling' option
by Takashi Sakamoto 25 Oct '17

25 Oct '17

Hi, This 3rd RFC patchset updates my previous one: [alsa-devel] [RFCv2][PATCH 00/38] alsa-utils: axfer: rewrite aplay http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125574.… For aim of this rewrite and its intension, please refer to my former posts. In this version, I add an option to support 'timer-based scheduling' scenario, which PulseAudio developers introduced. Please refer to patch 32: * axfer: add support for timer-based MMAP operation You can use this mode by adding '-sched-type=timer' option into command line. The other difference from my previous one: - fix some bugs to handle non-interleaved buffer on mapped page frame. - use snd_pcm_status() instead of snd_pcm_state() to execute hwsync. - minor fixes. In my plan, this is the last version including below patches. In next post, options added by below patches are categorized as obsoleted: * axfer: add an option to support volume unit meter * axfer: add a unit test for vumeter calculation * axfer: add an option for formatted filename * axfer: add an option to handle key events * axfer: add a parser for channel map API * axfer: add a feature to generate a file for process id Additionally, for next post, I'll prepare for help messages and man for this program. Regards Takashi Sakamoto (39): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a parser for command-line options axfer: add a common interface to transfer data frames axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add a common interface of waiter for I/O event notification axfer: add support of waiter for poll(2) axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add an implementation of waiter for epoll(7) axfer: add an option for waiter type axfer: add support for timer-based MMAP operation axfer: add an option to support volume unit meter axfer: add a unit test for vumeter calculation axfer: add an option for formatted filename axfer: add an option to handle key events axfer: add a parser for channel map API axfer: add a feature to generate a file for process id axfer: obsolete some test options Makefile.am | 2 +- axfer/Makefile.am | 61 +++ axfer/container-au.c | 207 ++++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 581 ++++++++++++++++++++++ axfer/container-voc.c | 843 ++++++++++++++++++++++++++++++++ axfer/container.c | 470 ++++++++++++++++++ axfer/container.h | 126 +++++ axfer/key-event.c | 121 +++++ axfer/key-event.h | 19 + axfer/main.c | 194 ++++++++ axfer/mapper-multiple.c | 271 +++++++++++ axfer/mapper-single.c | 195 ++++++++ axfer/mapper.c | 152 ++++++ axfer/mapper.h | 87 ++++ axfer/misc.h | 16 + axfer/options.c | 980 ++++++++++++++++++++++++++++++++++++++ axfer/options.h | 92 ++++ axfer/subcmd-list.c | 234 +++++++++ axfer/subcmd-transfer.c | 525 ++++++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 39 ++ axfer/test/container-test.c | 299 ++++++++++++ axfer/test/generator.c | 260 ++++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 +++++++++++++++++++ axfer/test/vumeter-test.c | 184 +++++++ axfer/vumeter.c | 546 +++++++++++++++++++++ axfer/vumeter.h | 55 +++ axfer/waiter-epoll.c | 81 ++++ axfer/waiter-poll.c | 68 +++ axfer/waiter.c | 84 ++++ axfer/waiter.h | 52 ++ axfer/xfer-libasound-irq-mmap.c | 271 +++++++++++ axfer/xfer-libasound-irq-rw.c | 520 ++++++++++++++++++++ axfer/xfer-libasound-timer-mmap.c | 474 ++++++++++++++++++ axfer/xfer-libasound.c | 737 ++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 60 +++ axfer/xfer.c | 191 ++++++++ axfer/xfer.h | 76 +++ configure.ac | 3 +- 41 files changed, 9796 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/key-event.c create mode 100644 axfer/key-event.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/options.c create mode 100644 axfer/options.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/test/vumeter-test.c create mode 100644 axfer/vumeter.c create mode 100644 axfer/vumeter.h create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound-timer-mmap.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h -- 2.11.0

2 40

Re: [alsa-devel] [Patch v6 1/7] slimbus: Device management on SLIMbus
by Rishi Bhattacharya 25 Oct '17

25 Oct '17

1 0

[alsa-devel] [PATCH] ALSA: wavefront: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- include/sound/snd_wavefront.h | 1 + sound/isa/wavefront/wavefront_midi.c | 10 +++++----- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/include/sound/snd_wavefront.h b/include/sound/snd_wavefront.h index cd0bab1ef6f1..fcd770fdcda5 100644 --- a/include/sound/snd_wavefront.h +++ b/include/sound/snd_wavefront.h @@ -28,6 +28,7 @@ struct _snd_wavefront_midi { struct snd_rawmidi_substream *substream_output[2]; struct snd_rawmidi_substream *substream_input[2]; struct timer_list timer; + snd_wavefront_card_t *timer_card; spinlock_t open; spinlock_t virtual; /* protects isvirtual */ }; diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c index 2aa05f3aaa38..556b14738970 100644 --- a/sound/isa/wavefront/wavefront_midi.c +++ b/sound/isa/wavefront/wavefront_midi.c @@ -349,10 +349,10 @@ static void snd_wavefront_midi_input_trigger(struct snd_rawmidi_substream *subst spin_unlock_irqrestore (&midi->virtual, flags); } -static void snd_wavefront_midi_output_timer(unsigned long data) +static void snd_wavefront_midi_output_timer(struct timer_list *t) { - snd_wavefront_card_t *card = (snd_wavefront_card_t *)data; - snd_wavefront_midi_t *midi = &card->wavefront.midi; + snd_wavefront_midi_t *midi = from_timer(midi, t, timer); + snd_wavefront_card_t *card = midi->timer_card; unsigned long flags; spin_lock_irqsave (&midi->virtual, flags); @@ -383,9 +383,9 @@ static void snd_wavefront_midi_output_trigger(struct snd_rawmidi_substream *subs if (up) { if ((midi->mode[mpu] & MPU401_MODE_OUTPUT_TRIGGER) == 0) { if (!midi->istimer) { - setup_timer(&midi->timer, + timer_setup(&midi->timer, snd_wavefront_midi_output_timer, - (unsigned long) substream->rmidi->card->private_data); + 0); mod_timer(&midi->timer, 1 + jiffies); } midi->istimer++; -- 2.7.4 -- Kees Cook Pixel Security

2 2

[alsa-devel] snd_pcm_ioplug_avail_update() confusion
by Rob Duncan 24 Oct '17

24 Oct '17

I'm working on an external IO plugin based on ioplug and I'm confused about the behavior of snd_pcm_ioplug_avail_update(). Specifically this call (src/pcm/pcm_ioplug.c:657): result = io->data->callback->transfer(io->data, areas, offset, size); Why isn't there a subsequent call to snd_pcm_mmap_appl_forward() like there is in ioplug_priv_transfer_areas()? It seems that without such a call the captured data will be lost, or have I missed something? Thanks, Rob.

2 1

[alsa-devel] [PATCH] ALSA: sb: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- include/sound/sb.h | 1 + sound/isa/sb/emu8000_pcm.c | 6 +++--- sound/isa/sb/sb8_midi.c | 12 ++++++------ 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/include/sound/sb.h b/include/sound/sb.h index bacefaee411a..4aef395189ec 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -83,6 +83,7 @@ struct snd_sb { unsigned int playback_format; unsigned int capture_format; struct timer_list midi_timer; + struct snd_rawmidi_substream *timer_substream; unsigned int p_dma_size; unsigned int p_period_size; unsigned int c_dma_size; diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index 8f34551abd8d..bc5af71d3bdb 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -193,9 +193,9 @@ static inline int emu8k_get_curpos(struct snd_emu8k_pcm *rec, int ch) * timer interrupt handler * check the current position and update the period if necessary. */ -static void emu8k_pcm_timer_func(unsigned long data) +static void emu8k_pcm_timer_func(struct timer_list *t) { - struct snd_emu8k_pcm *rec = (struct snd_emu8k_pcm *)data; + struct snd_emu8k_pcm *rec = from_timer(rec, t, timer); int ptr, delta; spin_lock(&rec->timer_lock); @@ -241,7 +241,7 @@ static int emu8k_pcm_open(struct snd_pcm_substream *subs) runtime->private_data = rec; spin_lock_init(&rec->timer_lock); - setup_timer(&rec->timer, emu8k_pcm_timer_func, (unsigned long)rec); + timer_setup(&rec->timer, emu8k_pcm_timer_func, 0); runtime->hw = emu8k_pcm_hw; runtime->hw.buffer_bytes_max = emu->mem_size - LOOP_BLANK_SIZE * 3; diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index bd672abb4854..cf23dc5d0507 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -209,10 +209,10 @@ static void snd_sb8dsp_midi_output_write(struct snd_rawmidi_substream *substream } } -static void snd_sb8dsp_midi_output_timer(unsigned long data) +static void snd_sb8dsp_midi_output_timer(struct timer_list *t) { - struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *) data; - struct snd_sb * chip = substream->rmidi->private_data; + struct snd_sb *chip = from_timer(chip, t, midi_timer); + struct snd_rawmidi_substream *substream = chip->timer_substream; unsigned long flags; spin_lock_irqsave(&chip->open_lock, flags); @@ -230,9 +230,9 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre spin_lock_irqsave(&chip->open_lock, flags); if (up) { if (!(chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER)) { - setup_timer(&chip->midi_timer, - snd_sb8dsp_midi_output_timer, - (unsigned long) substream); + chip->timer_substream = substream; + timer_setup(&chip->midi_timer, + snd_sb8dsp_midi_output_timer, 0); mod_timer(&chip->midi_timer, 1 + jiffies); chip->open |= SB_OPEN_MIDI_OUTPUT_TRIGGER; } -- 2.7.4 -- Kees Cook Pixel Security

2 2

[alsa-devel] [PATCH] ALSA: emux: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/synth/emux/emux.c | 2 +- sound/synth/emux/emux_synth.c | 4 ++-- sound/synth/emux/emux_voice.h | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index b9981e8c0027..b840ff2dcfbb 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -53,7 +53,7 @@ int snd_emux_new(struct snd_emux **remu) emu->max_voices = 0; emu->use_time = 0; - setup_timer(&emu->tlist, snd_emux_timer_callback, (unsigned long)emu); + timer_setup(&emu->tlist, snd_emux_timer_callback, 0); emu->timer_active = 0; *remu = emu; diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 599551b5af44..9fa696b0dbde 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -202,9 +202,9 @@ snd_emux_note_off(void *p, int note, int vel, struct snd_midi_channel *chan) * * release the pending note-offs */ -void snd_emux_timer_callback(unsigned long data) +void snd_emux_timer_callback(struct timer_list *t) { - struct snd_emux *emu = (struct snd_emux *) data; + struct snd_emux *emu = from_timer(emu, t, tlist); struct snd_emux_voice *vp; unsigned long flags; int ch, do_again = 0; diff --git a/sound/synth/emux/emux_voice.h b/sound/synth/emux/emux_voice.h index a7073c371bcc..326fa8993d7b 100644 --- a/sound/synth/emux/emux_voice.h +++ b/sound/synth/emux/emux_voice.h @@ -55,7 +55,7 @@ void snd_emux_update_channel(struct snd_emux_port *port, struct snd_midi_channel *chan, int update); void snd_emux_update_port(struct snd_emux_port *port, int update); -void snd_emux_timer_callback(unsigned long data); +void snd_emux_timer_callback(struct timer_list *t); /* emux_effect.c */ #ifdef SNDRV_EMUX_USE_RAW_EFFECT -- 2.7.4 -- Kees Cook Pixel Security

2 1

[alsa-devel] [PATCH] ALSA: drivers: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Arvind Yadav <arvind.yadav.cs(a)gmail.com> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Markus Elfring <elfring(a)users.sourceforge.net> Cc: David Howells <dhowells(a)redhat.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/drivers/aloop.c | 7 +++---- sound/drivers/dummy.c | 7 +++---- sound/drivers/mpu401/mpu401_uart.c | 7 +++---- sound/drivers/mtpav.c | 7 +++---- sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/drivers/opl3/opl3_seq.c | 2 +- sound/drivers/opl3/opl3_voice.h | 2 +- sound/drivers/serial-u16550.c | 7 +++---- 8 files changed, 19 insertions(+), 24 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 135adb17703c..afac886ffa28 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -529,9 +529,9 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) return running; } -static void loopback_timer_function(unsigned long data) +static void loopback_timer_function(struct timer_list *t) { - struct loopback_pcm *dpcm = (struct loopback_pcm *)data; + struct loopback_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; spin_lock_irqsave(&dpcm->cable->lock, flags); @@ -675,8 +675,7 @@ static int loopback_open(struct snd_pcm_substream *substream) } dpcm->loopback = loopback; dpcm->substream = substream; - setup_timer(&dpcm->timer, loopback_timer_function, - (unsigned long)dpcm); + timer_setup(&dpcm->timer, loopback_timer_function, 0); cable = loopback->cables[substream->number][dev]; if (!cable) { diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index c0939a0164a6..7b2b1f766b00 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -306,9 +306,9 @@ static int dummy_systimer_prepare(struct snd_pcm_substream *substream) return 0; } -static void dummy_systimer_callback(unsigned long data) +static void dummy_systimer_callback(struct timer_list *t) { - struct dummy_systimer_pcm *dpcm = (struct dummy_systimer_pcm *)data; + struct dummy_systimer_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; int elapsed = 0; @@ -343,8 +343,7 @@ static int dummy_systimer_create(struct snd_pcm_substream *substream) if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - setup_timer(&dpcm->timer, dummy_systimer_callback, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, dummy_systimer_callback, 0); spin_lock_init(&dpcm->lock); dpcm->substream = substream; return 0; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index b997222274bd..3e745f47dd2f 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -169,9 +169,9 @@ EXPORT_SYMBOL(snd_mpu401_uart_interrupt_tx); * timer callback * reprogram the timer and call the interrupt job */ -static void snd_mpu401_uart_timer(unsigned long data) +static void snd_mpu401_uart_timer(struct timer_list *t) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *)data; + struct snd_mpu401 *mpu = from_timer(mpu, t, timer); unsigned long flags; spin_lock_irqsave(&mpu->timer_lock, flags); @@ -191,8 +191,7 @@ static void snd_mpu401_uart_add_timer (struct snd_mpu401 *mpu, int input) spin_lock_irqsave (&mpu->timer_lock, flags); if (mpu->timer_invoked == 0) { - setup_timer(&mpu->timer, snd_mpu401_uart_timer, - (unsigned long)mpu); + timer_setup(&mpu->timer, snd_mpu401_uart_timer, 0); mod_timer(&mpu->timer, 1 + jiffies); } mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER : diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 0f6392001e30..547662e02fcc 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -406,10 +406,10 @@ static void snd_mtpav_input_trigger(struct snd_rawmidi_substream *substream, int * timer interrupt for outputs */ -static void snd_mtpav_output_timer(unsigned long data) +static void snd_mtpav_output_timer(struct timer_list *t) { unsigned long flags; - struct mtpav *chip = (struct mtpav *)data; + struct mtpav *chip = from_timer(chip, t, timer); int p; spin_lock_irqsave(&chip->spinlock, flags); @@ -707,8 +707,7 @@ static int snd_mtpav_probe(struct platform_device *dev) mtp_card->share_irq = 0; mtp_card->inmidistate = 0; mtp_card->outmidihwport = 0xffffffff; - setup_timer(&mtp_card->timer, snd_mtpav_output_timer, - (unsigned long) mtp_card); + timer_setup(&mtp_card->timer, snd_mtpav_output_timer, 0); card->private_free = snd_mtpav_free; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 13c0a7e1bc2b..bb3f3a5a6951 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -238,10 +238,10 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, /* * System timer interrupt function */ -void snd_opl3_timer_func(unsigned long data) +void snd_opl3_timer_func(struct timer_list *t) { - struct snd_opl3 *opl3 = (struct snd_opl3 *)data; + struct snd_opl3 *opl3 = from_timer(opl3, t, tlist); unsigned long flags; int again = 0; int i; diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index d3e91be8b23a..5f881c4cdf1f 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -248,7 +248,7 @@ static int snd_opl3_seq_probe(struct device *_dev) } /* setup system timer */ - setup_timer(&opl3->tlist, snd_opl3_timer_func, (unsigned long) opl3); + timer_setup(&opl3->tlist, snd_opl3_timer_func, 0); spin_lock_init(&opl3->sys_timer_lock); opl3->sys_timer_status = 0; diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h index eaef435e0528..a2445163008e 100644 --- a/sound/drivers/opl3/opl3_voice.h +++ b/sound/drivers/opl3/opl3_voice.h @@ -37,7 +37,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_chann void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset); void snd_opl3_calc_volume(unsigned char *reg, int vel, struct snd_midi_channel *chan); -void snd_opl3_timer_func(unsigned long data); +void snd_opl3_timer_func(struct timer_list *t); /* Prototypes for opl3_drums.c */ void snd_opl3_load_drums(struct snd_opl3 *opl3); diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 88e66ea0306d..0a67b8b9f176 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -309,12 +309,12 @@ static irqreturn_t snd_uart16550_interrupt(int irq, void *dev_id) } /* When the polling mode, this function calls snd_uart16550_io_loop. */ -static void snd_uart16550_buffer_timer(unsigned long data) +static void snd_uart16550_buffer_timer(struct timer_list *t) { unsigned long flags; struct snd_uart16550 *uart; - uart = (struct snd_uart16550 *)data; + uart = from_timer(uart, t, buffer_timer); spin_lock_irqsave(&uart->open_lock, flags); snd_uart16550_del_timer(uart); snd_uart16550_io_loop(uart); @@ -828,8 +828,7 @@ static int snd_uart16550_create(struct snd_card *card, uart->prev_in = 0; uart->rstatus = 0; memset(uart->prev_status, 0x80, sizeof(unsigned char) * SNDRV_SERIAL_MAX_OUTS); - setup_timer(&uart->buffer_timer, snd_uart16550_buffer_timer, - (unsigned long)uart); + timer_setup(&uart->buffer_timer, snd_uart16550_buffer_timer, 0); uart->timer_running = 0; /* Register device */ -- 2.7.4 -- Kees Cook Pixel Security

2 1

[alsa-devel] [PATCH] ALSA: hda controller - handle -1 value as error in snd_hdac_bus_parse_capabilities()
by Jaroslav Kysela 24 Oct '17

24 Oct '17

We have seen this value on a development platform. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/hda/hdac_controller.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc1801b3a..4a1823290b2d 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -278,6 +278,11 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) do { cur_cap = _snd_hdac_chip_readl(bus, offset); + if (cur_cap == 0xffffffff) { + dev_dbg(bus->dev, "Wrong capability -1\n"); + break; + } + dev_dbg(bus->dev, "Capability version: 0x%x\n", (cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF); -- 2.13.6

2 1

[alsa-devel] conflict between alsa-lib and libasound2-data regarding alsa.conf file:
by arne 24 Oct '17

24 Oct '17

Hi, Debian conflicts between alsa-lib_1.1.4 and libasound2-data 1.1.3: in the source tree of alsa-lib I do: ./configure && make && sudo checkinstall and the result is: Unpacking alsa-lib (1.1.4.1-1) ... dpkg: error processing archive /home/temp/alsa-lib-1.1.4.1/alsa-lib_1.1.4.1-1_amd64.deb (--install): trying to overwrite '/usr/share/alsa/alsa.conf', which is also in package libasound2-data 1.1.3-5 This is on a Debian amd64 stable computer Thanks!

3 2

[alsa-devel] New to this list. Proposed patch to soc-utils.c
by Albin Stigö 24 Oct '17

24 Oct '17

Hello, I'm new to this list so please excuse my ignorance. I'm an anesthesiologist by profession so I might be a bit out of my depth here... I was doing some experimentation with custom i2s codecs (FPGAs) using the snd-soc-dummy codec (defined in soc-utils.c). I found this codec to be needlessly restrictive with sample rates allowed (SNDRV_PCM_RATE_8000_192000). I needed to use more "exotic" sample rates so I came up with a patch. I purpose using SNDRV_PCM_RATE_KNOT and setting some limits: .rate_min = 8000, .rate_max = 384000, It's my understanding that SNDRV_PCM_RATE_CONTINIOUS is used for "known" rates and SNDRV_PCM_RATE_KNOT for more or less arbitrary rates? I'm willing to submit my patch but might need some mentoring in the process. I don't think this simple patch could break anything..? --Albin

1 1

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Subhransu S. Prusty 24 Oct '17

24 Oct '17

For certain platforms we require to start the clocks (mclk/sclk/fs) before the stream start. Example: for few systems, codec needs the mclk/sclk/fs to be enabled early for a successful clock synchronization and for some others, clock need to be enabled at boot and should be ON always. By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Skylake driver creates a platform driver with the clock information and register the clk ops callback. kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. set_dma_control API can be used with different payload configuration. Modify the arguments to take configurable parameters. Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Jaikrishna Nemallapudi (5): ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments ASoC: Intel: Skylake: Parse nhlt to populate clock information ASoC: Intel: Skylake: Prepare DMA control IPC to enable/disable clock ASoC: Intel: Skylake: Register clock device and ops ASoC: Intel: Skylake: Add ssp clock driver sound/soc/intel/Kconfig | 8 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 94 ++++++++- sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 56 ++++++ sound/soc/intel/skylake/skl-messages.c | 109 +++++++++- sound/soc/intel/skylake/skl-nhlt.c | 148 ++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.c | 287 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 124 ++++++++++++ sound/soc/intel/skylake/skl-topology.h | 4 +- sound/soc/intel/skylake/skl.c | 235 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 12 ++ 11 files changed, 1070 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-i2s.h create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h -- 1.9.1

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[alsa-devel] ASoC: writing basic custom codecs?
by Albin Stigö 24 Oct '17

24 Oct '17

Hello, I have implemented a custom i2s codec on an FPGA. Because of how the FPGA is clocked I'm getting samples at 39ksps, which to my surprise worked great after some simple kernel patching (the codec is bit & frame master). For experimentation I used the snd-soc-dummy coded defined in soc-utils.c. Now I'm trying to write a very simple codec to "lock" the rate to 39000: https://github.com/ast/vfzfpga-codec The driver loads just fine but I just get "not registered" when I try to use the codec. I feel I'm missing something very obvious here. I've been through the documentation and tried to read the sources but I'm a bit stuck.. What am I missing? --Albin

1 0

[alsa-devel] Prague Audio miniconference - draft schedule
by Mark Brown 24 Oct '17

24 Oct '17

As previously announced[1] by Takashi our annual Linux audio miniconference will be held this year at the SuSE offices in Prague on Friday 27th October (the week of ELC-E). Thanks again to SuSE for sponsoring this. Following the earlier discussion I've pulled together a list of topics. I've put people's names against everything, mostly the people who suggested things but sometimes also other people who should probably be involved. I'm proposing that we work through things in roughly this order. Does this make sense to people? Any other topics? Summary - API status summary (tiwai) - Userspace release schedule - Docs - Dropping old drivers - Treble overview (Eric) - aplay rewrite (Sakamoto-san) ABI issues - USB PCM handling (tiwai) - mmap no IRQ (Eric) - TLV API issues (Sakamoto-san) - Obsolete dimen from struct snd_ctl_elem_info (Sakamoto-san) - Y2038 (tiwai) Future stuff/internals - Clock handling (Peter) - CCF, multi-clock selection - Digital domains (Charles/Lars/Morimoto-san) - Large regmaps (Charles) [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124623.html

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[alsa-devel] [PATCH] ALSA: hda - fix headset mic problem for Dell machines with alc236
by Hui Wang 24 Oct '17

24 Oct '17

We have several Dell laptops which use the codec alc236, the headset mic can't work on these machines. Following the commit 736f20a70, we add the pin cfg table to make the headset mic work. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701eb7d..02c79d6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x1b, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60140}, + {0x14, 0x90170110}, + {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60140}, + {0x14, 0x90170150}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x02211020}), -- 2.7.4

2 1

[alsa-devel] [PATCH v2 34/37] ALSA/dummy: Replace tasklet with softirq hrtimer
by Anna-Maria Gleixner 24 Oct '17

24 Oct '17

From: Thomas Gleixner <tglx(a)linutronix.de> The tasklet is used to defer the execution of snd_pcm_period_elapsed() to the softirq context. Using the HRTIMER_MODE_SOFT mode invokes the timer callback in softirq context as well which renders the tasklet useless. [o-takashi: avoid stall due to a call of hrtimer_cancel() on a callback of hrtimer] Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/20170905161820.jtysvxtfleunbbmf@breakpoint.cc --- On 2017-09-06 01:05:43 [+0900], Takashi Sakamoto wrote: > As Iwai-san mentioned, in this function, .trigger can be called in two > cases; XRUN occurs and draining is done. Thus, let you change the comment as > 'In cases of XRUN and draining, this calls .trigger to stop PCM substream.'. > I'm sorry to trouble you. > > snd_pcm_period_elapsed() > ->snd_pcm_update_hw_ptr0() > ->snd_pcm_update_state() > ->snd_pcm_drain_done() > ... > ->.trigger(TRIGGER_STOP) > ->xrun() > ->snd_pcm_stop() > ... > ->.trigger(TRIGGER_STOP) > I think you asked me just to update the comment. Did I do it right? v2...v3: updated the comment as per Takashi Sakamoto's suggestion. v1...v2: merged Takashi Sakamoto fixup of the original patch into v2. sound/drivers/dummy.c | 27 ++++++++++++--------------- 1 file changed, 12 insertions(+), 15 deletions(-) --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -376,17 +376,9 @@ struct dummy_hrtimer_pcm { ktime_t period_time; atomic_t running; struct hrtimer timer; - struct tasklet_struct tasklet; struct snd_pcm_substream *substream; }; -static void dummy_hrtimer_pcm_elapsed(unsigned long priv) -{ - struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; - if (atomic_read(&dpcm->running)) - snd_pcm_period_elapsed(dpcm->substream); -} - static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) { struct dummy_hrtimer_pcm *dpcm; @@ -394,7 +386,14 @@ static enum hrtimer_restart dummy_hrtime dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); if (!atomic_read(&dpcm->running)) return HRTIMER_NORESTART; - tasklet_schedule(&dpcm->tasklet); + /* + * In cases of XRUN and draining, this calls .trigger to stop PCM + * substream. + */ + snd_pcm_period_elapsed(dpcm->substream); + if (!atomic_read(&dpcm->running)) + return HRTIMER_NORESTART; + hrtimer_forward_now(timer, dpcm->period_time); return HRTIMER_RESTART; } @@ -404,7 +403,7 @@ static int dummy_hrtimer_start(struct sn struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; dpcm->base_time = hrtimer_cb_get_time(&dpcm->timer); - hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL); + hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL_SOFT); atomic_set(&dpcm->running, 1); return 0; } @@ -414,14 +413,14 @@ static int dummy_hrtimer_stop(struct snd struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; atomic_set(&dpcm->running, 0); - hrtimer_cancel(&dpcm->timer); + if (!hrtimer_callback_running(&dpcm->timer)) + hrtimer_cancel(&dpcm->timer); return 0; } static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { hrtimer_cancel(&dpcm->timer); - tasklet_kill(&dpcm->tasklet); } static snd_pcm_uframes_t @@ -466,12 +465,10 @@ static int dummy_hrtimer_create(struct s if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT); dpcm->timer.function = dummy_hrtimer_callback; dpcm->substream = substream; atomic_set(&dpcm->running, 0); - tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, - (unsigned long)dpcm); return 0; }

2 1

[alsa-devel] [PATCH v3] ALSA: hda: Abort capability probe on invalid capability
by Rakesh Ughreja 24 Oct '17

24 Oct '17

On reading wrong capability pointer values driver may crash, so whenever driver discovers unknown HDA capability, log it as error and stop traversing the link list further. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..8f7d0d9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; break; } -- 2.7.4

3 2

[alsa-devel] how to dynamic link one cpu dai to codec dai?
by Gou, Kylin 24 Oct '17

24 Oct '17

Hi Sir, I registered 2 codec dai and 1 cpu dai . However , I want to link cpu dai to different codec dai , can we dynamic link cpu dai to each codec dai? I didn't find any document for it. Thanks, Kylin ******************************* IMPORTANT MESSAGE ****************************** This transmission may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the information contained herein (including any reliance thereon) is STRICTLY PROHIBITED. If you received this transmission in error, please immediately contact the sender and destroy the material in its entirety, whether in electronic or hard copy format. Thank you. ********************************************************************************

1 0

[alsa-devel] Prague Audio miniconference - topics for discussion?
by Mark Brown 23 Oct '17

23 Oct '17

As previously announced[1] by Takashi our annual Linux audio miniconference will be held this year at the SuSE offices in Prague on 27th October (the week of ELC-E). Thanks again to SuSe for sponsoring this. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! I'll start things off by mentining the TLV issues that Sakamoto-san was raising - do we have any better ideas to handle larger binary controls than what's currently being done? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124623.html

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[alsa-devel] [PATCH] ASoC: Intel: Add depends on X86
by Vinod Koul 23 Oct '17

23 Oct '17

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Make all Intel audio drivers dependent on X86 to avoid compilation errors for s390 and xtensa architectures. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 1dcc9e456610..a59dda39007e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_INTEL_SST_TOPLEVEL tristate "Intel ASoC SST drivers" + depends on X86 select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5651: Convert rt5651 micbias1 to a supply widget
by Carlo Caione 23 Oct '17

23 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> MICBIAS widget type has been deprecated. Convert it to a SUPPLY widget. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index da60b28ba3df..28f7210cec91 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -883,8 +883,8 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { /* micbias */ SND_SOC_DAPM_SUPPLY("LDO", RT5651_PWR_ANLG1, RT5651_PWR_LDO_BIT, 0, NULL, 0), - SND_SOC_DAPM_MICBIAS("micbias1", RT5651_PWR_ANLG2, - RT5651_PWR_MB1_BIT, 0), + SND_SOC_DAPM_SUPPLY("micbias1", RT5651_PWR_ANLG2, + RT5651_PWR_MB1_BIT, 0, NULL, 0), /* Input Lines */ SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), -- 2.14.2

3 3

[alsa-devel] [asoc:topic/intel 38/38] sound/soc/intel/common/sst-firmware.c:1278:20: error: expected declaration specifiers or '...' before string constant
by kbuild test robot 23 Oct '17

23 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 commit: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 [38/38] ASoC: Intel: clarify Kconfig dependencies config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout f6a118a800e35af2c63f90cbcc23093f4b53b3a2 # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): >> sound/soc/intel/common/sst-firmware.c:1278:20: error: expected declaration specifiers or '...' before string constant MODULE_DESCRIPTION("Intel SST Firmware Loader"); ^ sound/soc/intel/common/sst-firmware.c:1279:16: error: expected declaration specifiers or '...' before string constant MODULE_LICENSE("GPL v2"); ^ -- sound/soc/intel/haswell/sst-haswell-pcm.c: In function 'create_adsp_page_table': >> sound/soc/intel/haswell/sst-haswell-pcm.c:446:2: error: implicit declaration of function 'snd_sgbuf_aligned_pages' [-Werror=implicit-function-declaration] pages = snd_sgbuf_aligned_pages(size); ^ cc1: some warnings being treated as errors vim +1278 sound/soc/intel/common/sst-firmware.c a395bdd6 Takashi Iwai 2016-07-11 1277 a395bdd6 Takashi Iwai 2016-07-11 @1278 MODULE_DESCRIPTION("Intel SST Firmware Loader"); :::::: The code at line 1278 was first introduced by commit :::::: a395bdd6b24b692adbce0df6510ec9f2af57573e ASoC: intel: Fix sst-dsp dependency on dw stuff :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] HDMI LPE audio on Intel Compute Stick 2nd Gen (CHT)
by Ian W MORRISON 22 Oct '17

22 Oct '17

Hi Pierre, I've just noticed that non of the kernels from v4.13 and up run on my Intel Compute Stick (2nd gen Cherry Trail STK1AW32SC) unit and have HDMI LPE audio working. Before I spend anytime looking into this can you confirm if you have the same issue as I just want to rule out a hardware malfunction with my unit first? Thanks, Ian

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[alsa-devel] [asoc:topic/intel 38/38] sound/soc//intel/skylake/skl-sst-cldma.c:92:22: error: implicit declaration of function 'virt_to_phys'
by kbuild test robot 21 Oct '17

21 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 commit: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 [38/38] ASoC: Intel: clarify Kconfig dependencies config: s390-allyesconfig (attached as .config) compiler: s390x-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout f6a118a800e35af2c63f90cbcc23093f4b53b3a2 # save the attached .config to linux build tree make.cross ARCH=s390 All errors (new ones prefixed by >>): sound/soc//intel/skylake/skl-sst-cldma.c: In function 'skl_cldma_setup_bdle': >> sound/soc//intel/skylake/skl-sst-cldma.c:92:22: error: implicit declaration of function 'virt_to_phys' [-Werror=implicit-function-declaration] phys_addr_t addr = virt_to_phys(dmab_data->area + ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/virt_to_phys +92 sound/soc//intel/skylake/skl-sst-cldma.c a4386450b Jeeja KP 2015-12-18 82 914426c86 Vinod Koul 2015-07-10 83 /* Code loader helper APIs */ 914426c86 Vinod Koul 2015-07-10 84 static void skl_cldma_setup_bdle(struct sst_dsp *ctx, 914426c86 Vinod Koul 2015-07-10 85 struct snd_dma_buffer *dmab_data, 914426c86 Vinod Koul 2015-07-10 86 u32 **bdlp, int size, int with_ioc) 914426c86 Vinod Koul 2015-07-10 87 { 914426c86 Vinod Koul 2015-07-10 88 u32 *bdl = *bdlp; 914426c86 Vinod Koul 2015-07-10 89 914426c86 Vinod Koul 2015-07-10 90 ctx->cl_dev.frags = 0; 914426c86 Vinod Koul 2015-07-10 91 while (size > 0) { 914426c86 Vinod Koul 2015-07-10 @92 phys_addr_t addr = virt_to_phys(dmab_data->area + 914426c86 Vinod Koul 2015-07-10 93 (ctx->cl_dev.frags * ctx->cl_dev.bufsize)); 914426c86 Vinod Koul 2015-07-10 94 914426c86 Vinod Koul 2015-07-10 95 bdl[0] = cpu_to_le32(lower_32_bits(addr)); 914426c86 Vinod Koul 2015-07-10 96 bdl[1] = cpu_to_le32(upper_32_bits(addr)); 914426c86 Vinod Koul 2015-07-10 97 914426c86 Vinod Koul 2015-07-10 98 bdl[2] = cpu_to_le32(ctx->cl_dev.bufsize); 914426c86 Vinod Koul 2015-07-10 99 914426c86 Vinod Koul 2015-07-10 100 size -= ctx->cl_dev.bufsize; 914426c86 Vinod Koul 2015-07-10 101 bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); 914426c86 Vinod Koul 2015-07-10 102 914426c86 Vinod Koul 2015-07-10 103 bdl += 4; 914426c86 Vinod Koul 2015-07-10 104 ctx->cl_dev.frags++; 914426c86 Vinod Koul 2015-07-10 105 } 914426c86 Vinod Koul 2015-07-10 106 } 914426c86 Vinod Koul 2015-07-10 107 :::::: The code at line 92 was first introduced by commit :::::: 914426c8657c1e934b015ffa32e7bfd0cb34f47f ASoC: Intel: Skylake: add code loader DMA operations :::::: TO: Vinod Koul <vinod.koul(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/stm32 5/7] sound/soc/stm/stm32_sai_sub.c:467:7: warning: 'cr1' may be used uninitialized in this function
by kbuild test robot 21 Oct '17

21 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/stm32 head: 5914d285f6b782892a91d6621723fdc41a775b15 commit: 61fb4ff70377cd2c49a3487bdb0156eba6930072 [5/7] ASoC: stm32: sai: Move static settings to DAI init config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 61fb4ff70377cd2c49a3487bdb0156eba6930072 # save the attached .config to linux build tree make.cross ARCH=xtensa Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_hw_params': >> sound/soc/stm/stm32_sai_sub.c:467:7: warning: 'cr1' may be used uninitialized in this function [-Wmaybe-uninitialized] cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); ^ sound/soc/stm/stm32_sai_sub.c:451:6: note: 'cr1' was declared here int cr1, cr1_mask, ret; ^ vim +/cr1 +467 sound/soc/stm/stm32_sai_sub.c 3e086edf olivier moysan 2017-04-10 445 3e086edf olivier moysan 2017-04-10 446 static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, 3e086edf olivier moysan 2017-04-10 447 struct snd_pcm_substream *substream, 3e086edf olivier moysan 2017-04-10 448 struct snd_pcm_hw_params *params) 3e086edf olivier moysan 2017-04-10 449 { 3e086edf olivier moysan 2017-04-10 450 struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); 3e086edf olivier moysan 2017-04-10 451 int cr1, cr1_mask, ret; 3e086edf olivier moysan 2017-04-10 452 a4529d2b Olivier Moysan 2017-10-19 453 /* a4529d2b Olivier Moysan 2017-10-19 454 * DMA bursts increment is set to 4 words. a4529d2b Olivier Moysan 2017-10-19 455 * SAI fifo threshold is set to half fifo, to keep enough space a4529d2b Olivier Moysan 2017-10-19 456 * for DMA incoming bursts. a4529d2b Olivier Moysan 2017-10-19 457 */ 3e086edf olivier moysan 2017-04-10 458 regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, 3e086edf olivier moysan 2017-04-10 459 SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, a4529d2b Olivier Moysan 2017-10-19 460 SAI_XCR2_FFLUSH | a4529d2b Olivier Moysan 2017-10-19 461 SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); 3e086edf olivier moysan 2017-04-10 462 3e086edf olivier moysan 2017-04-10 463 /* Mode, data format and channel config */ 61fb4ff7 Olivier Moysan 2017-10-19 464 cr1_mask = SAI_XCR1_DS_MASK; 3e086edf olivier moysan 2017-04-10 465 switch (params_format(params)) { 3e086edf olivier moysan 2017-04-10 466 case SNDRV_PCM_FORMAT_S8: 3e086edf olivier moysan 2017-04-10 @467 cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); 3e086edf olivier moysan 2017-04-10 468 break; 3e086edf olivier moysan 2017-04-10 469 case SNDRV_PCM_FORMAT_S16_LE: 3e086edf olivier moysan 2017-04-10 470 cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_16); 3e086edf olivier moysan 2017-04-10 471 break; 3e086edf olivier moysan 2017-04-10 472 case SNDRV_PCM_FORMAT_S32_LE: 3e086edf olivier moysan 2017-04-10 473 cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_32); 3e086edf olivier moysan 2017-04-10 474 break; 3e086edf olivier moysan 2017-04-10 475 default: 3e086edf olivier moysan 2017-04-10 476 dev_err(cpu_dai->dev, "Data format not supported"); 3e086edf olivier moysan 2017-04-10 477 return -EINVAL; 3e086edf olivier moysan 2017-04-10 478 } 3e086edf olivier moysan 2017-04-10 479 3e086edf olivier moysan 2017-04-10 480 cr1_mask |= SAI_XCR1_MONO; 3e086edf olivier moysan 2017-04-10 481 if ((sai->slots == 2) && (params_channels(params) == 1)) 3e086edf olivier moysan 2017-04-10 482 cr1 |= SAI_XCR1_MONO; 3e086edf olivier moysan 2017-04-10 483 3e086edf olivier moysan 2017-04-10 484 ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); 3e086edf olivier moysan 2017-04-10 485 if (ret < 0) { 3e086edf olivier moysan 2017-04-10 486 dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); 3e086edf olivier moysan 2017-04-10 487 return ret; 3e086edf olivier moysan 2017-04-10 488 } 3e086edf olivier moysan 2017-04-10 489 3e086edf olivier moysan 2017-04-10 490 return 0; 3e086edf olivier moysan 2017-04-10 491 } 3e086edf olivier moysan 2017-04-10 492 :::::: The code at line 467 was first introduced by commit :::::: 3e086edfe0c73daaabd929b926bbe26536272d9a ASoC: stm32: add SAI driver :::::: TO: olivier moysan <olivier.moysan(a)st.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Applied "ASoC: stm32: sai: Fix DMA burst size" to the asoc tree
by Mark Brown 21 Oct '17

21 Oct '17

The patch ASoC: stm32: sai: Fix DMA burst size 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 a4529d2b885747bb9393d507cd28649052d733f3 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Thu, 19 Oct 2017 15:03:19 +0200 Subject: [PATCH] ASoC: stm32: sai: Fix DMA burst size Set best burst size tradeoff for 8, 16, 32 bits transfers. Signed-off-by: olivier moysan <olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 5e66bf310351..dde8ddf9e777 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -446,12 +446,16 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int cr1, cr1_mask, ret; - int fth = STM_SAI_FIFO_TH_HALF; - /* FIFO config */ + /* + * DMA bursts increment is set to 4 words. + * SAI fifo threshold is set to half fifo, to keep enough space + * for DMA incoming bursts. + */ regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, - SAI_XCR2_FFLUSH | SAI_XCR2_FTH_SET(fth)); + SAI_XCR2_FFLUSH | + SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* Mode, data format and channel config */ cr1 = SAI_XCR1_PRTCFG_SET(SAI_FREE_PROTOCOL); @@ -485,10 +489,6 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, return ret; } - /* DMA config */ - sai->dma_params.maxburst = STM_SAI_FIFO_SIZE * fth / sizeof(u32); - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&sai->dma_params); - return 0; } @@ -731,7 +731,12 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX); - sai->dma_params.maxburst = 1; + /* + * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice, + * as it allows bytes, half-word and words transfers. (See DMA fifos + * constraints). + */ + sai->dma_params.maxburst = 4; /* Buswidth will be set by framework at runtime */ sai->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_UNDEFINED; -- 2.14.1

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[alsa-devel] Applied "ASoC: stm32: sai: Move static settings to DAI init" to the asoc tree
by Mark Brown 21 Oct '17

21 Oct '17

The patch ASoC: stm32: sai: Move static settings to DAI init 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 61fb4ff70377cd2c49a3487bdb0156eba6930072 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Thu, 19 Oct 2017 15:03:18 +0200 Subject: [PATCH] ASoC: stm32: sai: Move static settings to DAI init Audio interface direction and protocol settings does not change at runtime. So, these settings are moved from hw_params function to dai_probe and set_fmt. Signed-off-by: olivier moysan <olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index dde8ddf9e777..815ef1036199 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -308,12 +308,15 @@ static int stm32_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); - int cr1 = 0, frcr = 0; - int cr1_mask = 0, frcr_mask = 0; + int cr1, frcr = 0; + int cr1_mask, frcr_mask = 0; int ret; dev_dbg(cpu_dai->dev, "fmt %x\n", fmt); + cr1_mask = SAI_XCR1_PRTCFG_MASK; + cr1 = SAI_XCR1_PRTCFG_SET(SAI_FREE_PROTOCOL); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { /* SCK active high for all protocols */ case SND_SOC_DAIFMT_I2S: @@ -340,7 +343,7 @@ static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return -EINVAL; } - cr1_mask |= SAI_XCR1_PRTCFG_MASK | SAI_XCR1_CKSTR; + cr1_mask |= SAI_XCR1_CKSTR; frcr_mask |= SAI_XFRCR_FSPOL | SAI_XFRCR_FSOFF | SAI_XFRCR_FSDEF; @@ -458,7 +461,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* Mode, data format and channel config */ - cr1 = SAI_XCR1_PRTCFG_SET(SAI_FREE_PROTOCOL); + cr1_mask = SAI_XCR1_DS_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); @@ -473,11 +476,6 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, dev_err(cpu_dai->dev, "Data format not supported"); return -EINVAL; } - cr1_mask = SAI_XCR1_DS_MASK | SAI_XCR1_PRTCFG_MASK; - - cr1_mask |= SAI_XCR1_RX_TX; - if (STM_SAI_IS_CAPTURE(sai)) - cr1 |= SAI_XCR1_RX_TX; cr1_mask |= SAI_XCR1_MONO; if ((sai->slots == 2) && (params_channels(params) == 1)) @@ -729,6 +727,7 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); + int cr1 = 0, cr1_mask; sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX); /* @@ -745,7 +744,11 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) else snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params); - return 0; + cr1_mask = SAI_XCR1_RX_TX; + if (STM_SAI_IS_CAPTURE(sai)) + cr1 |= SAI_XCR1_RX_TX; + + return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); } static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops = { -- 2.14.1

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[alsa-devel] Applied "ASoC: move ACPI common code out of Intel/sst tree" to the asoc tree
by Mark Brown 21 Oct '17

21 Oct '17

The patch ASoC: move ACPI common code out of Intel/sst tree 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 7feb2f786a46d34cac760d5e246dfbdafafa0e96 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 12 Oct 2017 18:49:38 -0500 Subject: [PATCH] ASoC: move ACPI common code out of Intel/sst tree ACPI support is not specific to the Intel/SST driver. Move the enumeration and matching code which is not hardware-dependent to sound/soc and rename relevant sst_acpi_ structures and functions with snd_soc_acpi_ prefix soc-acpi.h is protected by a #ifndef __LINUX_SND_SOC_ACPI_H for consistency with all other SoC .h files: grep -L __LINUX include/sound/soc* | wc -l 0 grep __LINUX include/sound/soc* | wc -l 14 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-acpi.h | 100 +++++++++++++++++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 ++ sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst/sst_acpi.c | 24 ++--- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 +-- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 11 +-- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/intel/common/Makefile | 2 - sound/soc/intel/common/sst-acpi.c | 18 ++-- sound/soc/intel/common/sst-acpi.h | 82 ----------------- sound/soc/intel/skylake/skl.c | 40 ++++----- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 ++++++------ 16 files changed, 203 insertions(+), 172 deletions(-) create mode 100644 include/sound/soc-acpi.h delete mode 100644 sound/soc/intel/common/sst-acpi.h rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h new file mode 100644 index 000000000000..918d78e8c8d6 --- /dev/null +++ b/include/sound/soc-acpi.h @@ -0,0 +1,100 @@ +/* + * Copyright (C) 2013-15, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __LINUX_SND_SOC_ACPI_H +#define __LINUX_SND_SOC_ACPI_H + +#include <linux/stddef.h> +#include <linux/acpi.h> + +struct snd_soc_acpi_package_context { + char *name; /* package name */ + int length; /* number of elements */ + struct acpi_buffer *format; + struct acpi_buffer *state; + bool data_valid; +}; + +#if IS_ENABLED(CONFIG_ACPI) +/* translation fron HID to I2C name, needed for DAI codec_name */ +const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], + struct snd_soc_acpi_package_context *ctx); +#else +static inline const char * +snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +static inline bool +snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], + struct snd_soc_acpi_package_context *ctx) +{ + return false; +} +#endif + +/* acpi match */ +struct snd_soc_acpi_mach * +snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines); + +/* acpi check hid */ +bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); + +/** + * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are + * related to the hardware, except for the firmware and topology file names. + * + * @id: ACPI ID (usually the codec's) used to find a matching machine driver. + * @drv_name: machine driver name + * @fw_filename: firmware file name. + * @board: board name + * @machine_quirk: pointer to quirk, usually based on DMI information when + * ACPI ID alone is not sufficient, wrong or misleading + * @quirk_data: data used to uniquely identify a machine, usually a list of + * audio codecs whose presence if checked with ACPI + * @pdata: intended for platform data or machine specific-ops. This structure + * is not constant since this field may be updated at run-time + */ +/* Descriptor for SST ASoC machine driver */ +struct snd_soc_acpi_mach { + const u8 id[ACPI_ID_LEN]; + const char *drv_name; + const char *fw_filename; + const char *board; + struct snd_soc_acpi_mach * (*machine_quirk)(void *arg); + const void *quirk_data; + void *pdata; +}; + +#define SND_SOC_ACPI_MAX_CODECS 3 + +/** + * struct snd_soc_acpi_codecs: Structure to hold secondary codec information + * apart from the matched one, this data will be passed to the quirk function + * to match with the ACPI detected devices + * + * @num_codecs: number of secondary codecs used in the platform + * @codecs: holds the codec IDs + * + */ +struct snd_soc_acpi_codecs { + int num_codecs; + u8 codecs[SND_SOC_ACPI_MAX_CODECS][ACPI_ID_LEN]; +}; + +/* check all codecs */ +struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg); + +#endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index c0abad2067e1..d22758165496 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,9 @@ config SND_SOC_COMPRESS config SND_SOC_TOPOLOGY bool +config SND_SOC_ACPI + tristate + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/amd/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 39c27a58158d..95f2a73f9aed 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,12 @@ ifneq ($(CONFIG_SND_SOC_AC97_BUS),) snd-soc-core-objs += soc-ac97.o endif +ifneq ($(CONFIG_SND_SOC_ACPI),) +snd-soc-acpi-objs := soc-acpi.o +endif + +obj-$(CONFIG_SND_SOC_ACPI) += snd-soc-acpi.o + obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3c7f554ec30..d03199cc61e0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_INTEL_SST_ACPI config SND_SOC_INTEL_SST_MATCH tristate + select SND_SOC_ACPI if ACPI config SND_SOC_INTEL_HASWELL tristate diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 4d85ead243f1..8d82bc8dc4ac 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -40,9 +40,9 @@ #include <acpi/acpi_bus.h> #include <asm/cpu_device_id.h> #include <asm/iosf_mbi.h> +#include <sound/soc-acpi.h> #include "../sst-mfld-platform.h" #include "../../common/sst-dsp.h" -#include "../../common/sst-acpi.h" #include "sst.h" /* LPE viewpoint addresses */ @@ -284,7 +284,7 @@ static int sst_acpi_probe(struct platform_device *pdev) int ret = 0; struct intel_sst_drv *ctx; const struct acpi_device_id *id; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; struct platform_device *mdev; struct platform_device *plat_dev; struct sst_platform_info *pdata; @@ -296,8 +296,8 @@ static int sst_acpi_probe(struct platform_device *pdev) return -ENODEV; dev_dbg(dev, "for %s\n", id->id); - mach = (struct sst_acpi_mach *)id->driver_data; - mach = sst_acpi_find_machine(mach); + mach = (struct snd_soc_acpi_mach *)id->driver_data; + mach = snd_soc_acpi_find_machine(mach); if (mach == NULL) { dev_err(dev, "No matching machine driver found\n"); return -ENODEV; @@ -435,7 +435,7 @@ static const struct dmi_system_id cht_table[] = { }; -static struct sst_acpi_mach cht_surface_mach = { +static struct snd_soc_acpi_mach cht_surface_mach = { .id = "10EC5640", .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", @@ -443,7 +443,7 @@ static struct sst_acpi_mach cht_surface_mach = { .pdata = &chv_platform_data, }; -static struct sst_acpi_mach byt_thinkpad_10 = { +static struct snd_soc_acpi_mach byt_thinkpad_10 = { .id = "10EC5640", .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_0f28.bin", @@ -451,9 +451,9 @@ static struct sst_acpi_mach byt_thinkpad_10 = { .pdata = &byt_rvp_platform_data, }; -static struct sst_acpi_mach *cht_quirk(void *arg) +static struct snd_soc_acpi_mach *cht_quirk(void *arg) { - struct sst_acpi_mach *mach = arg; + struct snd_soc_acpi_mach *mach = arg; dmi_check_system(cht_table); @@ -463,9 +463,9 @@ static struct sst_acpi_mach *cht_quirk(void *arg) return mach; } -static struct sst_acpi_mach *byt_quirk(void *arg) +static struct snd_soc_acpi_mach *byt_quirk(void *arg) { - struct sst_acpi_mach *mach = arg; + struct snd_soc_acpi_mach *mach = arg; dmi_check_system(byt_table); @@ -476,7 +476,7 @@ static struct sst_acpi_mach *byt_quirk(void *arg) } -static struct sst_acpi_mach sst_acpi_bytcr[] = { +static struct snd_soc_acpi_mach sst_acpi_bytcr[] = { { .id = "10EC5640", .drv_name = "bytcr_rt5640", @@ -560,7 +560,7 @@ static struct sst_acpi_mach sst_acpi_bytcr[] = { }; /* Cherryview-based platforms: CherryTrail and Braswell */ -static struct sst_acpi_mach sst_acpi_chv[] = { +static struct snd_soc_acpi_mach sst_acpi_chv[] = { { .id = "10EC5670", .drv_name = "cht-bsw-rt5672", diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 296b50640f46..c4d82ad41bd7 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -27,9 +27,9 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/da7213.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), @@ -224,7 +224,7 @@ static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ static int bytcht_da7213_probe(struct platform_device *pdev) { struct snd_soc_card *card; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int dai_index = 0; int ret_val = 0; @@ -243,7 +243,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 6acacc0aeedb..8088396717e3 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -29,8 +29,8 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" #include "../common/sst-dsp.h" struct byt_cht_es8316_private { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 3eeb02036aa1..f2c0fc415e52 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -32,9 +32,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5640.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" #include "../common/sst-dsp.h" enum { @@ -737,7 +737,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) { struct byt_rt5640_private *priv; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; int dai_index = 0; @@ -762,7 +762,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(byt_rt5640_codec_name, sizeof(byt_rt5640_codec_name), "%s%s", "i2c-", i2c_name); @@ -796,7 +796,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; struct acpi_buffer state = {0, NULL}; - struct sst_acpi_package_context pkg_ctx; + struct snd_soc_acpi_package_context pkg_ctx; bool pkg_found = false; state.length = sizeof(chan_package); @@ -808,7 +808,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) pkg_ctx.state = &state; pkg_ctx.data_valid = false; - pkg_found = sst_acpi_find_package_from_hid(mach->id, &pkg_ctx); + pkg_found = snd_soc_acpi_find_package_from_hid(mach->id, + &pkg_ctx); if (pkg_found) { if (chan_package.aif_value == 1) { dev_info(&pdev->dev, "BIOS Routing: AIF1 connected\n"); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1dad5c98c9ef..51de1bbe8ecd 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -30,9 +30,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5651.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" enum { BYT_RT5651_DMIC_MAP, @@ -443,7 +443,7 @@ static char byt_rt5651_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { struct byt_rt5651_private *priv; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; int dai_index = 0; @@ -468,7 +468,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(byt_rt5651_codec_name, sizeof(byt_rt5651_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 83c923c39393..18d129caa974 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -32,9 +32,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5645.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI1 "rt5645-aif1" @@ -523,7 +523,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_cht_mc_probe(struct platform_device *pdev) { struct snd_soc_card *card = snd_soc_cards[0].soc_card; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; struct cht_mc_private *drv; const char *i2c_name = NULL; bool found = false; @@ -566,7 +566,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "%s%s", "i2c-", i2c_name); @@ -599,7 +599,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; struct acpi_buffer state = {0, NULL}; - struct sst_acpi_package_context pkg_ctx; + struct snd_soc_acpi_package_context pkg_ctx; bool pkg_found = false; state.length = sizeof(chan_package); @@ -611,7 +611,8 @@ static int snd_cht_mc_probe(struct platform_device *pdev) pkg_ctx.state = &state; pkg_ctx.data_valid = false; - pkg_found = sst_acpi_find_package_from_hid(mach->id, &pkg_ctx); + pkg_found = snd_soc_acpi_find_package_from_hid(mach->id, + &pkg_ctx); if (pkg_found) { if (chan_package.aif_value == 1) { dev_info(&pdev->dev, "BIOS Routing: AIF1 connected\n"); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index d135640702e9..f8f21eee9b2d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -24,9 +24,10 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5670.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" + /* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ #define CHT_PLAT_CLK_3_HZ 19200000 @@ -383,7 +384,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) { int ret_val = 0; struct cht_mc_private *drv; - struct sst_acpi_mach *mach = pdev->dev.platform_data; + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; const char *i2c_name; int i; @@ -395,7 +396,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* fixup codec name based on HID */ if (mach) { - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", i2c_name); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 1a35149bcad7..2c958ec3f4fd 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,10 +1,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-match-objs := sst-match-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-firmware-objs := sst-firmware.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o -obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o obj-$(CONFIG_SND_SOC_INTEL_SST_FIRMWARE) += snd-soc-sst-firmware.o diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index 1285cc597b6b..9866998de8be 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -21,7 +21,7 @@ #include <linux/platform_device.h> #include "sst-dsp.h" -#include "sst-acpi.h" +#include <sound/soc-acpi.h> #define SST_LPT_DSP_DMA_ADDR_OFFSET 0x0F0000 #define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 @@ -30,7 +30,7 @@ /* Descriptor for setting up SST platform data */ struct sst_acpi_desc { const char *drv_name; - struct sst_acpi_mach *machines; + struct snd_soc_acpi_mach *machines; /* Platform resource indexes. Must set to -1 if not used */ int resindex_lpe_base; int resindex_pcicfg_base; @@ -49,7 +49,7 @@ struct sst_acpi_priv { struct platform_device *pdev_pcm; struct sst_pdata sst_pdata; struct sst_acpi_desc *desc; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; }; static void sst_acpi_fw_cb(const struct firmware *fw, void *context) @@ -59,7 +59,7 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; struct sst_acpi_desc *desc = sst_acpi->desc; - struct sst_acpi_mach *mach = sst_acpi->mach; + struct snd_soc_acpi_mach *mach = sst_acpi->mach; sst_pdata->fw = fw; if (!fw) { @@ -85,7 +85,7 @@ static int sst_acpi_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct sst_acpi_priv *sst_acpi; struct sst_pdata *sst_pdata; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; struct sst_acpi_desc *desc; struct resource *mmio; int ret = 0; @@ -99,7 +99,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return -ENODEV; desc = (struct sst_acpi_desc *)id->driver_data; - mach = sst_acpi_find_machine(desc->machines); + mach = snd_soc_acpi_find_machine(desc->machines); if (mach == NULL) { dev_err(dev, "No matching ASoC machine driver found\n"); return -ENODEV; @@ -179,7 +179,7 @@ static int sst_acpi_remove(struct platform_device *pdev) return 0; } -static struct sst_acpi_mach haswell_machines[] = { +static struct snd_soc_acpi_mach haswell_machines[] = { { "INT33CA", "haswell-audio", "intel/IntcSST1.bin", NULL, NULL, NULL }, {} }; @@ -197,7 +197,7 @@ static struct sst_acpi_desc sst_acpi_haswell_desc = { .dma_size = SST_LPT_DSP_DMA_SIZE, }; -static struct sst_acpi_mach broadwell_machines[] = { +static struct snd_soc_acpi_mach broadwell_machines[] = { { "INT343A", "broadwell-audio", "intel/IntcSST2.bin", NULL, NULL, NULL }, { "RT5677CE", "bdw-rt5677", "intel/IntcSST2.bin", NULL, NULL, NULL }, {} @@ -217,7 +217,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { }; #if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) -static struct sst_acpi_mach baytrail_machines[] = { +static struct snd_soc_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, {} diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h deleted file mode 100644 index afe9b87b8bd5..000000000000 --- a/sound/soc/intel/common/sst-acpi.h +++ /dev/null @@ -1,82 +0,0 @@ -/* - * Copyright (C) 2013-15, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - */ - -#include <linux/stddef.h> -#include <linux/acpi.h> - -struct sst_acpi_package_context { - char *name; /* package name */ - int length; /* number of elements */ - struct acpi_buffer *format; - struct acpi_buffer *state; - bool data_valid; -}; - -#if IS_ENABLED(CONFIG_ACPI) -/* translation fron HID to I2C name, needed for DAI codec_name */ -const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); -bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], - struct sst_acpi_package_context *ctx); -#else -static inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) -{ - return NULL; -} -static inline bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], - struct sst_acpi_package_context *ctx) -{ - return false; -} -#endif - -/* acpi match */ -struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); - -/* acpi check hid */ -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); - -/* Descriptor for SST ASoC machine driver */ -struct sst_acpi_mach { - /* ACPI ID for the matching machine driver. Audio codec for instance */ - const u8 id[ACPI_ID_LEN]; - /* machine driver name */ - const char *drv_name; - /* firmware file name */ - const char *fw_filename; - - /* board name */ - const char *board; - struct sst_acpi_mach * (*machine_quirk)(void *arg); - const void *quirk_data; - void *pdata; -}; - -#define SST_ACPI_MAX_CODECS 3 - -/** - * struct sst_codecs: Structure to hold secondary codec information apart from - * the matched one, this data will be passed to the quirk function to match - * with the ACPI detected devices - * - * @num_codecs: number of secondary codecs used in the platform - * @codecs: holds the codec IDs - * - */ -struct sst_codecs { - int num_codecs; - u8 codecs[SST_ACPI_MAX_CODECS][ACPI_ID_LEN]; -}; - -/* check all codecs */ -struct sst_acpi_mach *sst_acpi_codec_list(void *arg); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 4abbd90e394c..31d8634e8aa1 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -28,7 +28,7 @@ #include <linux/firmware.h> #include <linux/delay.h> #include <sound/pcm.h> -#include "../common/sst-acpi.h" +#include <sound/soc-acpi.h> #include <sound/hda_register.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> @@ -439,10 +439,10 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data) { struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct platform_device *pdev; - struct sst_acpi_mach *mach = driver_data; + struct snd_soc_acpi_mach *mach = driver_data; int ret; - mach = sst_acpi_find_machine(mach); + mach = snd_soc_acpi_find_machine(mach); if (mach == NULL) { dev_err(bus->dev, "No matching machine driver found\n"); return -ENODEV; @@ -878,27 +878,27 @@ static void skl_remove(struct pci_dev *pci) dev_set_drvdata(&pci->dev, NULL); } -static struct sst_codecs skl_codecs = { +static struct snd_soc_acpi_codecs skl_codecs = { .num_codecs = 1, .codecs = {"10508825"} }; -static struct sst_codecs kbl_codecs = { +static struct snd_soc_acpi_codecs kbl_codecs = { .num_codecs = 1, .codecs = {"10508825"} }; -static struct sst_codecs bxt_codecs = { +static struct snd_soc_acpi_codecs bxt_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} }; -static struct sst_codecs kbl_poppy_codecs = { +static struct snd_soc_acpi_codecs kbl_poppy_codecs = { .num_codecs = 1, .codecs = {"10EC5663"} }; -static struct sst_codecs kbl_5663_5514_codecs = { +static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { .num_codecs = 2, .codecs = {"10EC5663", "10EC5514"} }; @@ -907,7 +907,7 @@ static struct skl_machine_pdata cnl_pdata = { .use_tplg_pcm = true, }; -static struct sst_acpi_mach sst_skl_devdata[] = { +static struct snd_soc_acpi_mach sst_skl_devdata[] = { { .id = "INT343A", .drv_name = "skl_alc286s_i2s", @@ -917,7 +917,7 @@ static struct sst_acpi_mach sst_skl_devdata[] = { .id = "INT343B", .drv_name = "skl_n88l25_s4567", .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &skl_codecs, .pdata = &skl_dmic_data }, @@ -925,14 +925,14 @@ static struct sst_acpi_mach sst_skl_devdata[] = { .id = "MX98357A", .drv_name = "skl_n88l25_m98357a", .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &skl_codecs, .pdata = &skl_dmic_data }, {} }; -static struct sst_acpi_mach sst_bxtp_devdata[] = { +static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { { .id = "INT343A", .drv_name = "bxt_alc298s_i2s", @@ -942,13 +942,13 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = { .id = "DLGS7219", .drv_name = "bxt_da7219_max98357a_i2s", .fw_filename = "intel/dsp_fw_bxtn.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &bxt_codecs, }, {} }; -static struct sst_acpi_mach sst_kbl_devdata[] = { +static struct snd_soc_acpi_mach sst_kbl_devdata[] = { { .id = "INT343A", .drv_name = "kbl_alc286s_i2s", @@ -958,7 +958,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "INT343B", .drv_name = "kbl_n88l25_s4567", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_codecs, .pdata = &skl_dmic_data }, @@ -966,7 +966,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "MX98357A", .drv_name = "kbl_n88l25_m98357a", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_codecs, .pdata = &skl_dmic_data }, @@ -974,7 +974,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "MX98927", .drv_name = "kbl_r5514_5663_max", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_5663_5514_codecs, .pdata = &skl_dmic_data }, @@ -982,7 +982,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "MX98927", .drv_name = "kbl_rt5663_m98927", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_poppy_codecs, .pdata = &skl_dmic_data }, @@ -995,7 +995,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { {} }; -static struct sst_acpi_mach sst_glk_devdata[] = { +static struct snd_soc_acpi_mach sst_glk_devdata[] = { { .id = "INT343A", .drv_name = "glk_alc298s_i2s", @@ -1004,7 +1004,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = { {} }; -static const struct sst_acpi_mach sst_cnl_devdata[] = { +static const struct snd_soc_acpi_mach sst_cnl_devdata[] = { { .id = "INT34C2", .drv_name = "cnl_rt274", diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/soc-acpi.c similarity index 63% rename from sound/soc/intel/common/sst-match-acpi.c rename to sound/soc/soc-acpi.c index 56d26f36a3cb..f21df28bc28e 100644 --- a/sound/soc/intel/common/sst-match-acpi.c +++ b/sound/soc/soc-acpi.c @@ -1,5 +1,5 @@ /* - * sst_match_apci.c - SST (LPE) match for ACPI enumeration. + * soc-apci.c - support for ACPI enumeration. * * Copyright (c) 2013-15, Intel Corporation. * @@ -14,9 +14,9 @@ * more details. */ -#include "sst-acpi.h" +#include <sound/soc-acpi.h> -static acpi_status sst_acpi_find_name(acpi_handle handle, u32 level, +static acpi_status snd_soc_acpi_find_name(acpi_handle handle, u32 level, void *context, void **ret) { struct acpi_device *adev; @@ -34,12 +34,12 @@ static acpi_status sst_acpi_find_name(acpi_handle handle, u32 level, return AE_OK; } -const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) { const char *name = NULL; acpi_status status; - status = acpi_get_devices(hid, sst_acpi_find_name, NULL, + status = acpi_get_devices(hid, snd_soc_acpi_find_name, NULL, (void **)&name); if (ACPI_FAILURE(status) || name[0] == '\0') @@ -47,9 +47,9 @@ const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) return name; } -EXPORT_SYMBOL_GPL(sst_acpi_find_name_from_hid); +EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid); -static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, +static acpi_status snd_soc_acpi_mach_match(acpi_handle handle, u32 level, void *context, void **ret) { unsigned long long sta; @@ -63,26 +63,27 @@ static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, return AE_OK; } -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) +bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) { acpi_status status; bool found = false; - status = acpi_get_devices(hid, sst_acpi_mach_match, &found, NULL); + status = acpi_get_devices(hid, snd_soc_acpi_mach_match, &found, NULL); if (ACPI_FAILURE(status)) return false; return found; } -EXPORT_SYMBOL_GPL(sst_acpi_check_hid); +EXPORT_SYMBOL_GPL(snd_soc_acpi_check_hid); -struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) +struct snd_soc_acpi_mach * +snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (sst_acpi_check_hid(mach->id) == true) { + if (snd_soc_acpi_check_hid(mach->id) == true) { if (mach->machine_quirk == NULL) return mach; @@ -92,14 +93,14 @@ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) } return NULL; } -EXPORT_SYMBOL_GPL(sst_acpi_find_machine); +EXPORT_SYMBOL_GPL(snd_soc_acpi_find_machine); -static acpi_status sst_acpi_find_package(acpi_handle handle, u32 level, - void *context, void **ret) +static acpi_status snd_soc_acpi_find_package(acpi_handle handle, u32 level, + void *context, void **ret) { struct acpi_device *adev; acpi_status status = AE_OK; - struct sst_acpi_package_context *pkg_ctx = context; + struct snd_soc_acpi_package_context *pkg_ctx = context; pkg_ctx->data_valid = false; @@ -137,37 +138,38 @@ static acpi_status sst_acpi_find_package(acpi_handle handle, u32 level, return AE_OK; } -bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], - struct sst_acpi_package_context *ctx) +bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], + struct snd_soc_acpi_package_context *ctx) { acpi_status status; - status = acpi_get_devices(hid, sst_acpi_find_package, ctx, NULL); + status = acpi_get_devices(hid, snd_soc_acpi_find_package, ctx, NULL); if (ACPI_FAILURE(status) || !ctx->data_valid) return false; return true; } -EXPORT_SYMBOL_GPL(sst_acpi_find_package_from_hid); +EXPORT_SYMBOL_GPL(snd_soc_acpi_find_package_from_hid); -struct sst_acpi_mach *sst_acpi_codec_list(void *arg) +struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) { - struct sst_acpi_mach *mach = arg; - struct sst_codecs *codec_list = (struct sst_codecs *) mach->quirk_data; + struct snd_soc_acpi_mach *mach = arg; + struct snd_soc_acpi_codecs *codec_list = + (struct snd_soc_acpi_codecs *) mach->quirk_data; int i; if (mach->quirk_data == NULL) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (sst_acpi_check_hid(codec_list->codecs[i]) != true) + if (snd_soc_acpi_check_hid(codec_list->codecs[i]) != true) return NULL; } return mach; } -EXPORT_SYMBOL_GPL(sst_acpi_codec_list); +EXPORT_SYMBOL_GPL(snd_soc_acpi_codec_list); MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); +MODULE_DESCRIPTION("ALSA SoC ACPI module"); -- 2.14.1

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[alsa-devel] [RFC PATCH 0/8] Prepare Sound Open Firmware (SOF) driver integration
by Pierre-Louis Bossart 21 Oct '17

21 Oct '17

The Sound Open Firmware work [1] is progressing quickly and drivers for the 1.0 release will soon be submitted upstream. Before we do so, there is a new to reshuffle the ACPI integration and Intel Kconfig/Makefiles. This RFC is a preview of what's coming, feedback would be greatly appreciated. We already have three incompatible drivers (BYT/HSW/BDW, Atom/BYT/CHT, SKL+) for historical reasons, blindly adding yet another one would be silly. We need to avoid having the same ACPI routines and machine tables duplicated in 3 locations (soc/intel/common, soc/atom/sst and the upcoming soc/sof). The ACPI routines should be part of sound/soc since there is no real hardware dependency (ACPI is not limited to Intel). The ACPI table definitions and quirks are moved to a soc/intel/common module so that when new HIDs or quirks are added "legacy" and the SOF drivers are updated at once. The Makefiles/Kconfig are also modified to select the SOC(s) first and the machine driver(s) second. The existing code only exposes machine drivers which precludes reuse of machine drivers when SOF platform drivers are selected. SOF drivers will be exposed in parallel to the 'Intel SST' ones (with an exclusive choice needed), but machine drivers will be shared. These patches apply on top of broonie/for-next + all the patches I submitted this week (no dependency on SOF). For people with allergic reactions to patches over email, the code is here: https://github.com/plbossart/sound/commits/topic/acpi-updates This patchset does not address the case where two drivers are selected for the same ACPI HID. I just don't know of any technical solution to that problem. [1] https://www.alsa-project.org/main/index.php/Firmware Pierre-Louis Bossart (8): ASoC: move ACPI common code out of Intel/sst tree ASoC: Intel: common: use c99 syntax for ACPI/machine tables ASoC: ACPI: add new fields for SOF support ASoC: Intel: move all ACPI match tables to common module ASoC: Intel: add SOF firmare/topology file information ASoC: Intel: move machine drivers to dedicated KConfig ASoC: Intel: reorder boards Kconfig by chronological order ASoC: Intel: clarify Kconfig dependencies include/sound/soc-acpi-intel-match.h | 32 ++ include/sound/soc-acpi.h | 107 ++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 + sound/soc/intel/Kconfig | 299 ++------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 319 ++-------------- sound/soc/intel/boards/Kconfig | 286 +++++++++++++++ sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 +- sound/soc/intel/boards/bytcr_rt5651.c | 8 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 11 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/intel/common/Makefile | 4 +- sound/soc/intel/common/soc-acpi-intel-match.c | 400 +++++++++++++++++++++ sound/soc/intel/common/sst-acpi.c | 36 +- sound/soc/intel/common/sst-acpi.h | 82 ----- sound/soc/intel/skylake/skl.c | 40 +-- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 +-- 20 files changed, 965 insertions(+), 752 deletions(-) create mode 100644 include/sound/soc-acpi-intel-match.h create mode 100644 include/sound/soc-acpi.h create mode 100644 sound/soc/intel/boards/Kconfig create mode 100644 sound/soc/intel/common/soc-acpi-intel-match.c delete mode 100644 sound/soc/intel/common/sst-acpi.h rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%) -- 2.9.3

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[alsa-devel] [PATCH 0/8] ASoC: Prepare Sound Open Firmware (SOF) driver integration
by Pierre-Louis Bossart 21 Oct '17

21 Oct '17

The Sound Open Firmware work [1] is progressing quickly and drivers for the 1.0 release will soon be submitted upstream. Before we do so, there is a new to reshuffle the ACPI integration and Intel Kconfig/Makefiles. We already have three incompatible drivers (BYT/HSW/BDW, Atom/BYT/CHT, SKL+) for historical reasons, blindly adding yet another one would be silly. We need to avoid having the same ACPI routines and machine tables duplicated in 3 locations (soc/intel/common, soc/atom/sst and the upcoming soc/sof). The ACPI routines should be part of sound/soc since there is no real hardware dependency (ACPI is not limited to Intel). The ACPI table definitions and quirks are moved to a soc/intel/common module so that when new HIDs or quirks are added "legacy" and the SOF drivers are updated at once. The Makefiles/Kconfig are also modified to select the SOC(s) first and the machine driver(s) second. The existing code only exposes machine drivers which precludes reuse of machine drivers when SOF platform drivers are selected. SOF drivers will be exposed in parallel to the 'Intel SST' ones (with an exclusive choice needed), but machine drivers will be shared. These patches apply on top of broonie/for-next + all the patches I submitted this week (no dependency on SOF). For people with allergic reactions to patches over email, the code is here: https://github.com/plbossart/sound/commits/topic/acpi-updates6 This patchset does not address the case where two drivers are selected for the same ACPI HID. I just don't know of any technical solution to that problem. changes since initial RFC: rebased to v4.14-rc2, used kernel style doc for machine descriptors, removed useless fields, split tables for each SOC corrected missing newlines Pierre-Louis Bossart (8): ASoC: move ACPI common code out of Intel/sst tree ASoC: Intel: common: use c99 syntax for ACPI/machine tables ASoC: ACPI: add new fields for SOF support ASoC: Intel: move all ACPI match tables to common module ASoC: Intel: add SOF firmare/topology file information ASoC: Intel: move machine drivers to dedicated KConfig ASoC: Intel: reorder boards Kconfig by chronological order ASoC: Intel: clarify Kconfig dependencies include/sound/soc-acpi-intel-match.h | 32 +++ .../common/sst-acpi.h => include/sound/soc-acpi.h | 75 +++-- sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 + sound/soc/intel/Kconfig | 299 ++----------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 319 ++------------------- sound/soc/intel/boards/Kconfig | 286 ++++++++++++++++++ sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 11 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/intel/common/Makefile | 4 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 196 +++++++++++++ sound/soc/intel/common/soc-acpi-intel-cht-match.c | 194 +++++++++++++ .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 64 +++++ sound/soc/intel/common/sst-acpi.c | 36 +-- sound/soc/intel/skylake/skl.c | 40 +-- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 ++-- 21 files changed, 963 insertions(+), 692 deletions(-) create mode 100644 include/sound/soc-acpi-intel-match.h rename sound/soc/intel/common/sst-acpi.h => include/sound/soc-acpi.h (34%) create mode 100644 sound/soc/intel/boards/Kconfig create mode 100644 sound/soc/intel/common/soc-acpi-intel-byt-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cht-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%) -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: audio-graph-scu-card: tidyup for Example
by Kuninori Morimoto 21 Oct '17

21 Oct '17

Hi Mark These are not a big deal, but tidyup patches for audio-graph-scu-card Example. Kuninori Morimoto (2): ASoC: audio-graph-scu-card: add missing Capture routing on Example ASoC: audio-graph-scu-card: remove unnecessary route patch from Example 1 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: Intel: sst: remove redundant variable dma_dev_name
by Colin King 21 Oct '17

21 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The pointer dma_dev_name is assigned but never read, it is redundant and can therefore be removed. Cleans up clang warning: sound/soc/intel/common/sst-firmware.c:288:3: warning: Value stored to 'dma_dev_name' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/common/sst-firmware.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index a086c35f91bb..79a9fdf94d38 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -274,7 +274,6 @@ int sst_dma_new(struct sst_dsp *sst) struct sst_pdata *sst_pdata = sst->pdata; struct sst_dma *dma; struct resource mem; - const char *dma_dev_name; int ret = 0; if (sst->pdata->resindex_dma_base == -1) @@ -285,7 +284,6 @@ int sst_dma_new(struct sst_dsp *sst) * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { case SST_DMA_TYPE_DW: - dma_dev_name = "dw_dmac"; break; default: dev_err(sst->dev, "error: invalid DMA engine %d\n", -- 2.14.1

3 2

[alsa-devel] Driver-alsa for Icon Producer Works-192
by Андрей Король 20 Oct '17

20 Oct '17

Hello ! Is it possible to support the sound card Icon Producer Works-192 in the alsa? More about the map: VIA ICEnsemble ICE1724HT (Envy24HT/PT) http://soundcard.hw-specs.com/ICON_Producer_192/ https://nson.org/komplektuyushhie/zvukovye-karty/icon-producer-works-192x.h…

1 0

Re: [alsa-devel] Add support for ALC236
by Takashi Iwai 20 Oct '17

20 Oct '17

On Fri, 20 Oct 2017 09:13:44 +0200, Kailang wrote: > > Hi Takashi, > > Attach patch was support ALC236. Applied, thanks. Takashi

1 0