November 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: usb-audio: uac1: Invalidate ctl on interrupt
by Julian Scheel 17 Nov '17

17 Nov '17

When an interrupt occurs, the value of at least one of the belonging controls should have changed. To make sure they get re-read from device on the next read, invalidate the cache. This was correctly implemented for uac2 already, but missing for uac1. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 91bc8f18791e..1beb5b4d027e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2330,9 +2330,14 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + struct usb_mixer_elem_info *info = + (struct usb_mixer_elem_info *)list; + /* invalidate cache, so the value is read from the device */ + info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &list->kctl->id); + } } static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, -- 2.15.0

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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

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[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

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