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[alsa-devel] [PATCH] ALSA: oss: consolidate kmalloc/memset 0 call to kzalloc
by Nicholas Mc Guire 18 Dec '15

18 Dec '15

This is an API consolidation only. The use of kmalloc + memset to 0 is equivalent to kzalloc. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Found by coccinelle script (relaxed version of scripts/coccinelle/api/alloc/kzalloc-simple.cocci) Patch was compile tested with: x86_64_defconfig CONFIG_SND_PCM_OSS=y Patch is against linux-next (localversion-next is -next-20150518) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e557dbe..8cdd06f 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -850,7 +850,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -EINTR; - sw_params = kmalloc(sizeof(*sw_params), GFP_KERNEL); + sw_params = kmalloc(sizeof(*sw_params), GFP_KERNEL); params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); if (!sw_params || !params || !sparams) { @@ -988,7 +988,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) goto failure; } - memset(sw_params, 0, sizeof(*sw_params)); if (runtime->oss.trigger) { sw_params->start_threshold = 1; } else { -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2 1/4] ASoC: sun4i-codec: Rename codec dapm widgets and routes
by Hans de Goede 18 Dec '15

18 Dec '15

Rename the codec dapm widgets and routes with a _codec prefix. This is a preparation patch for adding card dapm widgets and routes. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: -New patch in v2 of this patchset --- sound/soc/sunxi/sun4i-codec.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 7a3fe1d..519ccb3 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -532,7 +532,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0), }; -static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { +static const struct snd_soc_dapm_widget sun4i_codec_codec_dapm_widgets[] = { /* Digital parts of the ADCs */ SND_SOC_DAPM_SUPPLY("ADC", SUN4I_CODEC_ADC_FIFOC, SUN4I_CODEC_ADC_FIFOC_EN_AD, 0, @@ -589,7 +589,7 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Left"), }; -static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { +static const struct snd_soc_dapm_route sun4i_codec_codec_dapm_routes[] = { /* Left ADC / DAC Routes */ { "Left ADC", NULL, "ADC" }, { "Left DAC", NULL, "DAC" }, @@ -628,10 +628,10 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { static struct snd_soc_codec_driver sun4i_codec_codec = { .controls = sun4i_codec_widgets, .num_controls = ARRAY_SIZE(sun4i_codec_widgets), - .dapm_widgets = sun4i_codec_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), - .dapm_routes = sun4i_codec_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), + .dapm_widgets = sun4i_codec_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_codec_dapm_widgets), + .dapm_routes = sun4i_codec_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_codec_dapm_routes), }; static const struct snd_soc_component_driver sun4i_codec_component = { -- 2.5.0

4 12

[alsa-devel] [PATCH] ALSA: consolidate allocation API
by Nicholas Mc Guire 18 Dec '15

18 Dec '15

This is an API consolidation only. The use of kmalloc + memset to 0 is equivalent to kzalloc. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Found by coccinelle script (relaxed version of scripts/coccinelle/api/alloc/kzalloc-simple.cocci) Patch was compile tested with: x86_64_defconfig CONFIG_SND_SEQUENCER=y, CONFIG_SND_SEQUENCER_OSS=y Patch is against linux-next (localversion-next is -next-20150518) sound/core/seq/oss/seq_oss_init.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b1221b2..9babc9e 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -79,7 +79,7 @@ snd_seq_oss_create_client(void) struct snd_seq_port_info *port; struct snd_seq_port_callback port_callback; - port = kmalloc(sizeof(*port), GFP_KERNEL); + port = kzalloc(sizeof(*port), GFP_KERNEL); if (!port) { rc = -ENOMEM; goto __error; @@ -94,7 +94,6 @@ snd_seq_oss_create_client(void) system_client = rc; /* create annoucement receiver port */ - memset(port, 0, sizeof(*port)); strcpy(port->name, "Receiver"); port->addr.client = system_client; port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* receive only */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH v3] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 18 Dec '15

18 Dec '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. This is a hardware bug/errata and reset is the only option. According to the reference manual, the software reset doesn't reset any control register but only internal hardware logics such as bit clock generator, status flags, and FIFO pointers. (Our purpose is just to reset the clock generator while the software reset is the only way to do that.) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. For asynchronous mode, TX will not be reset when RX is still running. In this case, i can't reproduce this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- v1->v2: 1) refine the commit log and comments 2) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. v2->v3: add a simple comment in code to describe the hardware bug. sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8832901..fef264d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -564,6 +564,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* + * For sai master mode, after several open/close sai, + * there will be no frame clock, and can't recover + * anymore. Add software reset to fix this issue. + * This is a hardware bug, and will be fix in the + * next sai version. + */ + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Mimic legacy behaviour of reading controls when DSP is on
by Charles Keepax 18 Dec '15

18 Dec '15

Older firmwares don't specify access flags for the controls, unfortunately the usage of some of these firmware relies on being able to read back values from the DSP. The current control code will only do this for volatile controls. This patch will read the control from the hardware if no flags are specified and the control is currently enabled, which should cover these legacy use-cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b083642..d1e0826 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -666,6 +666,9 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, else ret = -EPERM; } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + memcpy(p, ctl->cache, ctl->len); } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: fix VAG power up timing
by Jean-Michel Hautbois 18 Dec '15

18 Dec '15

When power up, a "pop" is heard on line-in and mic-in. An analysis of the PCM shows it lasts ~400ms and looks like a filter response. VAG power up should be delayed by 400ms as VAG power down is. Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois(a)veo-labs.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f540f82..08b4046 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -189,6 +189,7 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + msleep(400); break; case SND_SOC_DAPM_PRE_PMD: -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: enable to use tdm_slot on DT
by Kuninori Morimoto 18 Dec '15

18 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound driver will use tdm slot on TDM Multi Mode support. This patch enables tdm slot on rsrc card driver on DT. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index a3ec13f..3c308e2 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -50,6 +50,10 @@ MODULE_DEVICE_TABLE(of, rsrc_card_of_match); struct rsrc_card_dai { unsigned int fmt; unsigned int sysclk; + unsigned int tx_slot_mask; + unsigned int rx_slot_mask; + int slots; + int slot_width; struct clk *clk; char dai_name[DAI_NAME_NUM]; }; @@ -126,6 +130,18 @@ static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd) } } + if (dai_props->slots) { + ret = snd_soc_dai_set_tdm_slot(dai, + dai_props->tx_slot_mask, + dai_props->rx_slot_mask, + dai_props->slots, + dai_props->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "set_tdm_slot error\n"); + goto err; + } + } + ret = 0; err: @@ -198,6 +214,15 @@ static int rsrc_card_parse_links(struct device_node *np, if (ret) return ret; + /* Parse TDM slot */ + ret = snd_soc_of_parse_tdm_slot(np, + &dai_props->tx_slot_mask, + &dai_props->rx_slot_mask, + &dai_props->slots, + &dai_props->slot_width); + if (ret) + return ret; + if (is_fe) { /* BE is dummy */ dai_link->codec_of_node = NULL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: Remove support for setting differing DAI formats
by Kuninori Morimoto 18 Dec '15

18 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 1efb53a220 ("ASoC: simple-card: Remove support for setting differing DAI formats") removed set_fmt support from simple-card. rsrc-card follows same style, because it is based on simple-card. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 3c308e2..9f522ba 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -48,7 +48,6 @@ MODULE_DEVICE_TABLE(of, rsrc_card_of_match); #define DAI_NAME_NUM 32 struct rsrc_card_dai { - unsigned int fmt; unsigned int sysclk; unsigned int tx_slot_mask; unsigned int rx_slot_mask; @@ -114,14 +113,6 @@ static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd) rtd->cpu_dai : rtd->codec_dai; - if (dai_props->fmt) { - ret = snd_soc_dai_set_fmt(dai, dai_props->fmt); - if (ret && ret != -ENOTSUPP) { - dev_err(dai->dev, "set_fmt error\n"); - goto err; - } - } - if (dai_props->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, dai_props->sysclk, 0); if (ret && ret != -ENOTSUPP) { @@ -168,7 +159,7 @@ static int rsrc_card_parse_daifmt(struct device_node *node, struct rsrc_card_priv *priv, int idx, bool is_fe) { - struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; struct device_node *codec = is_fe ? NULL : np; @@ -188,7 +179,7 @@ static int rsrc_card_parse_daifmt(struct device_node *node, daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; - dai_props->fmt = daifmt; + dai_link->dai_fmt = daifmt; of_node_put(bitclkmaster); of_node_put(framemaster); @@ -340,6 +331,7 @@ static int rsrc_card_dai_link_of(struct device_node *node, int idx) { struct device *dev = rsrc_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, idx); bool is_fe = false; int ret; @@ -361,7 +353,7 @@ static int rsrc_card_dai_link_of(struct device_node *node, dev_dbg(dev, "\t%s / %04x / %d\n", dai_props->dai_name, - dai_props->fmt, + dai_link->dai_fmt, dai_props->sysclk); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: tidyup dai format for DPCM
by Kuninori Morimoto 18 Dec '15

18 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsrc-card is DPCM supported version of simple-card. Thus it has similar DT format. OTOH, snd_soc_dai_link requests cpu/codec, but one of them will be snd-soc-dummy in DPCM case, and DPCM requests frontend/backend dai_link. This means it might have multi backend/codec. And, SND_SOC_DAIFMT_xxx is based on "codec". Because of these difference, current rsrc card can't detect correct dai_fmt. This patch detect correct dai fmt from 1st "codec". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 82 +++++++++++++++++++++++++++++-------------- 1 file changed, 55 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 9f522ba..5fe0b51 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -155,14 +155,13 @@ static int rsrc_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, } static int rsrc_card_parse_daifmt(struct device_node *node, - struct device_node *np, + struct device_node *codec, struct rsrc_card_priv *priv, - int idx, bool is_fe) + struct snd_soc_dai_link *dai_link, + unsigned int *retfmt) { - struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; - struct device_node *codec = is_fe ? NULL : np; unsigned int daifmt; daifmt = snd_soc_of_parse_daifmt(node, NULL, @@ -179,11 +178,11 @@ static int rsrc_card_parse_daifmt(struct device_node *node, daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; - dai_link->dai_fmt = daifmt; - of_node_put(bitclkmaster); of_node_put(framemaster); + *retfmt = daifmt; + return 0; } @@ -325,24 +324,16 @@ static int rsrc_card_parse_clk(struct device_node *np, return 0; } -static int rsrc_card_dai_link_of(struct device_node *node, - struct device_node *np, - struct rsrc_card_priv *priv, - int idx) +static int rsrc_card_dai_sub_link_of(struct device_node *node, + struct device_node *np, + struct rsrc_card_priv *priv, + int idx, bool is_fe) { struct device *dev = rsrc_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, idx); - bool is_fe = false; int ret; - if (0 == strcmp(np->name, "cpu")) - is_fe = true; - - ret = rsrc_card_parse_daifmt(node, np, priv, idx, is_fe); - if (ret < 0) - return ret; - ret = rsrc_card_parse_links(np, priv, idx, is_fe); if (ret < 0) return ret; @@ -359,6 +350,48 @@ static int rsrc_card_dai_link_of(struct device_node *node, return ret; } +static int rsrc_card_dai_link_of(struct device_node *node, + struct rsrc_card_priv *priv) +{ + struct snd_soc_dai_link *dai_link; + struct device_node *np; + unsigned int daifmt = 0; + int ret, i; + bool is_fe; + + /* find 1st codec */ + i = 0; + for_each_child_of_node(node, np) { + dai_link = rsrc_priv_to_link(priv, i); + + if (strcmp(np->name, "codec") == 0) { + ret = rsrc_card_parse_daifmt(node, np, priv, + dai_link, &daifmt); + if (ret < 0) + return ret; + break; + } + i++; + } + + i = 0; + for_each_child_of_node(node, np) { + dai_link = rsrc_priv_to_link(priv, i); + dai_link->dai_fmt = daifmt; + + is_fe = false; + if (strcmp(np->name, "cpu") == 0) + is_fe = true; + + ret = rsrc_card_dai_sub_link_of(node, np, priv, i, is_fe); + if (ret < 0) + return ret; + i++; + } + + return 0; +} + static int rsrc_card_parse_of(struct device_node *node, struct rsrc_card_priv *priv, struct device *dev) @@ -366,9 +399,8 @@ static int rsrc_card_parse_of(struct device_node *node, const struct rsrc_card_of_data *of_data = rsrc_dev_to_of_data(dev); struct rsrc_card_dai *props; struct snd_soc_dai_link *links; - struct device_node *np; int ret; - int i, num; + int num; if (!node) return -EINVAL; @@ -409,13 +441,9 @@ static int rsrc_card_parse_of(struct device_node *node, priv->snd_card.name ? priv->snd_card.name : "", priv->convert_rate); - i = 0; - for_each_child_of_node(node, np) { - ret = rsrc_card_dai_link_of(node, np, priv, i); - if (ret < 0) - return ret; - i++; - } + ret = rsrc_card_dai_link_of(node, priv); + if (ret < 0) + return ret; if (!priv->snd_card.name) priv->snd_card.name = priv->snd_card.dai_link->name; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller
by Songjun Wu 18 Dec '15

18 Dec '15

Add driver for the Pulse Density Modulation Interface Controller. It comes with digitallly controlled gain, a High-Pass and a SINCC filter. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- Changes in v2: - Remove the function 'atmel_pdmic_codec_get_remap', since the regmap is initialized in the core. - Replace 'if' with 'switch' in function 'atmel_pdmic_codec_dai_hw_params'. - Fix the coding style issue in function 'atmel_pdmic_dt_init', 'atmel_pdmic_interrupt','pdmic_put_mic_volsw', 'atmel_pdmic_codec_probe', 'pdmic_get_mic_volsw' and 'atmel_pdmic_codec_dai_hw_params'. sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 738 +++++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 +++++ 4 files changed, 829 insertions(+) create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 2d30464..06e099e 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -68,4 +68,13 @@ config SND_ATMEL_SOC_CLASSD help Say Y if you want to add support for Atmel ASoC driver for boards using CLASSD. + +config SND_ATMEL_SOC_PDMIC + tristate "Atmel ASoC driver for boards using PDMIC" + depends on OF && (ARCH_AT91 || COMPILE_TEST) + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y if you want to add support for Atmel ASoC driver for boards using + PDMIC. endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index f6f7db4..a2b127b 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -12,8 +12,10 @@ snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o snd-atmel-soc-wm8904-objs := atmel_wm8904.o snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o snd-atmel-soc-classd-objs := atmel-classd.o +snd-atmel-soc-pdmic-objs := atmel-pdmic.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o +obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c new file mode 100644 index 0000000..aee4787 --- /dev/null +++ b/sound/soc/atmel/atmel-pdmic.c @@ -0,0 +1,738 @@ +/* Atmel PDMIC driver + * + * Copyright (C) 2015 Atmel + * + * Author: Songjun Wu <songjun.wu(a)atmel.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 or later + * as published by the Free Software Foundation. + */ + +#include <linux/of.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "atmel-pdmic.h" + +struct atmel_pdmic_pdata { + u32 mic_min_freq; + u32 mic_max_freq; + s32 mic_offset; + const char *card_name; +}; + +struct atmel_pdmic { + dma_addr_t phy_base; + struct regmap *regmap; + struct clk *pclk; + struct clk *gclk; + int irq; + struct snd_pcm_substream *substream; + const struct atmel_pdmic_pdata *pdata; +}; + +static const struct of_device_id atmel_pdmic_of_match[] = { + { + .compatible = "atmel,sama5d2-pdmic", + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, atmel_pdmic_of_match); + +#define PDMIC_OFFSET_MAX_VAL S16_MAX +#define PDMIC_OFFSET_MIN_VAL S16_MIN + +static struct atmel_pdmic_pdata *atmel_pdmic_dt_init(struct device *dev) +{ + struct device_node *np = dev->of_node; + struct atmel_pdmic_pdata *pdata; + + if (!np) { + dev_err(dev, "device node not found\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + if (of_property_read_string(np, "atmel,model", &pdata->card_name)) + pdata->card_name = "PDMIC"; + + if (of_property_read_u32(np, "atmel,mic-min-freq", + &pdata->mic_min_freq)) { + dev_err(dev, "failed to get mic-min-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (of_property_read_u32(np, "atmel,mic-max-freq", + &pdata->mic_max_freq)) { + dev_err(dev, "failed to get mic-max-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (pdata->mic_max_freq < pdata->mic_min_freq) { + dev_err(dev, + "mic-max-freq should not less than mic-min-freq\n"); + return ERR_PTR(-EINVAL); + } + + if (of_property_read_s32(np, "atmel,mic-offset", &pdata->mic_offset)) + pdata->mic_offset = 0; + + if (pdata->mic_offset > PDMIC_OFFSET_MAX_VAL) { + dev_warn(dev, + "mic-offset value %d is larger than the max value %d, the max value is specified\n", + pdata->mic_offset, PDMIC_OFFSET_MAX_VAL); + pdata->mic_offset = PDMIC_OFFSET_MAX_VAL; + } else if (pdata->mic_offset < PDMIC_OFFSET_MIN_VAL) { + dev_warn(dev, + "mic-offset value %d is less than the min value %d, the min value is specified\n", + pdata->mic_offset, PDMIC_OFFSET_MIN_VAL); + pdata->mic_offset = PDMIC_OFFSET_MIN_VAL; + } + + return pdata; +} + +/* cpu dai component */ +static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = clk_prepare_enable(dd->gclk); + if (ret) + return ret; + + ret = clk_prepare_enable(dd->pclk); + if (ret) + return ret; + + /* Clear all bits in the Control Register(PDMIC_CR) */ + regmap_write(dd->regmap, PDMIC_CR, 0); + + dd->substream = substream; + + /* Enable the overrun error interrupt */ + regmap_write(dd->regmap, PDMIC_IER, PDMIC_IER_OVRE); + + return 0; +} + +static void atmel_pdmic_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + + /* Disable the overrun error interrupt */ + regmap_write(dd->regmap, PDMIC_IDR, PDMIC_IDR_OVRE); + + clk_disable_unprepare(dd->gclk); + clk_disable_unprepare(dd->pclk); +} + +static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + u32 val; + + /* Clean the PDMIC Converted Data Register */ + return regmap_read(dd->regmap, PDMIC_CDR, &val); +} + +static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { + .startup = atmel_pdmic_cpu_dai_startup, + .shutdown = atmel_pdmic_cpu_dai_shutdown, + .prepare = atmel_pdmic_cpu_dai_prepare, +}; + +#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = ATMEL_PDMIC_FORMATS,}, + .ops = &atmel_pdmic_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { + .name = "atmel-pdmic", +}; + +/* platform */ +#define ATMEL_PDMIC_MAX_BUF_SIZE (64 * 1024) +#define ATMEL_PDMIC_PREALLOC_BUF_SIZE ATMEL_PDMIC_MAX_BUF_SIZE + +static const struct snd_pcm_hardware atmel_pdmic_hw = { + .info = SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE, + .formats = ATMEL_PDMIC_FORMATS, + .buffer_bytes_max = ATMEL_PDMIC_MAX_BUF_SIZE, + .period_bytes_min = 256, + .period_bytes_max = 32 * 1024, + .periods_min = 2, + .periods_max = 256, +}; + +static int +atmel_pdmic_platform_configure_dma(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + int ret; + + ret = snd_hwparams_to_dma_slave_config(substream, params, + slave_config); + if (ret) { + dev_err(rtd->platform->dev, + "hw params to dma slave configure failed\n"); + return ret; + } + + slave_config->src_addr = dd->phy_base + PDMIC_CDR; + slave_config->src_maxburst = 1; + slave_config->dst_maxburst = 1; + + return 0; +} + +static const struct snd_dmaengine_pcm_config +atmel_pdmic_dmaengine_pcm_config = { + .prepare_slave_config = atmel_pdmic_platform_configure_dma, + .pcm_hardware = &atmel_pdmic_hw, + .prealloc_buffer_size = ATMEL_PDMIC_PREALLOC_BUF_SIZE, +}; + +/* codec */ +/* Mic Gain = dgain * 2^(-scale) */ +struct mic_gain { + unsigned int dgain; + unsigned int scale; +}; + +/* range from -90 dB to 90 dB */ +static const struct mic_gain mic_gain_table[] = { +{ 1, 15}, { 1, 14}, /* -90, -84 dB */ +{ 3, 15}, { 1, 13}, { 3, 14}, { 1, 12}, /* -81, -78, -75, -72 dB */ +{ 5, 14}, { 13, 15}, /* -70, -68 dB */ +{ 9, 14}, { 21, 15}, { 23, 15}, { 13, 14}, /* -65 ~ -62 dB */ +{ 29, 15}, { 33, 15}, { 37, 15}, { 41, 15}, /* -61 ~ -58 dB */ +{ 23, 14}, { 13, 13}, { 58, 15}, { 65, 15}, /* -57 ~ -54 dB */ +{ 73, 15}, { 41, 14}, { 23, 13}, { 13, 12}, /* -53 ~ -50 dB */ +{ 29, 13}, { 65, 14}, { 73, 14}, { 41, 13}, /* -49 ~ -46 dB */ +{ 23, 12}, { 207, 15}, { 29, 12}, { 65, 13}, /* -45 ~ -42 dB */ +{ 73, 13}, { 41, 12}, { 23, 11}, { 413, 15}, /* -41 ~ -38 dB */ +{ 463, 15}, { 519, 15}, { 583, 15}, { 327, 14}, /* -37 ~ -34 dB */ +{ 367, 14}, { 823, 15}, { 231, 13}, { 1036, 15}, /* -33 ~ -30 dB */ +{ 1163, 15}, { 1305, 15}, { 183, 12}, { 1642, 15}, /* -29 ~ -26 dB */ +{ 1843, 15}, { 2068, 15}, { 145, 11}, { 2603, 15}, /* -25 ~ -22 dB */ +{ 365, 12}, { 3277, 15}, { 3677, 15}, { 4125, 15}, /* -21 ~ -18 dB */ +{ 4629, 15}, { 5193, 15}, { 5827, 15}, { 3269, 14}, /* -17 ~ -14 dB */ +{ 917, 12}, { 8231, 15}, { 9235, 15}, { 5181, 14}, /* -13 ~ -10 dB */ +{11627, 15}, {13045, 15}, {14637, 15}, {16423, 15}, /* -9 ~ -6 dB */ +{18427, 15}, {20675, 15}, { 5799, 13}, {26029, 15}, /* -5 ~ -2 dB */ +{ 7301, 13}, { 1, 0}, {18383, 14}, {10313, 13}, /* -1 ~ 2 dB */ +{23143, 14}, {25967, 14}, {29135, 14}, {16345, 13}, /* 3 ~ 6 dB */ +{ 4585, 11}, {20577, 13}, { 1443, 9}, {25905, 13}, /* 7 ~ 10 dB */ +{14533, 12}, { 8153, 11}, { 2287, 9}, {20529, 12}, /* 11 ~ 14 dB */ +{11517, 11}, { 6461, 10}, {28997, 12}, { 4067, 9}, /* 15 ~ 18 dB */ +{18253, 11}, { 10, 0}, {22979, 11}, {25783, 11}, /* 19 ~ 22 dB */ +{28929, 11}, {32459, 11}, { 9105, 9}, {20431, 10}, /* 23 ~ 26 dB */ +{22925, 10}, {12861, 9}, { 7215, 8}, {16191, 9}, /* 27 ~ 30 dB */ +{ 9083, 8}, {20383, 9}, {11435, 8}, { 6145, 7}, /* 31 ~ 34 dB */ +{ 3599, 6}, {32305, 9}, {18123, 8}, {20335, 8}, /* 35 ~ 38 dB */ +{ 713, 3}, { 100, 0}, { 7181, 6}, { 8057, 6}, /* 39 ~ 42 dB */ +{ 565, 2}, {20287, 7}, {11381, 6}, {25539, 7}, /* 43 ~ 46 dB */ +{ 1791, 3}, { 4019, 4}, { 9019, 5}, {20239, 6}, /* 47 ~ 50 dB */ +{ 5677, 4}, {25479, 6}, { 7147, 4}, { 8019, 4}, /* 51 ~ 54 dB */ +{17995, 5}, {20191, 5}, {11327, 4}, {12709, 4}, /* 55 ~ 58 dB */ +{ 3565, 2}, { 1000, 0}, { 1122, 0}, { 1259, 0}, /* 59 ~ 62 dB */ +{ 2825, 1}, {12679, 3}, { 7113, 2}, { 7981, 2}, /* 63 ~ 66 dB */ +{ 8955, 2}, {20095, 3}, {22547, 3}, {12649, 2}, /* 67 ~ 70 dB */ +{28385, 3}, { 3981, 0}, {17867, 2}, {20047, 2}, /* 71 ~ 74 dB */ +{11247, 1}, {12619, 1}, {14159, 1}, {31773, 2}, /* 75 ~ 78 dB */ +{17825, 1}, {10000, 0}, {11220, 0}, {12589, 0}, /* 79 ~ 82 dB */ +{28251, 1}, {15849, 0}, {17783, 0}, {19953, 0}, /* 83 ~ 86 dB */ +{22387, 0}, {25119, 0}, {28184, 0}, {31623, 0}, /* 87 ~ 90 dB */ +}; + +static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, + 0, 1, TLV_DB_SCALE_ITEM(-9000, 600, 0), + 2, 5, TLV_DB_SCALE_ITEM(-8100, 300, 0), + 6, 7, TLV_DB_SCALE_ITEM(-7000, 200, 0), + 8, ARRAY_SIZE(mic_gain_table)-1, TLV_DB_SCALE_ITEM(-6500, 100, 0), +); + +int pdmic_get_mic_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int dgain_val, scale_val; + int i; + + dgain_val = (snd_soc_read(codec, PDMIC_DSPR1) & PDMIC_DSPR1_DGAIN_MASK) + >> PDMIC_DSPR1_DGAIN_SHIFT; + + scale_val = (snd_soc_read(codec, PDMIC_DSPR0) & PDMIC_DSPR0_SCALE_MASK) + >> PDMIC_DSPR0_SCALE_SHIFT; + + for (i = 0; i < ARRAY_SIZE(mic_gain_table); i++) { + if ((mic_gain_table[i].dgain == dgain_val) && + (mic_gain_table[i].scale == scale_val)) + ucontrol->value.integer.value[0] = i; + } + + return 0; +} + +static int pdmic_put_mic_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + int max = mc->max; + unsigned int val; + int ret; + + val = ucontrol->value.integer.value[0]; + + if (val > max) + return -EINVAL; + + ret = snd_soc_update_bits(codec, PDMIC_DSPR1, PDMIC_DSPR1_DGAIN_MASK, + mic_gain_table[val].dgain << PDMIC_DSPR1_DGAIN_SHIFT); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, PDMIC_DSPR0, PDMIC_DSPR0_SCALE_MASK, + mic_gain_table[val].scale << PDMIC_DSPR0_SCALE_SHIFT); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_kcontrol_new atmel_pdmic_snd_controls[] = { +SOC_SINGLE_EXT_TLV("Mic Capture Volume", PDMIC_DSPR1, PDMIC_DSPR1_DGAIN_SHIFT, + ARRAY_SIZE(mic_gain_table)-1, 0, + pdmic_get_mic_volsw, pdmic_put_mic_volsw, mic_gain_tlv), + +SOC_SINGLE("High Pass Filter Switch", PDMIC_DSPR0, + PDMIC_DSPR0_HPFBYP_SHIFT, 1, 1), + +SOC_SINGLE("SINCC Filter Switch", PDMIC_DSPR0, PDMIC_DSPR0_SINBYP_SHIFT, 1, 1), +}; + +static int atmel_pdmic_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec); + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card); + + snd_soc_update_bits(codec, PDMIC_DSPR1, PDMIC_DSPR1_OFFSET_MASK, + (u32)(dd->pdata->mic_offset << PDMIC_DSPR1_OFFSET_SHIFT)); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pdmic = { + .probe = atmel_pdmic_codec_probe, + .controls = atmel_pdmic_snd_controls, + .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), +}; + +/* codec dai component */ +#define PDMIC_MR_PRESCAL_MAX_VAL 127 + +static int +atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int rate_min = substream->runtime->hw.rate_min; + unsigned int rate_max = substream->runtime->hw.rate_max; + int fs = params_rate(params); + int bits = params_width(params); + unsigned long pclk_rate, gclk_rate; + unsigned int f_pdmic; + u32 mr_val, dspr0_val, pclk_prescal, gclk_prescal; + + if (params_channels(params) != 1) { + dev_err(codec->dev, + "only supports one channel\n"); + return -EINVAL; + } + + if ((fs < rate_min) || (fs > rate_max)) { + dev_err(codec->dev, + "sample rate is %dHz, min rate is %dHz, max rate is %dHz\n", + fs, rate_min, rate_max); + + return -EINVAL; + } + + switch (bits) { + case 16: + dspr0_val = (PDMIC_DSPR0_SIZE_16_BITS + << PDMIC_DSPR0_SIZE_SHIFT); + break; + case 32: + dspr0_val = (PDMIC_DSPR0_SIZE_32_BITS + << PDMIC_DSPR0_SIZE_SHIFT); + break; + default: + return -EINVAL; + } + + if ((fs << 7) > (rate_max << 6)) { + f_pdmic = fs << 6; + dspr0_val |= PDMIC_DSPR0_OSR_64 << PDMIC_DSPR0_OSR_SHIFT; + } else { + f_pdmic = fs << 7; + dspr0_val |= PDMIC_DSPR0_OSR_128 << PDMIC_DSPR0_OSR_SHIFT; + } + + pclk_rate = clk_get_rate(dd->pclk); + gclk_rate = clk_get_rate(dd->gclk); + + /* PRESCAL = SELCK/(2*f_pdmic) - 1*/ + pclk_prescal = (u32)(pclk_rate/(f_pdmic << 1)) - 1; + gclk_prescal = (u32)(gclk_rate/(f_pdmic << 1)) - 1; + + if ((pclk_prescal > PDMIC_MR_PRESCAL_MAX_VAL) || + (gclk_rate/((gclk_prescal + 1) << 1) < + pclk_rate/((pclk_prescal + 1) << 1))) { + mr_val = gclk_prescal << PDMIC_MR_PRESCAL_SHIFT; + mr_val |= PDMIC_MR_CLKS_GCK << PDMIC_MR_CLKS_SHIFT; + } else { + mr_val = pclk_prescal << PDMIC_MR_PRESCAL_SHIFT; + mr_val |= PDMIC_MR_CLKS_PCK << PDMIC_MR_CLKS_SHIFT; + } + + snd_soc_update_bits(codec, PDMIC_MR, + PDMIC_MR_PRESCAL_MASK | PDMIC_MR_CLKS_MASK, mr_val); + + snd_soc_update_bits(codec, PDMIC_DSPR0, + PDMIC_DSPR0_OSR_MASK | PDMIC_DSPR0_SIZE_MASK, dspr0_val); + + return 0; +} + +static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + snd_soc_update_bits(codec, PDMIC_CR, PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); + + return 0; +} + +static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u32 val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = PDMIC_CR_ENPDM_EN << PDMIC_CR_ENPDM_SHIFT; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, PDMIC_CR, PDMIC_CR_ENPDM_MASK, val); + + return 0; +} + +static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = { + .hw_params = atmel_pdmic_codec_dai_hw_params, + .prepare = atmel_pdmic_codec_dai_prepare, + .trigger = atmel_pdmic_codec_dai_trigger, +}; + +#define ATMEL_PDMIC_CODEC_DAI_NAME "atmel-pdmic-hifi" + +static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { + .name = ATMEL_PDMIC_CODEC_DAI_NAME, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = ATMEL_PDMIC_FORMATS, + }, + .ops = &atmel_pdmic_codec_dai_ops, +}; + +/* ASoC sound card */ +static int atmel_pdmic_asoc_card_init(struct device *dev, + struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card); + + dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL); + if (!dai_link) + return -ENOMEM; + + dai_link->name = "PDMIC"; + dai_link->stream_name = "PDMIC PCM"; + dai_link->codec_dai_name = ATMEL_PDMIC_CODEC_DAI_NAME; + dai_link->cpu_dai_name = dev_name(dev); + dai_link->codec_name = dev_name(dev); + dai_link->platform_name = dev_name(dev); + + card->dai_link = dai_link; + card->num_links = 1; + card->name = dd->pdata->card_name; + card->dev = dev; + + return 0; +} + +static void atmel_pdmic_get_sample_rate(struct atmel_pdmic *dd, + unsigned int *rate_min, unsigned int *rate_max) +{ + u32 mic_min_freq = dd->pdata->mic_min_freq; + u32 mic_max_freq = dd->pdata->mic_max_freq; + u32 clk_max_rate = (u32)(clk_get_rate(dd->pclk) >> 1); + u32 clk_min_rate = (u32)(clk_get_rate(dd->gclk) >> 8); + + if (mic_max_freq > clk_max_rate) + mic_max_freq = clk_max_rate; + + if (mic_min_freq < clk_min_rate) + mic_min_freq = clk_min_rate; + + *rate_min = DIV_ROUND_CLOSEST(mic_min_freq, 128); + *rate_max = mic_max_freq >> 6; +} + +/* PDMIC interrupt handler */ +static irqreturn_t atmel_pdmic_interrupt(int irq, void *dev_id) +{ + struct atmel_pdmic *dd = (struct atmel_pdmic *)dev_id; + u32 pdmic_isr; + irqreturn_t ret = IRQ_NONE; + + regmap_read(dd->regmap, PDMIC_ISR, &pdmic_isr); + + if (pdmic_isr & PDMIC_ISR_OVRE) { + regmap_update_bits(dd->regmap, PDMIC_CR, PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); + + snd_pcm_stop_xrun(dd->substream); + + ret = IRQ_HANDLED; + } + + return ret; +} + +/* regmap configuration */ +#define ATMEL_PDMIC_REG_MAX 0x124 +static const struct regmap_config atmel_pdmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = ATMEL_PDMIC_REG_MAX, +}; + +static int atmel_pdmic_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct atmel_pdmic *dd; + struct resource *res; + void __iomem *io_base; + const struct atmel_pdmic_pdata *pdata; + struct snd_soc_card *card; + unsigned int rate_min, rate_max; + int ret; + + pdata = atmel_pdmic_dt_init(dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); + + dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL); + if (!dd) + return -ENOMEM; + + dd->pdata = pdata; + + dd->irq = platform_get_irq(pdev, 0); + if (dd->irq < 0) { + ret = dd->irq; + dev_err(dev, "failed to could not get irq: %d\n", ret); + return ret; + } + + dd->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(dd->pclk)) { + ret = PTR_ERR(dd->pclk); + dev_err(dev, "failed to get peripheral clock: %d\n", ret); + return ret; + } + + dd->gclk = devm_clk_get(dev, "gclk"); + if (IS_ERR(dd->gclk)) { + ret = PTR_ERR(dd->gclk); + dev_err(dev, "failed to get GCK: %d\n", ret); + return ret; + } + + /* The gclk clock frequency must always be tree times + * lower than the pclk clock frequency + */ + ret = clk_set_rate(dd->gclk, clk_get_rate(dd->pclk)/3); + if (ret) { + dev_err(dev, "failed to set GCK clock rate: %d\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "no memory resource\n"); + return -ENXIO; + } + + io_base = devm_ioremap_resource(dev, res); + if (IS_ERR(io_base)) { + ret = PTR_ERR(io_base); + dev_err(dev, "failed to remap register memory: %d\n", ret); + return ret; + } + + dd->phy_base = res->start; + + dd->regmap = devm_regmap_init_mmio(dev, io_base, + &atmel_pdmic_regmap_config); + if (IS_ERR(dd->regmap)) { + ret = PTR_ERR(dd->regmap); + dev_err(dev, "failed to init register map: %d\n", ret); + return ret; + } + + ret = devm_request_irq(dev, dd->irq, atmel_pdmic_interrupt, 0, + "PDMIC", (void *)dd); + if (ret < 0) { + dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", + dd->irq, ret); + return ret; + } + + /* Get the minimal and maximal sample rate that micphone supports */ + atmel_pdmic_get_sample_rate(dd, &rate_min, &rate_max); + + /* register cpu dai */ + atmel_pdmic_cpu_dai.capture.rate_min = rate_min; + atmel_pdmic_cpu_dai.capture.rate_max = rate_max; + ret = devm_snd_soc_register_component(dev, + &atmel_pdmic_cpu_dai_component, + &atmel_pdmic_cpu_dai, 1); + if (ret) { + dev_err(dev, "could not register CPU DAI: %d\n", ret); + return ret; + } + + /* register platform */ + ret = devm_snd_dmaengine_pcm_register(dev, + &atmel_pdmic_dmaengine_pcm_config, + 0); + if (ret) { + dev_err(dev, "could not register platform: %d\n", ret); + return ret; + } + + /* register codec and codec dai */ + atmel_pdmic_codec_dai.capture.rate_min = rate_min; + atmel_pdmic_codec_dai.capture.rate_max = rate_max; + ret = snd_soc_register_codec(dev, &soc_codec_dev_pdmic, + &atmel_pdmic_codec_dai, 1); + if (ret) { + dev_err(dev, "could not register codec: %d\n", ret); + return ret; + } + + /* register sound card */ + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) { + ret = -ENOMEM; + goto unregister_codec; + } + + snd_soc_card_set_drvdata(card, dd); + platform_set_drvdata(pdev, card); + + ret = atmel_pdmic_asoc_card_init(dev, card); + if (ret) { + dev_err(dev, "failed to init sound card: %d\n", ret); + goto unregister_codec; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret) { + dev_err(dev, "failed to register sound card: %d\n", ret); + goto unregister_codec; + } + + return 0; + +unregister_codec: + snd_soc_unregister_codec(dev); + return ret; +} + +static int atmel_pdmic_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver atmel_pdmic_driver = { + .driver = { + .name = "atmel-pdmic", + .of_match_table = of_match_ptr(atmel_pdmic_of_match), + .pm = &snd_soc_pm_ops, + }, + .probe = atmel_pdmic_probe, + .remove = atmel_pdmic_remove, +}; +module_platform_driver(atmel_pdmic_driver); + +MODULE_DESCRIPTION("Atmel PDMIC driver under ALSA SoC architecture"); +MODULE_AUTHOR("Songjun Wu <songjun.wu(a)atmel.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/atmel/atmel-pdmic.h b/sound/soc/atmel/atmel-pdmic.h new file mode 100644 index 0000000..4527ac7 --- /dev/null +++ b/sound/soc/atmel/atmel-pdmic.h @@ -0,0 +1,80 @@ +#ifndef __ATMEL_PDMIC_H_ +#define __ATMEL_PDMIC_H_ + +#include <linux/bitops.h> + +#define PDMIC_CR 0x00000000 + +#define PDMIC_CR_SWRST 0x1 +#define PDMIC_CR_SWRST_MASK BIT(0) +#define PDMIC_CR_SWRST_SHIFT (0) + +#define PDMIC_CR_ENPDM_DIS 0x0 +#define PDMIC_CR_ENPDM_EN 0x1 +#define PDMIC_CR_ENPDM_MASK BIT(4) +#define PDMIC_CR_ENPDM_SHIFT (4) + +#define PDMIC_MR 0x00000004 + +#define PDMIC_MR_CLKS_PCK 0x0 +#define PDMIC_MR_CLKS_GCK 0x1 +#define PDMIC_MR_CLKS_MASK BIT(4) +#define PDMIC_MR_CLKS_SHIFT (4) + +#define PDMIC_MR_PRESCAL_MASK GENMASK(14, 8) +#define PDMIC_MR_PRESCAL_SHIFT (8) + +#define PDMIC_CDR 0x00000014 + +#define PDMIC_IER 0x00000018 +#define PDMIC_IER_OVRE BIT(25) + +#define PDMIC_IDR 0x0000001c +#define PDMIC_IDR_OVRE BIT(25) + +#define PDMIC_IMR 0x00000020 + +#define PDMIC_ISR 0x00000024 +#define PDMIC_ISR_OVRE BIT(25) + +#define PDMIC_DSPR0 0x00000058 + +#define PDMIC_DSPR0_HPFBYP_DIS 0x1 +#define PDMIC_DSPR0_HPFBYP_EN 0x0 +#define PDMIC_DSPR0_HPFBYP_MASK BIT(1) +#define PDMIC_DSPR0_HPFBYP_SHIFT (1) + +#define PDMIC_DSPR0_SINBYP_DIS 0x1 +#define PDMIC_DSPR0_SINBYP_EN 0x0 +#define PDMIC_DSPR0_SINBYP_MASK BIT(2) +#define PDMIC_DSPR0_SINBYP_SHIFT (2) + +#define PDMIC_DSPR0_SIZE_16_BITS 0x0 +#define PDMIC_DSPR0_SIZE_32_BITS 0x1 +#define PDMIC_DSPR0_SIZE_MASK BIT(3) +#define PDMIC_DSPR0_SIZE_SHIFT (3) + +#define PDMIC_DSPR0_OSR_128 0x0 +#define PDMIC_DSPR0_OSR_64 0x1 +#define PDMIC_DSPR0_OSR_MASK GENMASK(6, 4) +#define PDMIC_DSPR0_OSR_SHIFT (4) + +#define PDMIC_DSPR0_SCALE_MASK GENMASK(11, 8) +#define PDMIC_DSPR0_SCALE_SHIFT (8) + +#define PDMIC_DSPR0_SHIFT_MASK GENMASK(15, 12) +#define PDMIC_DSPR0_SHIFT_SHIFT (12) + +#define PDMIC_DSPR1 0x0000005c + +#define PDMIC_DSPR1_DGAIN_MASK GENMASK(14, 0) +#define PDMIC_DSPR1_DGAIN_SHIFT (0) + +#define PDMIC_DSPR1_OFFSET_MASK GENMASK(31, 16) +#define PDMIC_DSPR1_OFFSET_SHIFT (16) + +#define PDMIC_WPMR 0x000000e4 + +#define PDMIC_WPSR 0x000000e8 + +#endif -- 1.7.9.5

2 1

[alsa-devel] Applied "ASoC: atmel-classd: DT binding for PDMIC driver" to the asoc tree
by Mark Brown 18 Dec '15

18 Dec '15

The patch ASoC: atmel-classd: DT binding for PDMIC driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4bc5145cec89b3df719d87d0a6876aa00941d66c Mon Sep 17 00:00:00 2001 From: Songjun Wu <songjun.wu(a)atmel.com> Date: Thu, 17 Dec 2015 17:50:00 +0800 Subject: [PATCH] ASoC: atmel-classd: DT binding for PDMIC driver DT binding documentation for this new ASoC driver. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ++++++++++++++++++++++ 1 file changed, 55 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt diff --git a/Documentation/devicetree/bindings/sound/atmel-pdmic.txt b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt new file mode 100644 index 000000000000..e0875f17c229 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-pdmic.txt @@ -0,0 +1,55 @@ +* Atmel PDMIC driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-pdmic". +- reg + Should contain PDMIC registers location and length. +- interrupts + Should contain the IRQ line for the PDMIC. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "rx". +- clock-names + Required elements: + - "pclk" peripheral clock + - "gclk" generated clock +- clocks + Must contain an entry for each required entry in clock-names. + Please refer to clock-bindings.txt. +- atmel,mic-min-freq + The minimal frequency that the micphone supports. +- atmel,mic-max-freq + The maximal frequency that the micphone supports. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound card. + The default value is "PDMIC". +- atmel,mic-offset + The offset that should be added. + The range is from -32768 to 32767. + The default value is 0. + +Example: + pdmic@f8018000 { + compatible = "atmel,sama5d2-pdmic"; + reg = <0xf8018000 0x124>; + interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(50))>; + dma-names = "rx"; + clocks = <&pdmic_clk>, <&pdmic_gclk>; + clock-names = "pclk", "gclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pdmic_default>; + atmel,model = "PDMIC @ sama5d2_xplained"; + atmel,mic-min-freq = <1000000>; + atmel,mic-max-freq = <3246000>; + atmel,mic-offset = <0x0>; + }; -- 2.6.2

1 0

[alsa-devel] [PATCH 2/2] ASoC: wm8974: add devicetree support
by Mans Rullgard 18 Dec '15

18 Dec '15

This adds devicetree support to the wm8974 codec driver. With a DT-based kernel, there is no board-specific setting to select the driver so allow it to be manually chosen. Signed-off-by: Mans Rullgard <mans(a)mansr.com> --- sound/soc/codecs/Kconfig | 3 ++- sound/soc/codecs/wm8974.c | 7 +++++++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cfdafc4..e36b14c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -838,7 +838,8 @@ config SND_SOC_WM8971 tristate config SND_SOC_WM8974 - tristate + tristate "Wolfson Microelectronics WM8974 codec" + depends on I2C config SND_SOC_WM8978 tristate "Wolfson Microelectronics WM8978 codec" diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 4c29bd2..c284c7b 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -632,9 +632,16 @@ static const struct i2c_device_id wm8974_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); +static const struct of_device_id wm8974_of_match[] = { + { .compatible = "wlf,wm8974", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8974_of_match); + static struct i2c_driver wm8974_i2c_driver = { .driver = { .name = "wm8974", + .of_match_table = wm8974_of_match, }, .probe = wm8974_i2c_probe, .remove = wm8974_i2c_remove, -- 2.6.3

3 3

[alsa-devel] Applied "ASoC: wm8974: add binding for WM8974 codec" to the asoc tree
by Mark Brown 18 Dec '15

18 Dec '15

The patch ASoC: wm8974: add binding for WM8974 codec 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 47d358bbf22d504a41007e7fbf71b5b4b1b45b79 Mon Sep 17 00:00:00 2001 From: Mans Rullgard <mans(a)mansr.com> Date: Wed, 16 Dec 2015 13:55:13 +0000 Subject: [PATCH] ASoC: wm8974: add binding for WM8974 codec This adds a binding for the Wolfson WM8974 mono audio codec. Signed-off-by: Mans Rullgard <mans(a)mansr.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/wlf,wm8974.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt new file mode 100644 index 000000000000..01d3a7c83419 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt @@ -0,0 +1,15 @@ +WM8974 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "wlf,wm8974" + - reg: the I2C address or SPI chip select number of the device + +Examples: + +codec: wm8974@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; +}; -- 2.6.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Set intel hda controller power at freeze() and thaw()
by Xiong Zhang 18 Dec '15

18 Dec '15

It takes three minutes to enter into hibernation on some OEM SKL machines and we see many codec spurious response after thaw() opertion. This is because HDA is still in D0 state after freeze() call and pci_pm_freeze/pci_pm_freeze_noirq() don't set D3 hot in pci_bus driver. It seems bios still access HDA when system enter into freeze state, HDA will receive codec response interrupt immediately after thaw() call. Because of this unexpected interrupt, HDA enter into a abnormal state and slow down the system enter into hibernation. In this patch, we put HDA into D3 hot state in azx_freeze_noirq() and put HDA into D0 state in azx_thaw_noirq(). Signed-off-by: Xiong Zhang <xiong.y.zhang(a)intel.com> --- sound/pci/hda/hda_intel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c38c68f..3ba7cbc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -890,6 +890,16 @@ static int azx_suspend(struct device *dev) return 0; } +static int azx_freeze_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (pci->vendor == PCI_VENDOR_ID_INTEL) + pci_set_power_state(pci, PCI_D3hot); + + return 0; +} + static int azx_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -924,6 +934,14 @@ static int azx_resume(struct device *dev) trace_azx_resume(chip); return 0; } + +static int azx_thaw_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (pci->vendor == PCI_VENDOR_ID_INTEL) + pci_set_power_state(pci, PCI_D0); +} #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ #ifdef CONFIG_PM @@ -1035,6 +1053,8 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + .freeze_noirq = azx_freeze_noirq, + .thaw_noirq = azx_thaw_noirq, SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; -- 1.8.2.1

4 7

[alsa-devel] HT Omega eCLARO sound card
by Patricio Rossi 18 Dec '15

18 Dec '15

I got this card few days ago, after connected it into my PC got detected by lspci, but I cannot get a module to load, looks like the oxigen is the correct one but even when I load it manually nothing happens. I did some search on the web and I found people that was able to use it but I can't find how, I read the oxigen source code and the card is not there, is very similar to the ASUS Xonar. Is this the correct place to look for help? What is the status of this driver? where I can get more information? Thank you!

2 6

[alsa-devel] mixer controls not detected for class compliant USB audio interface (RME Babyface Pro)
by Be 17 Dec '15

17 Dec '15

Hello, I recently got a USB audio class compliant RME Babyface Pro interface. I previously posted about an issue getting all of the channels to work, but that ended up being an issue with PortAudio that is trivial to work around. Anyway, there is still the issue that the device's mixer controls are not detected by Linux. alsamixer sees no controls at all. The mixer controls must be accessible in the device's class compliant mode because the manufacturer's iPad mixer program can control its mixer and onboard DSP effects. I have attached lsusb -vvv output for the device. Below is dmesg output when the device is plugged in: [191013.268711] usb 2-1.2: new high-speed USB device number 11 using ehci-pci [191013.355392] usb 2-1.2: New USB device found, idVendor=2a39, idProduct=3fb0 [191013.355403] usb 2-1.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [191013.355409] usb 2-1.2: Product: Babyface Pro (70785713) [191013.355413] usb 2-1.2: Manufacturer: RME [191013.355417] usb 2-1.2: SerialNumber: 9399A998B9CB3C8 [191013.360347] usb 2-1.2: unit 2 not found!

2 3

[alsa-devel] [PATCH] ASoC: rt5677: Reconfigure PLL1 after resume
by Ben Zhang 17 Dec '15

17 Dec '15

Sometimes PLL1 stops working if the codec loses power during suspend (when pow-ldo2 or reset gpio is used). MX-7Bh(RT5677_PLL1_CTRL2) is cleared and won't be restored by regcache since it's volatile. MX-7Bh has one status bit and M code for PLL1. rt5677_set_dai_pll doesn't reconfigure PLL1 after resume because it thinks the PLL params are not changed. This patch clears the cached PLL params at resume so that rt5677_set_dai_pll can reconfigure the PLL after resume. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f73fd12..c404f51 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4792,6 +4792,9 @@ static int rt5677_resume(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); if (!rt5677->dsp_vad_en) { + rt5677->pll_src = 0; + rt5677->pll_in = 0; + rt5677->pll_out = 0; gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); gpiod_set_value_cansleep(rt5677->reset_pin, 0); if (rt5677->pow_ldo2 || rt5677->reset_pin) -- 2.6.0.rc2.230.g3dd15c0

3 4

[alsa-devel] [sound:for-next 38/40] sound/pci/hda/hda_tegra.c:484:36: error: 'AZX_DCAPS_RIRB_DELAY' undeclared
by kbuild test robot 17 Dec '15

17 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 64cfd2fd5325dbe24b990dbe003c22f5675d9742 commit: 119a42ac2b249abd6b5195e0606a0d70af2953eb [38/40] ALSA: hda - Move Nvidia-specific check into hda_intel.c config: arm-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 119a42ac2b249abd6b5195e0606a0d70af2953eb # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/pci/hda/hda_tegra.c: In function 'hda_tegra_probe': >> sound/pci/hda/hda_tegra.c:484:36: error: 'AZX_DCAPS_RIRB_DELAY' undeclared (first use in this function) const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | ^ sound/pci/hda/hda_tegra.c:484:36: note: each undeclared identifier is reported only once for each function it appears in vim +/AZX_DCAPS_RIRB_DELAY +484 sound/pci/hda/hda_tegra.c 3c320f3f Dylan Reid 2014-05-19 478 {}, 3c320f3f Dylan Reid 2014-05-19 479 }; f73387cb Dylan Reid 2014-05-20 480 MODULE_DEVICE_TABLE(of, hda_tegra_match); 3c320f3f Dylan Reid 2014-05-19 481 3c320f3f Dylan Reid 2014-05-19 482 static int hda_tegra_probe(struct platform_device *pdev) 3c320f3f Dylan Reid 2014-05-19 483 { 88871dd2 Thierry Reding 2015-05-05 @484 const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | 88871dd2 Thierry Reding 2015-05-05 485 AZX_DCAPS_CORBRP_SELF_CLEAR; 3c320f3f Dylan Reid 2014-05-19 486 struct snd_card *card; 3c320f3f Dylan Reid 2014-05-19 487 struct azx *chip; :::::: The code at line 484 was first introduced by commit :::::: 88871dd25f64b9bf16222e829ad766fac201866a ALSA: hda/tegra - Set CORBRP self-clear flag :::::: TO: Thierry Reding <treding(a)nvidia.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [RESEND PATCH v2 0/2] ASoC: atmel-pdmic: add driver for Atmel PDMIC
by Songjun Wu 17 Dec '15

17 Dec '15

The Pulse Density Modulation Inteface Controller driver includes two parts. 1) Driver code to implement the PDMIC function. 2) Device tree binding documentation, it describes how to add the PDMIC in device tree. Changes in v2: - Remove the function 'atmel_pdmic_codec_get_remap', since the regmap is initialized in the core. - Replace 'if' with 'switch' in function 'atmel_pdmic_codec_dai_hw_params'. - Fix the coding style issue in function 'atmel_pdmic_dt_init', 'atmel_pdmic_interrupt','pdmic_put_mic_volsw', 'atmel_pdmic_codec_probe', 'pdmic_get_mic_volsw' and 'atmel_pdmic_codec_dai_hw_params'. Songjun Wu (2): ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller ASoC: atmel-classd: DT binding for PDMIC driver .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 738 ++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 +++ 5 files changed, 884 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h -- 1.7.9.5

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[alsa-devel] [PATCH 0/4] Be ecologist; save more bits in AZX_DCAPS_*
by Takashi Iwai 17 Dec '15

17 Dec '15

Hi, this patchset cleans up messy AZX_DCAPS_* definitions and remove a few usages there. Takashi === Takashi Iwai (4): ALSA: hda - Merge RIRB_PRE_DELAY into CTX_WORKAROUND caps ALSA: hda - Move Nvidia-specific check into hda_intel.c ALSA: hda - Drop AZX_DCAPS_POSFIX_VIA bit ALSA: hda - Drop unused AZX_DCAPS_REVERSE_ASSIGN sound/pci/hda/hda_controller.c | 7 +------ sound/pci/hda/hda_controller.h | 8 ++++---- sound/pci/hda/hda_intel.c | 20 +++++++++++--------- 3 files changed, 16 insertions(+), 19 deletions(-) -- 2.6.4

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[alsa-devel] [PATCH 0/2] ASoC: atmel-pdmic: add driver for Atmel PDMIC
by Songjun Wu 17 Dec '15

17 Dec '15

The Pulse Density Modulation Inteface Controller driver includes two parts. 1) Driver code to implement the PDMIC function. 2) Device tree binding documentation, it describes how to add the PDMIC in device tree. Songjun Wu (2): ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller ASoC: atmel-pdmic: DT binding for PDMIC driver .../devicetree/bindings/sound/atmel-pdmic.txt | 55 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pdmic.c | 745 ++++++++++++++++++++ sound/soc/atmel/atmel-pdmic.h | 80 +++ 5 files changed, 891 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h -- 1.7.9.5

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[alsa-devel] [patch] ASoC: Intel: Skylake: Fix a couple signedness bugs
by Dan Carpenter 16 Dec '15

16 Dec '15

These need to be unsigned because they hold negative error codes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 8cd5cdb..e26f474 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -208,7 +208,7 @@ static unsigned int skl_get_errorcode(struct sst_dsp *ctx) * since get/set_module are called from DAPM context, * we don't need lock for usage count */ -static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_get_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -220,7 +220,7 @@ static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) return -EINVAL; } -static unsigned int skl_put_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_put_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -340,7 +340,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id) { - unsigned int usage_cnt; + int usage_cnt; struct skl_sst *skl = ctx->thread_context; int ret = 0;

3 4

[alsa-devel] [PATCH] ASoC: arizona: In arizona_calc_fratio make new codecs the default case
by Richard Fitzgerald 16 Dec '15

16 Dec '15

This patch rearranges the switch statement in arizona_calc_fratio so that older codecs are the special cases, with the default case applying to newer codecs (WM8998 and later). This is preferable because it avoids having to patch new cases in every time a new codec is added. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d90b3c5..38a73e3 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2020,18 +2020,18 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } switch (fll->arizona->type) { + case WM5102: + case WM8997: + return init_ratio; case WM5110: case WM8280: if (fll->arizona->rev < 3 || sync) return init_ratio; break; - case WM8998: - case WM1814: + default: if (sync) return init_ratio; break; - default: - return init_ratio; } cfg->fratio = init_ratio - 1; -- 1.9.1

2 1

[alsa-devel] [patch] ASoC: Intel: Skylake: pointer math issue
by Dan Carpenter 16 Dec '15

16 Dec '15

"data" is a u32 pointer so this copies the information to wrong place entirely. Fixes: 140adfba5280 ('ASoC: Intel: Skylake: Add tlv byte kcontrols') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index dde8d3c..2c4405e 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -885,9 +885,9 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, if (bc->params) { if (copy_to_user(data, &bc->param_id, sizeof(u32))) return -EFAULT; - if (copy_to_user(data + sizeof(u32), &size, sizeof(u32))) + if (copy_to_user(data + 1, &size, sizeof(u32))) return -EFAULT; - if (copy_to_user(data + 2 * sizeof(u32), bc->params, size)) + if (copy_to_user(data + 2, bc->params, size)) return -EFAULT; }

3 3

[alsa-devel] [patch v2] ASoC: Intel: Skylake: Fix a couple signedness bugs
by Dan Carpenter 16 Dec '15

16 Dec '15

These need to be signed because they hold negative error codes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Fix a typo in the change log. s/unsigned/signed/ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 8cd5cdb..e26f474 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -208,7 +208,7 @@ static unsigned int skl_get_errorcode(struct sst_dsp *ctx) * since get/set_module are called from DAPM context, * we don't need lock for usage count */ -static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_get_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -220,7 +220,7 @@ static unsigned int skl_get_module(struct sst_dsp *ctx, u16 mod_id) return -EINVAL; } -static unsigned int skl_put_module(struct sst_dsp *ctx, u16 mod_id) +static int skl_put_module(struct sst_dsp *ctx, u16 mod_id) { struct skl_module_table *module; @@ -340,7 +340,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id) { - unsigned int usage_cnt; + int usage_cnt; struct skl_sst *skl = ctx->thread_context; int ret = 0;

2 1

[alsa-devel] [PATCH v5 1/2] ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug
by libin.yang＠linux.intel.com 16 Dec '15

16 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> Dynamically bind/unbind the PCM to pin when HDMI/DP monitor hotplug. When monitor is connected, find a proper PCM for the monitor. When monitor is disconnected, unbind the PCM from the pin. The binding policy (use Intel platform as example) is: 1. Try to use the legacy pin-pcm mapping for the device entry 0 of the pin. 2. If step 1 fails, try to bind pin to the backup PCMs. For example, on Intel platform, if DP MST is enabled, 5 PCMs will be created. PCM 3, PCM 7, PCM 8 are supposed to be used by device entry 0 of pin 5, pin 6 and pin 7. PCM 9 and PCM 10 are the backup PCMs. 3. If step 2 fails, try to find any PCM to bind to the pin. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 82 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 80 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19bf782..56b7625 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -73,6 +73,8 @@ struct hdmi_spec_per_cvt { struct hdmi_spec_per_pin { hda_nid_t pin_nid; + /* pin idx, different device entries on the same pin use the same idx */ + int pin_nid_idx; int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; int mux_idx; @@ -85,6 +87,7 @@ struct hdmi_spec_per_pin { struct snd_kcontrol *eld_ctl; struct snd_jack *acomp_jack; /* jack via audio component */ struct hda_pcm *pcm; /* pointer to spec->pcm_rec[n] dynamically*/ + int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ int repoll_count; bool setup; /* the stream has been set up by prepare callback */ int channels; /* current number of channels */ @@ -138,6 +141,8 @@ struct hdmi_spec { struct snd_array pins; /* struct hdmi_spec_per_pin */ struct hda_pcm *pcm_rec[16]; struct mutex pcm_lock; + /* pcm_bitmap means which pcms have been assigned to pins*/ + unsigned long pcm_bitmap; int pcm_used; /* counter of pcm_rec[] */ unsigned int channels_max; /* max over all cvts */ @@ -1671,6 +1676,60 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } +static int hdmi_find_pcm_slot(struct hdmi_spec *spec, + struct hdmi_spec_per_pin *per_pin) +{ + int i; + + /* try the prefer PCM */ + if (!test_bit(per_pin->pin_nid_idx, &spec->pcm_bitmap)) + return per_pin->pin_nid_idx; + + /* have a second try; check the "reserved area" over num_pins */ + for (i = spec->num_pins; i < spec->pcm_used; i++) { + if (!test_bit(i, &spec->pcm_bitmap)) + return i; + } + + /* the last try; check the empty slots in pins */ + for (i = 0; i < spec->num_pins; i++) { + if (!test_bit(i, &spec->pcm_bitmap)) + return i; + } + return -EBUSY; +} + +static void hdmi_attach_hda_pcm(struct hdmi_spec *spec, + struct hdmi_spec_per_pin *per_pin) +{ + int idx; + + /* pcm already be attached to the pin */ + if (per_pin->pcm) + return; + idx = hdmi_find_pcm_slot(spec, per_pin); + if (idx == -ENODEV) + return; + per_pin->pcm_idx = idx; + per_pin->pcm = spec->pcm_rec[idx]; + set_bit(idx, &spec->pcm_bitmap); +} + +static void hdmi_detach_hda_pcm(struct hdmi_spec *spec, + struct hdmi_spec_per_pin *per_pin) +{ + int idx; + + /* pcm already be detached from the pin */ + if (!per_pin->pcm) + return; + idx = per_pin->pcm_idx; + per_pin->pcm_idx = -1; + per_pin->pcm = NULL; + if (idx >= 0 && idx < spec->pcm_used) + clear_bit(idx, &spec->pcm_bitmap); +} + /* update per_pin ELD from the given new ELD; * setup info frame and notification accordingly */ @@ -1679,9 +1738,17 @@ static void update_eld(struct hda_codec *codec, struct hdmi_eld *eld) { struct hdmi_eld *pin_eld = &per_pin->sink_eld; + struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; + if (spec->dyn_pcm_assign) { + if (eld->eld_valid) + hdmi_attach_hda_pcm(spec, per_pin); + else + hdmi_detach_hda_pcm(spec, per_pin); + } + if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1825,13 +1892,19 @@ static void sync_eld_via_acomp(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; + struct hdmi_spec *spec = codec->spec; + int ret; + mutex_lock(&spec->pcm_lock); if (codec_has_acomp(codec)) { sync_eld_via_acomp(codec, per_pin); - return false; /* don't call snd_hda_jack_report_sync() */ + ret = false; /* don't call snd_hda_jack_report_sync() */ } else { - return hdmi_present_sense_via_verbs(per_pin, repoll); + ret = hdmi_present_sense_via_verbs(per_pin, repoll); } + mutex_unlock(&spec->pcm_lock); + + return ret; } static void hdmi_repoll_eld(struct work_struct *work) @@ -1875,6 +1948,11 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pin_nid = pin_nid; per_pin->non_pcm = false; + if (spec->dyn_pcm_assign) + per_pin->pcm_idx = -1; + else + per_pin->pcm_idx = pin_idx; + per_pin->pin_nid_idx = pin_idx; err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: sst: fix sst_memcpy32 wrong with non-4x bytes issue
by Jie Yang 16 Dec '15

16 Dec '15

sst_memcpy32() only copied bytes/4 32bits, which means it dropped the remaining bytes%4 bytes wrongly. Here add copying those missing bytes, first to a 32bits tmp, and then write the tmp to 32bits iomem. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/common/sst-firmware.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 60652cb..d745a78 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -51,8 +51,22 @@ struct sst_dma { static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { + u32 tmp = 0; + int i, m, n; + const u8 *src_byte = src; + + m = bytes / 4; + n = bytes % 4; + /* __iowrite32_copy use 32bit size values so divide by 4 */ - __iowrite32_copy((void *)dest, src, bytes/4); + __iowrite32_copy((void *)dest, src, m); + + if (n) { + for (i = 0; i < n; i++) + tmp |= (u32)*(src_byte + m * 4 + i) << (i * 8); + __iowrite32_copy((void *)(dest + m * 4), &tmp, 1); + } + } static void sst_dma_transfer_complete(void *arg) -- 2.5.0

2 1

[alsa-devel] [PATCH 0/4] ALSA: oxfw: localize stuffs of old firewire-speakers module
by Takashi Sakamoto 16 Dec '15

16 Dec '15

Hi, ALSA oxfw module inherits some stuffs from old firewire-speakers module. Currently, the module's common structure has some members for the stuffs. This is not good because these stuffs are not common between all of supported models. It's better to localize them. The purpose of this patchset is to enable the module to have memory block and push model-specific stuffs to each structure. This patchset is a part of my previous RFCv2 (patch 06-09) and a preparation for rest of the patches: [alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101558.h… Takashi Sakamoto (4): ALSA: oxfw: enable to keep memory block for model-specific structure ALSA: oxfw: move model-specific members from common structure ALSA: oxfw: move model-specific parameters from common structure ALSA: oxfw: rename a structure so that it means backward compatibility to old drivers sound/firewire/oxfw/oxfw-spkr.c | 96 ++++++++++++++++++++++++++--------------- sound/firewire/oxfw/oxfw.c | 31 +++++++------ sound/firewire/oxfw/oxfw.h | 17 +------- 3 files changed, 79 insertions(+), 65 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 0/5] ALSA: oxfw: code refactoring for model-dependent quirks
by Takashi Sakamoto 16 Dec '15

16 Dec '15

Hi, ALSA oxfw driver has got some patches to expand its functionalities. As a result, current code includes some inefficiencies and confusions. This patchset is a code refactoring about model-dependent quirks. This patchset is a part of my previous RFCv2 (patch 01-05) and a preparation for the rest of patches: [alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101558.h… Takashi Sakamoto (5): ALSA: oxfw: rename a file for control elements so that it's for model-specific ALSA: oxfw: rename local functions for control elements so that they represent as local ALSA: oxfw: change function prototype for AV/C Audio Subunit command ALSA: oxfw: reuse driver entry to detect quirk ALSA: oxfw: gather model-dependent conditions to a function sound/firewire/oxfw/Makefile | 4 +- .../firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} | 80 ++++++++++++---------- sound/firewire/oxfw/oxfw.c | 59 +++++++++------- sound/firewire/oxfw/oxfw.h | 6 +- 4 files changed, 86 insertions(+), 63 deletions(-) rename sound/firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} (71%) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: Intel: sst: fix the IRQ locked issue
by Jie Yang 15 Dec '15

15 Dec '15

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> If driver received a message that it can't handle, it won't clear the corresponding bit and unmask interrupt, this may lock the IRQ and DSP can't send message anymore. To fix the issue, we should Always update IMRX after IPC. Here we always clear the DONE/BUSY bit and unmask the IRQ source, even when IPC failures have occurred previously. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Modified-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 31 +++++++++++++------------------ 1 file changed, 13 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index b27f25f..ac60f13 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -778,7 +778,6 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) struct sst_hsw *hsw = sst_dsp_get_thread_context(sst); struct sst_generic_ipc *ipc = &hsw->ipc; u32 ipcx, ipcd; - int handled; unsigned long flags; spin_lock_irqsave(&sst->spinlock, flags); @@ -790,34 +789,30 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) if (ipcx & SST_IPCX_DONE) { /* Handle Immediate reply from DSP Core */ - handled = hsw_process_reply(hsw, ipcx); + hsw_process_reply(hsw, ipcx); - if (handled > 0) { - /* clear DONE bit - tell DSP we have completed */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, - SST_IPCX_DONE, 0); + /* clear DONE bit - tell DSP we have completed */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, + SST_IPCX_DONE, 0); - /* unmask Done interrupt */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, - SST_IMRX_DONE, 0); - } + /* unmask Done interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, 0); } /* new message from DSP */ if (ipcd & SST_IPCD_BUSY) { /* Handle Notification and Delayed reply from DSP Core */ - handled = hsw_process_notification(hsw); + hsw_process_notification(hsw); /* clear BUSY bit and set DONE bit - accept new messages */ - if (handled > 0) { - sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, - SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, + SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); - /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, - SST_IMRX_BUSY, 0); - } + /* unmask busy interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, 0); } spin_unlock_irqrestore(&sst->spinlock, flags); -- 2.5.0

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[alsa-devel] [PATCH 1/3] rcar: ctu: Avoid use of ret uninitialised
by Mark Brown 15 Dec '15

15 Dec '15

We use ret as the return value from the rsnd_ctu_probe() but if there are no child nodes and no errors then we will never initialize ret leading to build warnings. Ensure ret is initialized before we iterate over the child nodes to avoid this. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Perhaps this should be an error instead? It's not like it's a sensible configuration... sound/soc/sh/rcar/ctu.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 7c1e190cd389..d53a225d19e9 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -111,6 +111,7 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) priv->ctu = ctu; i = 0; + ret = 0; for_each_child_of_node(node, np) { ctu = rsnd_ctu_get(priv, i); -- 2.6.2

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[alsa-devel] [RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming
by Takashi Sakamoto 15 Dec '15

15 Dec '15

This patchset updates my previous one: [alsa-devel] [RFC][PATCH 0/8] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100670.h… I hope more users to test this patchset in this developing cycle. [Background] I've frequently got private messages from owners of dice-based models since below patchset was merged to mainline: [alsa-devel] [PATCH 00/15 v5] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085142.h… Some of them addressed that ALSA dice driver fails to handle their models, while their experiences are not the same. The summary: * ETIMEOUT at starting streaming * detecting packet discontinuity at starting streaming * ETIMEOUT at probing snd-dice * no sound even if starting streaming This patchset is my solutions for these issues. It consists of four parts; [Patch 01] ETIMEDOUT at starting streaming means that ALSA dice driver has never received dice notification within current timeout (200 msec). This patch expands the timeout up to 2 sec. There's some disadvantages of this change in usage of ALSA PCM API, see commit comment. [Patch 02] Packet discontinuity at starting streaming means that the device fail to get constant clock signals from phase lock loop (PLL) circuit. In this case, ALSA dice driver successfully starts streaming and receives AMDTP packet from the device, therefore any dice notifications are already transferred. It's reasonable to judge that the device has a quirk to transfer the notification before set its PLL circuit steadily. This patch continue to check a register so that the device get phase lock after receiving dice notification. [Patch 03-04] When connected to IEEE 1394 bus, some models generate bus reset after appearing on bus topology. Usually, boot processing of Dice based models consists of three stages; boot loader (RedBoot) is loaded, it loads eCos, the operating system executes vendor's configurations for hardware initialization. It's reasonable to assume that bus reset occurs at each of these stages. Furthermore, I observed that ImpactTwin (TC Electronic produced) changes some registers before/after ALSA dice driver is probed with my RFC v1. This means that the hardware initialization requires a bit time. These patches postpone unit probe processing after continuous bus resets. These patches have advantages to start communication to the device after the hardware initialization. These patches may improve handling Dice II based models such as ImpactTwin and IO26 (Alesis produces). [Patch 05-10] When seeing sound/firewire/dice/dice-interface.h, Dice interface gives no ways for drivers to get all formats of AMDTP stream. Current ALSA dice driver manage to generate 'format cache' at unit probing processing, while this requires to change the state of actual clock on devices. This idea is not good because of ignoring hardware design. This idea is not good in a case that users save their favorite configuration to the device. Furthermore, this is not also good in a case that the stream format is not decided according to sampling transfer frequency only, i.e. data format of digital interface is also dominant to the stream format. These patches add limitation to constrain PCM substream at current sampling transfer frequency. As a result, userspace applications are allowed to request according to current sampling transfer frequency. When users change the frequency, they should use userspace applications with fw character devices to set it in advance. ALSA dice driver gives no way to achieve it in describes reasons. Although ALSA dice driver goes backward with these patchset, we don't beat any actual hardware designs. [Changes from RFC v1] * Patch 03-04 are newly added. This enables to handle bus reset at unit probe processing, and drops one of reasons to constrain PCM substreams at current sampling transfer frequency. * Minor arrangements Takashi Sakamoto (10): ALSA: dice: expand timeout to wait for Dice notification ALSA: dice: wait for ensuring phase lock ALSA: dice: split subaddress check from category check ALSA: dice: postpone probe processing ALSA: dice: limit to current sampling transfer frequency ALSA: dice: limit stream to current sampling transfer frequency. ALSA: dice: add MIDI ports according to current number of MIDI substreams ALSA: dice: get the number of MBLA data channel at opening PCM substream ALSA: dice: purge generating channel cache ALSA: dice: ensure phase lock before starting streaming sound/firewire/dice/dice-midi.c | 25 +++- sound/firewire/dice/dice-pcm.c | 201 ++++++++------------------- sound/firewire/dice/dice-stream.c | 84 ++++++++---- sound/firewire/dice/dice-transaction.c | 158 ++++++++++----------- sound/firewire/dice/dice.c | 242 ++++++++++++--------------------- sound/firewire/dice/dice.h | 11 +- 6 files changed, 293 insertions(+), 428 deletions(-) -- 2.5.0

2 12

[alsa-devel] [PATCH 0/3] Yet more noise fixes for Thinkpad
by Takashi Iwai 15 Dec '15

15 Dec '15

Hi, here is a series of small patches to reduce still existing crackling noises on Thinkpad with Realtek codecs. Takashi Iwai (3): ALSA: hda - Apply click noise workaround for Thinkpads generically ALSA: hda - Set codec to D3 at reboot/shutdown on Thinkpads ALSA: hda - Add a fixup for Thinkpad X1 Carbon 2nd sound/pci/hda/patch_realtek.c | 39 +++++++++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 14 deletions(-) -- 2.6.4

1 3

[alsa-devel] [PATCH] ALSA: hda - Fix headphone mic input on a few Dell ALC293 machines
by David Henningsson 15 Dec '15

15 Dec '15

These laptops support both headphone, headset and mic modes for the 3.5mm jack. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1526330 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8dd2ac1..b745a72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4606,6 +4606,7 @@ enum { ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, + ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, @@ -5169,6 +5170,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE }, + [ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC292_FIXUP_DELL_E7X] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, @@ -5247,11 +5254,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Clean up the code to check bdl_pos_adj option
by Takashi Iwai 15 Dec '15

15 Dec '15

Just a minor cleanup; instead of passing an array, pass the assigned bdl_pos_adj option value directory in struct azx. Also split the code to get the default bdl_pos_adj value for the change that will follow after this. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 3 +-- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++------------- 3 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 22dbfa563919..10c77074b4dc 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1050,8 +1050,7 @@ int azx_bus_init(struct azx *chip, const char *model, if (chip->get_position[0] != azx_get_pos_lpib || chip->get_position[1] != azx_get_pos_lpib) bus->core.use_posbuf = true; - if (chip->bdl_pos_adj) - bus->core.bdl_pos_adj = chip->bdl_pos_adj[chip->dev_index]; + bus->core.bdl_pos_adj = chip->bdl_pos_adj; if (chip->driver_caps & AZX_DCAPS_CORBRP_SELF_CLEAR) bus->core.corbrp_self_clear = true; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c1d28a657f19..a32ec9004edd 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -147,7 +147,7 @@ struct azx { #endif /* flags */ - const int *bdl_pos_adj; + int bdl_pos_adj; int poll_count; unsigned int running:1; unsigned int single_cmd:1; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe9bef339cea..a17bf0467edc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -632,7 +632,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (wallclk < (azx_dev->core.period_wallclk * 5) / 4 && pos % azx_dev->core.period_bytes > azx_dev->core.period_bytes / 2) /* NG - it's below the first next period boundary */ - return chip->bdl_pos_adj[chip->dev_index] ? 0 : -1; + return chip->bdl_pos_adj ? 0 : -1; azx_dev->core.start_wallclk += wallclk; return 1; /* OK, it's fine */ } @@ -1488,6 +1488,17 @@ static void azx_probe_work(struct work_struct *work) azx_probe_continue(&hda->chip); } +static int default_bdl_pos_adj(struct azx *chip) +{ + switch (chip->driver_type) { + case AZX_DRIVER_ICH: + case AZX_DRIVER_PCH: + return 1; + default: + return 32; + } +} + /* * constructor */ @@ -1541,18 +1552,10 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->single_cmd = single_cmd; azx_check_snoop_available(chip); - if (bdl_pos_adj[dev] < 0) { - switch (chip->driver_type) { - case AZX_DRIVER_ICH: - case AZX_DRIVER_PCH: - bdl_pos_adj[dev] = 1; - break; - default: - bdl_pos_adj[dev] = 32; - break; - } - } - chip->bdl_pos_adj = bdl_pos_adj; + if (bdl_pos_adj[dev] < 0) + chip->bdl_pos_adj = default_bdl_pos_adj(chip); + else + chip->bdl_pos_adj = bdl_pos_adj[dev]; err = azx_bus_init(chip, model[dev], &pci_hda_io_ops); if (err < 0) { -- 2.6.4

1 1

[alsa-devel] [PATCH] ASoC: Atmel: ClassD: unregister codec when error occurs
by Songjun Wu 15 Dec '15

15 Dec '15

Add code to unregister codec in probe function, when the error occurs after the codec is registered. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/atmel/atmel-classd.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index dca6141..6107de9 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -640,8 +640,10 @@ static int atmel_classd_probe(struct platform_device *pdev) /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); - if (!card) - return -ENOMEM; + if (!card) { + ret = -ENOMEM; + goto unregister_codec; + } snd_soc_card_set_drvdata(card, dd); platform_set_drvdata(pdev, card); @@ -649,16 +651,20 @@ static int atmel_classd_probe(struct platform_device *pdev) ret = atmel_classd_asoc_card_init(dev, card); if (ret) { dev_err(dev, "failed to init sound card\n"); - return ret; + goto unregister_codec; } ret = devm_snd_soc_register_card(dev, card); if (ret) { dev_err(dev, "failed to register sound card: %d\n", ret); - return ret; + goto unregister_codec; } return 0; + +unregister_codec: + snd_soc_unregister_codec(dev); + return ret; } static int atmel_classd_remove(struct platform_device *pdev) -- 1.7.9.5

3 2

[alsa-devel] Can I stop sound from driver as system error ?
by Kuninori Morimoto 15 Dec '15

15 Dec '15

Hi I wonder can I stop sound from driver side (not from user-land) as system error during playback or capture ? How to do it if we can ? Best regards --- Kuninori Morimoto

2 3

[alsa-devel] Intel Cherry Trail -- No sound in linux
by Zhihe Tu 14 Dec '15

14 Dec '15

Hi All, I'm trying to make the CherryTrail tablet works in ubuntu14.04.3, there's no sound at all, but it works in win8 & android. ... [ 5.463371] cht-bsw-rt5672 cht-bsw-rt5672: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 5.463406] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 5.475786] cht-bsw-rt5672 cht-bsw-rt5672: rt5670-aif1 <-> ssp2-port mapping ok [ 5.507355] hid-generic 0003:413C:2003.0002: input,hidraw1: USB HID v1.10 Keyboard [Dell Dell USB Keyboard] on usb-0000:00:14.0-4.2/input0 [ 5.695892] Console: switching to colour frame buffer device 170x48 [ 5.712458] i915_bpo 0000:00:02.0: fb0: inteldrmfb frame buffer device [ 5.712461] i915_bpo 0000:00:02.0: registered panic notifier [ 5.725983] cfg80211: Calling CRDA to update world regulatory domain [ 5.872710] scsi 0:0:0:0: Direct-Access ASMT 2115 0 PQ: 0 ANSI: 6 [ 5.873522] sd 0:0:0:0: Attached scsi generic sg0 type 0 [ 5.875671] sd 0:0:0:0: [sda] Attached SCSI removable disk [ 5.907038] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.922073] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.935876] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.949941] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.968241] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.968267] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.968292] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.968318] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! ... [ 6.617144] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 6.632519] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 6.647501] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 6.662176] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 6.677049] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 6.692739] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 6.692765] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 6.692792] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 7.541669] init: plymouth-upstart-bridge main process ended, respawning [ 9.804555] Audio Port: ASoC: no backend DAIs enabled for Audio Port [ 10.139952] Audio Port: ASoC: no backend DAIs enabled for Audio Port [ 10.147834] Audio Port: ASoC: no backend DAIs enabled for Audio Port [ 10.159472] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.162179] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.163874] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.164488] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.165994] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.166626] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.168438] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.169549] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.171607] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 10.189856] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 10.204244] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 10.205026] intel_sst_acpi 808622A8:00: Free for str 1 pipe 0x90 [ 10.215195] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 12.182265] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 13.181361] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 13.181374] intel_sst_acpi 808622A8:00: fw returned err -16 [ 14.181735] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 14.181764] intel_sst_acpi 808622A8:00: fw returned err -16 [ 15.181611] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 15.181641] intel_sst_acpi 808622A8:00: fw returned err -16 [ 16.181346] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 16.181376] intel_sst_acpi 808622A8:00: fw returned err -16 [ 16.210605] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 17.209115] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 17.209144] intel_sst_acpi 808622A8:00: fw returned err -16 [ 17.209183] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 18.262035] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.262053] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.260719] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 19.260749] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.313903] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 19.445415] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 19.445862] intel_sst_acpi 808622A8:00: FW sent error response 0x40034 [ 19.445883] intel_sst_acpi 808622A8:00: FW alloc failed ret -4 [ 19.445888] intel_sst_acpi 808622A8:00: sst_get_stream returned err -5 [ 19.445893] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 19.445898] Audio Port: ASoC: prepare FE Audio Port failed [ 20.444493] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.444522] intel_sst_acpi 808622A8:00: fw returned err -16 [ 21.444427] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 21.444456] intel_sst_acpi 808622A8:00: fw returned err -16 [ 22.444126] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 22.444155] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.443906] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 23.443934] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.474492] Audio Port: sst: error code = -22 [ 23.474504] snd_pcm_update_hw_ptr0: 123 callbacks suppressed [ 23.474512] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.474589] Audio Port: sst: error code = -22 [ 23.474596] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.474612] Audio Port: sst: error code = -22 [ 23.474617] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741085] Audio Port: sst: error code = -22 [ 23.741104] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741140] Audio Port: sst: error code = -22 [ 23.741147] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741206] Audio Port: sst: error code = -22 [ 23.741213] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741233] Audio Port: sst: error code = -22 [ 23.741240] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741350] Audio Port: sst: error code = -22 [ 23.741358] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741379] Audio Port: sst: error code = -22 [ 23.741385] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741427] Audio Port: sst: error code = -22 [ 23.741434] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 23.741455] Audio Port: sst: error code = -22 [ 23.785062] Audio Port: sst: error code = -22 [ 23.785108] Audio Port: sst: error code = -22 [ 23.785185] Audio Port: sst: error code = -22 [ 23.785210] Audio Port: sst: error code = -22 [ 23.785343] Audio Port: sst: error code = -22 [ 23.785367] Audio Port: sst: error code = -22 [ 23.785416] Audio Port: sst: error code = -22 [ 28.496573] snd_pcm_update_hw_ptr0: 13 callbacks suppressed [ 28.496582] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.496707] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 29.494259] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 29.494269] intel_sst_acpi 808622A8:00: fw returned err -16 [ 29.494284] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 30.562054] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 30.562065] intel_sst_acpi 808622A8:00: fw returned err -16 [ 31.561892] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 31.561904] intel_sst_acpi 808622A8:00: fw returned err -16 [ 906.106300] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 906.249375] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 907.248864] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 907.248894] intel_sst_acpi 808622A8:00: fw returned err -16 [ 908.248393] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 908.248421] intel_sst_acpi 808622A8:00: fw returned err -16 [ 909.248611] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 909.248641] intel_sst_acpi 808622A8:00: fw returned err -16 [ 910.248108] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 910.248137] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1053.563879] isofs_fill_super: bread failed, dev=sdb1, iso_blknum=16, block=32 [ 1185.717932] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 1186.716247] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 1186.716277] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1186.716315] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 1187.783992] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 1187.784021] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1188.783551] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 1188.783584] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1188.821911] rt5670 i2c-10EC5670:00: Sysclk is 0Hz and clock id is 2 [ 1207.389279] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 1207.404620] rt5670 i2c-10EC5670:00: Sysclk is 24576000Hz and clock id is 1 [ 1207.404646] rt5670 i2c-10EC5670:00: bclk is 1536000Hz and lrck is 48000Hz [ 1207.404660] rt5670 i2c-10EC5670:00: bclk_ms is 0 and pre_div is 1 for iis 1 [ 1207.533606] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 1208.535662] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 1208.535691] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1209.535853] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 1209.535883] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1210.535333] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 1210.535361] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1211.535576] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 1211.535606] intel_sst_acpi 808622A8:00: fw returned err -16 [ 1286.976450] usb 1-4.1: USB disconnect, device number 8 [11996.101738] r8169 0000:01:00.0 eth0: link up [11996.101780] IPv6: ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [12019.249846] [drm:intel_pipe_update_end [i915_bpo]] *ERROR* Atomic update failure on pipe A (start=93814 end=93815) [12157.754218] [drm:intel_pipe_update_end [i915_bpo]] *ERROR* Atomic update failure on pipe A (start=102079 end=102080) [12194.085543] [drm:intel_pipe_update_end [i915_bpo]] *ERROR* Atomic update failure on pipe A (start=104247 end=104248) [13567.332608] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [13568.332857] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [13568.332886] intel_sst_acpi 808622A8:00: fw returned err -16 [13568.332924] intel_sst_acpi 808622A8:00: free stream returned err -19 [13569.400219] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [13569.400247] intel_sst_acpi 808622A8:00: fw returned err -16 [13570.399826] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [13570.399837] intel_sst_acpi 808622A8:00: fw returned err -16 [13570.437188] rt5670 i2c-10EC5670:00: Sysclk is 0Hz and clock id is 2 >From /sys/kernel/debug/asoc/cherrytrailcraudio/, some necessary widgets are all power on, from media1_in to ssp2 Tx end with Headphone. I really don't know how to fix the sst fw error, any anyone give a look at this issue? ................ Thanks Tu __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com

3 7

[alsa-devel] [PATCH v3] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by me＠mx0a-001ae601.pphosted.com 14 Dec '15

14 Dec '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 933 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 272 ++++++ 5 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3317da..f59aa4b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -444,6 +445,10 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cdadb9b..4f1ab1d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -51,6 +51,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..3d5cac9 +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,933 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, THMS_PDN }, + { CS53L30_MCLKCTL, MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, INTERNAL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, ASP_RATE_48K }, + { CS53L30_ASP1_CTL, ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, MUTEP_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, INBIAS_CTLX_DFLT }, + { CS53L30_INBIAS_CTL2, INBIAS_CTLX_DFLT }, + { CS53L30_DMIC1_STR_CTL, DMIC1_STEREO_DFLT }, + { CS53L30_DMIC2_STR_CTL, DMIC2_STEREO_DFLT }, + { CS53L30_ADCDMIC1_CTL1, ADC1_ON_CH_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, ADC1_DMIC1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, ADC2_ON_CH_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +unsigned int const input1_sel_values[] = { DMIC1_ON_CH_AB_IN, DMIC1_ON_CH_A_IN, + DMIC1_ON_CH_B_IN, ADC1_ON_CH_AB_IN, ADC1_ON_CH_A_IN, ADC1_ON_CH_B_IN, + DMIC1_OFF_ADC1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +unsigned int const input2_sel_values[] = { DMIC2_ON_CH_AB_IN, DMIC2_ON_CH_A_IN, + DMIC2_ON_CH_B_IN, ADC2_ON_CH_AB_IN, ADC2_ON_CH_A_IN, ADC2_ON_CH_B_IN, + DMIC2_OFF_ADC2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 1); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, + CS53L30_ASP1_CTL, ASP1_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, + CS53L30_ASP2_CTL, ASP2_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= ASP_SCLK_INV; + else + asp_config_ctl &= ~ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = + cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_write(priv->regmap, CS53L30_ASPCFG_CTL, priv->asp_config_ctl); + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= INTERNAL_FS_RATIO; + else + int_sr_ctl &= ~INTERNAL_FS_RATIO; + regmap_write(priv->regmap, CS53L30_INT_SR_CTL, int_sr_ctl); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= MCLK_INT_SCALE; + else + mclk_ctl &= ~MCLK_INT_SCALE; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->asp_config_ctl &= ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & ASP_RATE); + regmap_write(priv->regmap, CS53L30_MCLKCTL, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & DIGSFT) + inter_max_check = PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + usleep_range(1000, 1000); + msleep(1); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & PDN_DONE) + break; + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, MCLK_DIS); + break; + } + dapm->bias_level = level; + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, ASP1_3ST, + (ASP1_3ST_VAL(tristate) & ASP1_3ST)); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .owner = THIS_MODULE, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..f45e91f +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,272 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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 __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define PDN_ULP (1 << 7) +#define PDN_LP (1 << 6) +#define DISCHARGE_FILT (1 << 5) +#define THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define MCLK_DIS (1 << 7) +#define MCLK_INT_SCALE (1 << 6) +#define DMIC_DRIVE (1 << 5) +#define MCLK_DIV (3 << 2) +#define MCLK_DIV_DFLT (1 << 2) +#define SYNC_EN (1 << 1) + +/* CS53L30_INT_SR_CTL */ +#define INTERNAL_FS_RATIO (1 << 4) +#define INTERNAL_FS_DFLT (7 << 2) +#define MCLK_19MHZ_EN (1 << 0) + +/* CS53L30_MICBIAS_CTL */ +#define MIC4_BIAS_PDN (1 << 7) +#define MIC3_BIAS_PDN (1 << 6) +#define MIC2_BIAS_PDN (1 << 5) +#define MIC1_BIAS_PDN (1 << 4) +#define VP_MIN (1 << 2) +#define MIC_BIAS_CTRL (3 << 0) +#define MIC_BIAS_ALL_PDN 0xF0 +#define MIC_BIAS_DFLT (MIC_BIAS_ALL_PDN | VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define ASP_MS (1 << 7) +#define ASP_SCLK_INV (1 << 4) +#define ASP_RATE_48K (3 << 2) +#define ASP_RATE 0x0F +#define ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define ASP1_TDM_PDN (1 << 7) +#define ASP1_SDOUT_PDN (1 << 6) +#define ASP1_3ST (1 << 5) +#define SHIFT_LEFT (1 << 4) +#define ASP1_DRIVE (1 << 0) +#define ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define ASP1_CHX_TX_STATE (1 << 7) +#define ASP1_CHX_TX_LOC 0x3F +#define ASP1_CHX_TX_DFLT_SLT47 0x2F +#define ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define ASP2_SDOUT_PDN (1 << 6) +#define ASP2_DRIVE (1 << 0) +#define ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define DIGSFT (1 << 5) +#define SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define LRCK_50_NPW (1 << 3) +#define LRCK_TPWH (7 << 0) +#define LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define MUTE_PDN_ULP (1 << 7) +#define MUTE_PDN_LP (1 << 6) +#define MUTE_M4B_PDN (1 << 4) +#define MUTE_M3B_PDN (1 << 3) +#define MUTE_M2B_PDN (1 << 2) +#define MUTE_M1B_PDN (1 << 1) +#define MUTE_MB_ALL_PDN (1 << 0) +#define MUTEP_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define MUTE_PIN_POLARITY (1 << 7) +#define MUTE_ASP_TDM_PDN (1 << 6) +#define MUTE_ASP_SDOUT2_PDN (1 << 5) +#define MUTE_ASP_SDOUT1_PDN (1 << 4) +#define MUTE_ADC2B_PDN (1 << 3) +#define MUTE_ADC2A_PDN (1 << 2) +#define MUTE_ADC1B_PDN (1 << 1) +#define MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define IN4M_BIAS (3 << 6) +#define IN4P_BIAS (3 << 4) +#define IN3M_BIAS (3 << 2) +#define IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define IN2M_BIAS (3 << 6) +#define IN2P_BIAS (3 << 4) +#define IN1M_BIAS (3 << 2) +#define IN1P_BIAS (3 << 0) +#define INBIAS_CTLX_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define DMIC1_STEREO_ENB (1 << 5) +#define DMIC1_STEREO_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define DMIC2_STEREO_EN (1 << 5) +#define DMIC2_STEREO_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define ADC1B_PDN (1 << 7) +#define ADC1A_PDN (1 << 6) +#define DMIC1_PDN (1 << 2) +#define DMIC1_SCLK_DIV (1 << 1) +#define CH_TYPE (1 << 0) +#define DMIC1_ON_CH_AB_IN (CH_TYPE) +#define DMIC1_ON_CH_A_IN (ADC1B_PDN | CH_TYPE) +#define DMIC1_ON_CH_B_IN (ADC1A_PDN | CH_TYPE) +#define ADC1_ON_CH_AB_IN (DMIC1_PDN) +#define ADC1_ON_CH_A_IN (ADC1B_PDN | DMIC1_PDN) +#define ADC1_ON_CH_B_IN (ADC1A_PDN | DMIC1_PDN) +#define DMIC1_OFF_ADC1_OFF (ADC1A_PDN | ADC1B_PDN | DMIC1_PDN) +#define ADC1_DMIC1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define ADC1_NOTCH_DIS (1 << 7) +#define ADC1B_INV (1 << 5) +#define ADC1A_INV (1 << 4) +#define ADC1B_DIG_BOOST (1 << 1) +#define ADC1A_DIG_BOOST (1 << 0) +#define ADC1_DMIC1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define ADC1_HPF_EN (1 << 3) +#define ADC1_HPF_CF (3 << 1) +#define ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define ADC1B_NG (1 << 7) +#define ADC1A_NG (1 << 6) +#define ADC1_NG_BOOST (1 << 5) +#define ADC1_NG_THRESH (7 << 2) +#define ADC1_NG_DELAY (3 << 0) +#define ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define ADC1A_PREAMP (3 << 6) +#define ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define ADC1B_PREAMP (3 << 6) +#define ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define ADC2B_PDN (1 << 7) +#define ADC2A_PDN (1 << 6) +#define DMIC2_PDN (1 << 2) +#define DMIC2_CLKDIV (1 << 1) +#define DMIC2_ON_CH_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_A_IN (ADC2B_PDN) /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_B_IN (ADC2A_PDN) /* CH_TYPE must = 1 */ +#define ADC2_ON_CH_AB_IN (DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_A_IN (ADC2B_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_B_IN (ADC2A_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define DMIC2_OFF_ADC2_OFF (ADC2A_PDN | ADC2B_PDN | DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define ADC2_NOTCH_DIS (1 << 7) +#define ADC2B_INV (1 << 5) +#define ADC2A_INV (1 << 4) +#define ADC2B_DIG_BOOST (1 << 1) +#define ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define ADC2_HPF_EN (1 << 3) +#define ADC2_HPF_CF (3 << 1) +#define ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define PDN_DONE (1 << 7) +#define THMS_TRIP (1 << 6) +#define SYNC_DONE (1 << 5) +#define ADC2B_OVFL (1 << 4) +#define ADC2A_OVFL (1 << 3) +#define ADC1B_OVFL (1 << 2) +#define ADC1A_OVFL (1 << 1) +#define MUTE_PIN (1 << 0) +#define DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

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[alsa-devel] vsp1 BUG_ON() and crash (Re: [PATCH v9 03/12] media: Entities, pads and links)
by Geert Uytterhoeven 14 Dec '15

14 Dec '15

On Mon, Feb 14, 2011 at 1:20 PM, Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> wrote: > +media_entity_create_link(struct media_entity *source, u16 source_pad, > + struct media_entity *sink, u16 sink_pad, u32 flags) > +{ > + struct media_link *link; > + struct media_link *backlink; > + > + BUG_ON(source == NULL || sink == NULL); > + BUG_ON(source_pad >= source->num_pads); > + BUG_ON(sink_pad >= sink->num_pads); This triggers on r8a7791/koelsch: vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 bru was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 hsi was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 hst was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 lut was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.0 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.1 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.2 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.3 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 rpf.4 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 sru was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 uds.0 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 uds.1 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 uds.2 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.0 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.1 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.2 was not initialized! vsp1 fe928000.vsp1: Entity type for entity fe928000.vsp1 wpf.3 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 bru was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 hsi was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 hst was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 lif was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 lut was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.0 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.1 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.2 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 rpf.3 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 uds.0 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.0 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.1 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.2 was not initialized! vsp1 fe930000.vsp1: Entity type for entity fe930000.vsp1 wpf.3 was not initialized! ------------[ cut here ]------------ kernel BUG at drivers/media/media-entity.c:521! Internal error: Oops - BUG: 0 [#1] SMP ARM CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.4.0-rc5-03225-g596de581263544d2 #400 Hardware name: Generic R8A7791 (Flattened Device Tree) task: ee84fb40 ti: ee850000 task.ti: ee850000 PC is at media_create_pad_link+0x50/0x13c LR is at vsp1_wpf_create_links+0x4c/0x54 pc : [<c034fd90>] lr : [<c03761cc>] psr: 60000013 sp : ee851d50 ip : ee851d80 fp : ee851d7c r10: 00000001 r9 : ee9e9600 r8 : 00000000 r7 : ee111c80 r6 : 00000000 r5 : ee114a40 r4 : ee114b18 r3 : 00000000 r2 : ee114b18 r1 : 00000001 r0 : ee114a40 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none Control: 10c5307d Table: 4000406a DAC: 00000051 Process swapper/0 (pid: 1, stack limit = 0xee850210) Stack: (0xee851d50 to 0xee852000) 1d40: 00000000 ee111c10 ee114a10 ee111c90 1d60: ee111c80 ee9e9610 ee9e9600 ee111c98 ee851d94 ee851d80 c03761cc c034fd4c 1d80: 00000001 ee111c98 ee851de4 ee851d98 c03730c0 c037618c 00000000 ee9eac00 1da0: ee111c10 ee851db0 00000000 ee114840 ee114c40 00000000 c071b678 00000000 1dc0: ee9e9610 c06d3414 c071b678 c06d3414 00000000 00000000 ee851e04 ee851de8 1de0: c028f2b0 c0372bbc c028f258 ee9e9610 c071b668 00000000 ee851e2c ee851e08 1e00: c028dd4c c028f264 ee9e9610 ee9e9644 c06d3414 c06cb3a8 c069e010 c06e2000 1e20: ee851e4c ee851e30 c028df04 c028dc1c 00000001 00000000 c06d3414 c028de94 1e40: ee851e74 ee851e50 c028c47c c028dea0 ee884f5c ee9e7bb4 ee884f70 c06d3414 1e60: 00000000 eeaeb980 ee851e84 ee851e78 c028e138 c028c414 ee851eac ee851e88 1e80: c028cc18 c028e124 c0646a6b ee851e98 c06d3414 c0681dc8 00000000 c069e010 1ea0: ee851ec4 ee851eb0 c028e848 c028cb50 ee0bd380 c0681dc8 ee851ed4 ee851ec8 1ec0: c028fcbc c028e7b0 ee851ee4 ee851ed8 c0681de0 c028fc90 ee851f5c ee851ee8 1ee0: c0664df8 c0681dd4 ee851f0c ee851ef8 c0664604 c01c94bc efffcc00 efffcc10 1f00: ee851f5c ee851f10 c003f90c c06645f4 00000000 c0661740 c0661740 00000006 1f20: 00000006 000000b6 c0660908 efffcc38 ee851f5c 00000006 00000006 000000b6 1f40: c069e83c c06aad08 c06e2000 c06e2000 ee851f94 ee851f60 c0664fd4 c0664cfc 1f60: 00000006 00000006 00000000 c06645e8 00000000 c04e752c 00000000 00000000 1f80: 00000000 00000000 ee851fac ee851f98 c04e753c c0664eb8 ee850000 00000000 1fa0: 00000000 ee851fb0 c000fe98 c04e7538 00000000 00000000 00000000 00000000 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 Backtrace: [<c034fd40>] (media_create_pad_link) from [<c03761cc>] (vsp1_wpf_create_links+0x4c/0x54) r10:ee111c98 r9:ee9e9600 r8:ee9e9610 r7:ee111c80 r6:ee111c90 r5:ee114a10 r4:ee111c10 r3:00000000 [<c0376180>] (vsp1_wpf_create_links) from [<c03730c0>] (vsp1_probe+0x510/0x700) [<c0372bb0>] (vsp1_probe) from [<c028f2b0>] (platform_drv_probe+0x58/0xa8) r10:00000000 r9:00000000 r8:c06d3414 r7:c071b678 r6:c06d3414 r5:ee9e9610 r4:00000000 While bisecting, I arrived at a NULL pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 00000250 pgd = c0004000 [00000250] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.4.0-rc2-00170-gf9836d92a3c8c143 #417 ata1: SATA link down (SStatus 0 SControl 300) Hardware name: Generic R8A7791 (Flattened Device Tree) task: ee84fb40 ti: ee850000 task.ti: ee850000 PC is at media_gobj_init+0x20/0x68 LR is at media_entity_create_link+0xe4/0x108 pc : [<c0342428>] lr : [<c0342a80>] psr: 60000013 sp : ee851d10 ip : ee851d20 fp : ee851d1c r10: 00000000 r9 : ee103890 r8 : 00000000 r7 : ee159920 r6 : ee103740 r5 : ee103840 r4 : ee159840 r3 : 00000003 r2 : ee103840 r1 : 00000002 r0 : 00000000 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none Control: 10c5307d Table: 4000406a DAC: 00000051 Process swapper/0 (pid: 1, stack limit = 0xee850210) Stack: (0xee851d10 to 0xee852000) 1d00: ee851d4c ee851d20 c0342a80 c0342414 1d20: 00000000 ee159810 00000000 ee159840 ee159910 00000000 ee9e6800 ee102898 1d40: ee851d94 ee851d50 c036506c c03429a8 00000003 00000000 000000ff 00000000 1d60: 00000001 00000000 000000ff 00000000 c0366740 ee102810 ee102890 00000000 1d80: ee102858 ee9e6810 ee851de4 ee851d98 c0362a44 c0364ef4 00000080 ee9e8f00 1da0: ee102810 ee851db0 c012bcac ee9e6810 ee9e6810 c06b1f88 c06e76b4 00000000 1dc0: ee9e6810 c06b1f88 c06e76b4 00000000 c06b1f88 00000000 ee851e04 ee851de8 1de0: c0285b14 c0362694 c0285abc ee9e6810 00000000 c06e76a8 ee851e2c ee851e08 1e00: c028457c c0285ac8 ee9e6810 ee9e6844 c06b1f88 c06aab38 c067d010 c06c1000 1e20: ee851e4c ee851e30 c0284770 c028446c 00000001 00000000 c06b1f88 c0284700 1e40: ee851e74 ee851e50 c0282d78 c028470c ee884f5c ee9e4cb4 ee884f70 c06b1f88 1e60: 00000000 ee9c4700 ee851e84 ee851e78 c028499c c0282d10 ee851eac ee851e88 1e80: c0283514 c0284988 c0627e00 ee851e98 c06b1f88 c06618a8 00000000 c067d010 1ea0: ee851ec4 ee851eb0 c02850ac c028344c eea9a200 c06618a8 ee851ed4 ee851ec8 1ec0: c02864e0 c0285014 ee851ee4 ee851ed8 c06618c0 c02864b4 ee851f5c ee851ee8 1ee0: c0644df8 c06618b4 ee851f0c ee851ef8 c0644604 c01c91bc efffcc00 efffcc10 1f00: ee851f5c ee851f10 c003f908 c06445f4 00000000 c0641c1c c0641c1c 00000006 1f20: 00000006 000000b1 c0640e48 efffcc38 ee851f5c 00000006 00000006 000000b1 1f40: c067d83c c0689c84 c06c1000 c06c1000 ee851f94 ee851f60 c0644fd4 c0644cfc 1f60: 00000006 00000006 00000000 c06445e8 00000000 c04d00ac 00000000 00000000 1f80: 00000000 00000000 ee851fac ee851f98 c04d00bc c0644eb8 ee850000 00000000 1fa0: 00000000 ee851fb0 c000fe98 c04d00b8 00000000 00000000 00000000 00000000 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 Backtrace: [<c0342408>] (media_gobj_init) from [<c0342a80>] (media_entity_create_link+0xe4/0x108) [<c034299c>] (media_entity_create_link) from [<c036506c>] (vsp1_rpf_create+0x184/0x1c8) r10:ee102898 r9:ee9e6800 r8:00000000 r7:ee159910 r6:ee159840 r5:00000000 r4:ee159810 r3:00000000 [<c0364ee8>] (vsp1_rpf_create) from [<c0362a44>] (vsp1_probe+0x3bc/0x6c0) r8:ee9e6810 r7:ee102858 r6:00000000 r5:ee102890 r4:ee102810 [<c0362688>] (vsp1_probe) from [<c0285b14>] (platform_drv_probe+0x58/0xa8) r10:00000000 r9:c06b1f88 r8:00000000 r7:c06e76b4 r6:c06b1f88 r5:ee9e6810 r4:00000000 The NULL pointer dereference was introduced by: commit f9836d92a3c8c143c6f5fe89c558c37767628fa6 Author: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> Date: Fri Aug 14 12:54:36 2015 -0300 [media] media: use media_gobj inside links and triggers until the following commit, after which the BUG_ON() takes over: commit eb9e998983338811c3328294ed2f093367161df1 Author: Javier Martinez Canillas <javier(a)osg.samsung.com> Date: Thu Sep 3 12:19:25 2015 -0300 [media] v4l: vsp1: separate links creation from entities init Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH v2] sound: codecs: pcm1792a: clean function exit
by Raphael Poggi 14 Dec '15

14 Dec '15

Signed-off-by: Raphael Poggi <poggi.raph(a)gmail.com> --- sound/soc/codecs/pcm1792a.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 08bb486..27d54bc 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -91,14 +91,9 @@ static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); - int ret; - ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + return regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, PCM1792A_MUTE_MASK, !!mute); - if (ret < 0) - return ret; - - return 0; } static int pcm1792a_hw_params(struct snd_pcm_substream *substream, @@ -107,7 +102,7 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); - int val = 0, ret; + int val = 0; priv->rate = params_rate(params); @@ -145,12 +140,8 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; - ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + return regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); - if (ret < 0) - return ret; - - return 0; } static const struct snd_soc_dai_ops pcm1792a_dai_ops = { -- 2.1.0

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[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Add a volume scale quirk for AudioQuest DragonFly
by Anssi Hannula 14 Dec '15

14 Dec '15

AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and causes software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. The actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of -53...0 dB. Add a quirk for DragonFly to use a custom dB mapping, based on my measurements, using a 10-item range TLV (so it still fits in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- sound/usb/mixer_quirks.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317ead6f..39d7f34e44e6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include <sound/control.h> #include <sound/hwdep.h> #include <sound/info.h> +#include <sound/tlv.h> #include "usbaudio.h" #include "mixer.h" @@ -1733,6 +1734,38 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return 0; } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer) +{ + struct usb_mixer_elem_list *list; + struct usb_mixer_elem_info *cval; + static const int unit_id = 7; + + /* approximation using 10 ranges based on output measurement on hw v1.2 */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + for (list = mixer->id_elems[unit_id]; list; list = list->next_id_elem) { + cval = (struct usb_mixer_elem_info *)list; + if (cval->control == UAC_FU_VOLUME && + cval->min == 0 && cval->max == 50) { + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + list->kctl->tlv.p = scale; + list->kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + list->kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1810,6 +1843,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ err = snd_scarlett_controls_create(mixer); break; + + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + snd_dragonfly_quirk_db_scale(mixer); + break; } return err; -- 2.3.8

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[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi support dynmaic pcm assignment
by libin.yang＠linux.intel.com 14 Dec '15

14 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> The patches are trying to support dynamically binding PCM to pin in HDMI/DP audio. The pin will not bind to any PCM if there is not monitor connected. When there is a monitor connected to the port (pin), driver will try to find a proper PCM to assign to the port and setup the pin if necessary. Libin Yang (5): ALSA: hda - hdmi begin to support dynamic PCM assignment ALSA: hda - hdmi playback without monitor in dynamic pcm bind mode ALSA: hda - hdmi operate spdif based on pcm ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug ALSA: hda - hdmi setup pin when monitor hotplug in pcm dynamic assignment mode sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 369 ++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 347 insertions(+), 23 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 00/13] ASoC: Intel: Skylake: Update to SKL driver
by Vinod Koul 14 Dec '15

14 Dec '15

This update bring in new APIs in core and usage of those in SKL driver. The series is broadly of three parts - first two are fixes on the driver - then we add DMA resume capability in HDA core and use it to save restore DMA link values and add DMA resume capability in driver - lastly, we add support for MISCBDCGE configuration in core and use it while resetting and also for phrase detection per HW recommendation This should be merged thru ASoC tree due to obvious dependencies. Takashi would need your ack on core patches change is v3: - fix the missing export symbol, reported by lkp changes in v2: - move msicbdge api to driver and as a consequence sqash into into first use - fix lkp warning Dharageswari.R (1): ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification Jayachandran B (2): ALSA: hdac: Increase timeout value for link power check ASoC: Intel: Skylake: fix reset controller sequencing Jeeja KP (8): ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting ALSA: hdac: Add support for hda DMA Resume capability ALSA: hdac: couple the hda DMA stream in cleanup ASoC: Intel: Skylake: enable interrupt as wake source in active suspend ASoC: Intel: Skylake: Add DMA resume position in Trigger resume/suspend ASoC: Intel: Skylake: Reconfigure Link stream on suspend/resume ASoC: Intel: Skylake: Add Resume capability in PCM info. Vinod Koul (2): ALSA: hdac: add snd_hdac_ext_bus_link_power_up_all ASoC: Intel: Skylake: manage link power in active suspend include/sound/hda_register.h | 9 +++++ include/sound/hdaudio_ext.h | 15 +++++++ sound/hda/ext/hdac_ext_controller.c | 29 ++++++++++++- sound/hda/ext/hdac_ext_stream.c | 72 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 2 + sound/soc/intel/skylake/skl-pcm.c | 56 ++++++++++++++++++++----- sound/soc/intel/skylake/skl-sst-cldma.c | 38 ++++++++++------- sound/soc/intel/skylake/skl-sst-ipc.c | 15 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 3 ++ sound/soc/intel/skylake/skl.c | 58 ++++++++++++++++++++++++-- sound/soc/intel/skylake/skl.h | 5 +++ 11 files changed, 272 insertions(+), 30 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH v2 08/13] ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification
by Vinod Koul 14 Dec '15

14 Dec '15

On Sun, Dec 13, 2015 at 02:46:08AM +0800, kbuild test robot wrote: > Hi Dharageswari.R, > > [auto build test ERROR on asoc/for-next] > [also build test ERROR on next-20151211] > [cannot apply to v4.4-rc4] > > url: https://github.com/0day-ci/linux/commits/Vinod-Koul/ASoC-Intel-Skylake-Upda… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: i386-allmodconfig (attached as .config) > reproduce: > # save the attached .config to linux build tree > make ARCH=i386 > > All errors (new ones prefixed by >>): > > >> ERROR: "skl_enable_miscbdcge" undefined! I will update this one, thanks for reporting -- ~Vinod

1 0

[alsa-devel] [PATCH] ASoC: es8328: Fix shifts for mixer switches
by John Keeping 13 Dec '15

13 Dec '15

These are all off by one; the playback and bypass switches are the top two bits of the registers, which are at shifts 7 and 6 not 8 and 7. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/es8328.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 969e337..84f5eb0 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -205,18 +205,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), - SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), }; /* Right Mixer */ static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { - SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; static const char * const es8328_pga_sel[] = { -- 2.6.3.462.gbe2c914

2 1

[alsa-devel] [PATCH] ASoC: wm8974: set cache type for regmap
by Mans Rullgard 13 Dec '15

13 Dec '15

Attempting to use this codec driver triggers a BUG() in regcache_sync() since no cache type is set. The register map of this device is fairly small and has few holes so a flat cache is suitable. Signed-off-by: Mans Rullgard <mans(a)mansr.com> --- sound/soc/codecs/wm8974.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 45ba828..c284c7b 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -574,6 +574,7 @@ static const struct regmap_config wm8974_regmap = { .max_register = WM8974_MONOMIX, .reg_defaults = wm8974_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults), + .cache_type = REGCACHE_FLAT, }; static int wm8974_probe(struct snd_soc_codec *codec) -- 2.6.3

3 2

[alsa-devel] [PATCH] ASoC: rockchip: spdif: Set transmit data level to 16 samples
by Sjoerd Simons 13 Dec '15

13 Dec '15

Explicitly set the transmit data level on the transceiver to 16 samples rather then the default 0. This matches both the level set in the vendor kernel and the (seemingly very similar) i2s engine. This fixes audio glitches when playing back at 192k rate. At the same time, fix a trivial typo in the TDL mask definition Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> --- sound/soc/rockchip/rockchip_spdif.c | 6 ++++-- sound/soc/rockchip/rockchip_spdif.h | 2 +- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ac72ff5..5a806da 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -152,8 +152,10 @@ static int rk_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, - SPDIF_DMACR_TDE_ENABLE, - SPDIF_DMACR_TDE_ENABLE); + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL_MASK, + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL(16)); if (ret != 0) return ret; diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h index 921b409..3ef1277 100644 --- a/sound/soc/rockchip/rockchip_spdif.h +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -42,7 +42,7 @@ #define SPDIF_DMACR_TDL_SHIFT 0 #define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT) -#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT) +#define SPDIF_DMACR_TDL_MASK (0x1f << SPDIF_DMACR_TDL_SHIFT) /* * XFER -- 2.6.2

2 1

[alsa-devel] [PATCH V2 0/3] Add support for Texas Instruments pcm3168a codec
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add SOC_DOUBLE_STS macro Add driver and binding docuemnt for Texas Instruments pcm3168a codec Changes in V2: Addressed comments from Mark Brown Damien.Horsley (3): ASoC: Add SOC_DOUBLE_STS macro ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 ++ include/sound/soc.h | 8 + sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3168a-i2c.c | 66 ++ sound/soc/codecs/pcm3168a-spi.c | 65 ++ sound/soc/codecs/pcm3168a.c | 767 +++++++++++++++++++++ sound/soc/codecs/pcm3168a.h | 100 +++ 8 files changed, 1077 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h -- 2.1.4

2 5

[alsa-devel] [PATCH 0/2] Add support for Texas Instruments pcm3168a codec
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add driver and binding docuemnt for Texas Instruments pcm3168a codec Damien.Horsley (2): ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 ++ sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3168a-i2c.c | 66 ++ sound/soc/codecs/pcm3168a-spi.c | 65 ++ sound/soc/codecs/pcm3168a.c | 766 +++++++++++++++++++++ sound/soc/codecs/pcm3168a.h | 111 +++ 7 files changed, 1079 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h -- 2.1.4

2 6

[alsa-devel] [PATCH 0/2] Add support for Pistachio internal DAC
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 350 +++++++++++++++++++++ 4 files changed, 378 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

2 6

[alsa-devel] [PATCH V4 0/2] Add driver for Pistachio internal DAC
by Damien Horsley 13 Dec '15

13 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Changes in V4: Renamed Mute control and change definition to inverted Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 287 +++++++++++++++++++++ 4 files changed, 315 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

2 3

[alsa-devel] [RFC PATCH] ASoC: dpcm: Apply symmetry for DPCM
by Koro Chen 13 Dec '15

13 Dec '15

From: PC Liao <pc.liao(a)mediatek.com> DPCM does not fully support symmetry attributes. soc_pcm_apply_symmetry() is skipped in soc_pcm_open() for DPCM, without being applied elsewhere. So HW parameters cannot be correctly limited, and user space can do playback/capture at different rates while HW actually does not support it. soc_pcm_params_symmetry() will return error and the second stream stops. This patch adds soc_pcm_apply_symmetry() for FE, BE, and codec DAIs in DPCM path that was skipped in soc_pcm_open(). Signed-off-by: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-pcm.c | 57 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 67ef1a0..2a2ca22 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1614,6 +1614,56 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, snd_pcm_stream_unlock_irq(substream); } +static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, + int stream) +{ + struct snd_soc_dpcm *dpcm; + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dai *fe_cpu_dai = fe->cpu_dai; + int err; + + /* apply symmetry for FE */ + if (soc_pcm_has_symmetry(fe_substream)) + fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + + /* Symmetry only applies if we've got an active stream. */ + if (fe_cpu_dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); + if (err < 0) + return err; + } + + /* apply symmetry for BE */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + int i; + + if (soc_pcm_has_symmetry(be_substream)) + be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + + /* Symmetry only applies if we've got an active stream. */ + if (rtd->cpu_dai->active) { + err = soc_pcm_apply_symmetry(be_substream, rtd->cpu_dai); + if (err < 0) + return err; + } + + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->active) { + err = soc_pcm_apply_symmetry(be_substream, + rtd->codec_dais[i]); + if (err < 0) + return err; + } + } + } + + return 0; +} + static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; @@ -1642,6 +1692,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_runtime(fe_substream); snd_pcm_limit_hw_rates(runtime); + ret = dpcm_apply_symmetry(fe_substream, stream); + if (ret < 0) { + dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n", + ret); + goto unwind; + } + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; -- 1.7.9.5

3 2

[alsa-devel] [PATCH 181/182] ASoC: ac97: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/soc-ac97.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index a2012652f212..eaec7177eab2 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -59,8 +59,7 @@ static void soc_ac97_device_release(struct device *dev) #ifdef CONFIG_GPIOLIB static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip) { - struct snd_ac97_gpio_priv *gpio_priv = - container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); return gpio_priv->codec; } @@ -98,8 +97,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct snd_ac97_gpio_priv *gpio_priv = - container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); struct snd_soc_codec *codec = gpio_to_codec(chip); gpio_priv->gpios_set &= ~(1 << offset); @@ -145,7 +143,7 @@ static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, gpio_priv->gpio_chip.parent = codec->dev; gpio_priv->gpio_chip.base = -1; - ret = gpiochip_add(&gpio_priv->gpio_chip); + ret = gpiochip_add_data(&gpio_priv->gpio_chip, gpio_priv); if (ret != 0) dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); return ret; -- 2.4.3

2 1

[alsa-devel] [PATCH 180/182] ASoC: wm8996: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm8996.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8d7d6c01a2f7..f9656df9dccc 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio/driver.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -2139,14 +2140,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } #ifdef CONFIG_GPIOLIB -static inline struct wm8996_priv *gpio_to_wm8996(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8996_priv, gpio_chip); -} - static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, WM8996_GP1_LVL, !!value << WM8996_GP1_LVL_SHIFT); @@ -2155,7 +2151,7 @@ static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm8996_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); int val; val = (1 << WM8996_GP1_FN_SHIFT) | (!!value << WM8996_GP1_LVL_SHIFT); @@ -2167,7 +2163,7 @@ static int wm8996_gpio_direction_out(struct gpio_chip *chip, static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); unsigned int reg; int ret; @@ -2180,7 +2176,7 @@ static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm8996_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); return regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, WM8996_GP1_FN_MASK | WM8996_GP1_DIR, @@ -2211,7 +2207,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) else wm8996->gpio_chip.base = -1; - ret = gpiochip_add(&wm8996->gpio_chip); + ret = gpiochip_add_data(&wm8996->gpio_chip, wm8996); if (ret != 0) dev_err(wm8996->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 179/182] ASoC: wm8962: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm8962.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b563d6746ac4..1577d2e62e76 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -18,7 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> -#include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/pm_runtime.h> @@ -3308,14 +3308,9 @@ static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) } #ifdef CONFIG_GPIOLIB -static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8962_priv, gpio_chip); -} - static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); /* The WM8962 GPIOs aren't linearly numbered. For simplicity * we export linear numbers and error out if the unsupported @@ -3338,7 +3333,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, @@ -3348,7 +3343,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm8962_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); struct snd_soc_codec *codec = wm8962->codec; int ret, val; @@ -3387,7 +3382,7 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) else wm8962->gpio_chip.base = -1; - ret = gpiochip_add(&wm8962->gpio_chip); + ret = gpiochip_add_data(&wm8962->gpio_chip, wm8962); if (ret != 0) dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 178/182] ASoC: wm8903: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm8903.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 2512def0d349..d1315ce5f0a9 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -20,7 +20,7 @@ #include <linux/init.h> #include <linux/completion.h> #include <linux/delay.h> -#include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1766,11 +1766,6 @@ static int wm8903_resume(struct snd_soc_codec *codec) } #ifdef CONFIG_GPIOLIB -static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8903_priv, gpio_chip); -} - static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) { if (offset >= WM8903_NUM_GPIO) @@ -1781,7 +1776,7 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int mask, val; int ret; @@ -1799,7 +1794,7 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int reg; regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg); @@ -1810,7 +1805,7 @@ static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int mask, val; int ret; @@ -1828,7 +1823,7 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, WM8903_GP1_LVL_MASK, @@ -1860,7 +1855,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) else wm8903->gpio_chip.base = -1; - ret = gpiochip_add(&wm8903->gpio_chip); + ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903); if (ret != 0) dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 177/182] ASoC: wm5100: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/wm5100.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 171a23ddd15d..512a9d25fe6f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -17,6 +17,7 @@ #include <linux/export.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio/driver.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/pm_runtime.h> @@ -2236,14 +2237,9 @@ static irqreturn_t wm5100_edge_irq(int irq, void *data) } #ifdef CONFIG_GPIOLIB -static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) -{ - return container_of(chip, struct wm5100_priv, gpio_chip); -} - static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); @@ -2252,7 +2248,7 @@ static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm5100_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); int val, ret; val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); @@ -2268,7 +2264,7 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip, static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); unsigned int reg; int ret; @@ -2281,7 +2277,7 @@ static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); return regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, WM5100_GP1_FN_MASK | WM5100_GP1_DIR, @@ -2313,7 +2309,7 @@ static void wm5100_init_gpio(struct i2c_client *i2c) else wm5100->gpio_chip.base = -1; - ret = gpiochip_add(&wm5100->gpio_chip); + ret = gpiochip_add_data(&wm5100->gpio_chip, wm5100); if (ret != 0) dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

3 2

[alsa-devel] [PATCH 176/182] ASoC: rt5677: use gpiochip data pointer
by Linus Walleij 12 Dec '15

12 Dec '15

This makes the driver use the data pointer added to the gpio_chip to store a pointer to the state container instead of relying on container_of(). 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: alsa-devel(a)alsa-project.org Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Liam/Mark: please ACK this so I can merge it into the GPIO tree. --- sound/soc/codecs/rt5677.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 1f590b5a6718..89dda8962c7c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4500,14 +4500,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_GPIOLIB -static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip) -{ - return container_of(chip, struct rt5677_priv, gpio_chip); -} - static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4528,7 +4523,7 @@ static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int rt5677_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4552,7 +4547,7 @@ static int rt5677_gpio_direction_out(struct gpio_chip *chip, static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); int value, ret; ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value); @@ -4564,7 +4559,7 @@ static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4618,7 +4613,7 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); struct regmap_irq_chip_data *data = rt5677->irq_data; int irq; @@ -4677,7 +4672,7 @@ static void rt5677_init_gpio(struct i2c_client *i2c) rt5677->gpio_chip.parent = &i2c->dev; rt5677->gpio_chip.base = -1; - ret = gpiochip_add(&rt5677->gpio_chip); + ret = gpiochip_add_data(&rt5677->gpio_chip, rt5677); if (ret != 0) dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } -- 2.4.3

2 1

[alsa-devel] [PATCH v2 00/13] ASoC: Intel: Skylake: Update to SKL driver
by Vinod Koul 12 Dec '15

12 Dec '15

This update bring in new APIs in core and usage of those in SKL driver. The series is broadly of three parts - first two are fixes on the driver - then we add DMA resume capability in HDA core and use it to save restore DMA link values and add DMA resume capability in driver - lastly, we add support for MISCBDCGE configuration in core and use it while resetting and also for phrase detection per HW recommendation This should be merged thru ASoC tree due to obvious dependencies. Takashi would need your ack on core patches changes in v2: - move msicbdge api to driver and as a consequence sqash into into first use - fix lkp warning Dharageswari.R (1): ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification Jayachandran B (2): ALSA: hdac: Increase timeout value for link power check ASoC: Intel: Skylake: fix reset controller sequencing Jeeja KP (8): ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting ALSA: hdac: Add support for hda DMA Resume capability ALSA: hdac: couple the hda DMA stream in cleanup ASoC: Intel: Skylake: enable interrupt as wake source in active suspend ASoC: Intel: Skylake: Add DMA resume position in Trigger resume/suspend ASoC: Intel: Skylake: Reconfigure Link stream on suspend/resume ASoC: Intel: Skylake: Add Resume capability in PCM info. Vinod Koul (2): ALSA: hdac: add snd_hdac_ext_bus_link_power_up_all ASoC: Intel: Skylake: manage link power in active suspend include/sound/hda_register.h | 9 +++++ include/sound/hdaudio_ext.h | 15 +++++++ sound/hda/ext/hdac_ext_controller.c | 29 ++++++++++++- sound/hda/ext/hdac_ext_stream.c | 72 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 2 + sound/soc/intel/skylake/skl-pcm.c | 56 ++++++++++++++++++++----- sound/soc/intel/skylake/skl-sst-cldma.c | 38 ++++++++++------- sound/soc/intel/skylake/skl-sst-ipc.c | 15 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 3 ++ sound/soc/intel/skylake/skl.c | 51 +++++++++++++++++++++-- sound/soc/intel/skylake/skl.h | 5 +++ 11 files changed, 265 insertions(+), 30 deletions(-) -- 1.9.1

1 13

Re: [alsa-devel] [PATCH 05/14] ALSA: hdac: Add MISCBDCGE support
by Vinod Koul 12 Dec '15

12 Dec '15

On Sat, Dec 12, 2015 at 05:14:48AM +0800, kbuild test robot wrote: > Hi Jayachandran, > > [auto build test WARNING on asoc/for-next] > [also build test WARNING on next-20151211] > [cannot apply to sound/for-next v4.4-rc4] > > url: https://github.com/0day-ci/linux/commits/Vinod-Koul/ASoC-Intel-Skylake-Upda… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: mn10300-allmodconfig (attached as .config) > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=mn10300 > > All warnings (new ones prefixed by >>): > > sound/hda/ext/hdac_ext_controller.c: In function 'update_pci_dword': > >> sound/hda/ext/hdac_ext_controller.c:317:7: warning: 'data' is used uninitialized in this function [-Wuninitialized] > data &= ~mask; Thanks, will fix it in v2 -- ~Vinod

1 0

[alsa-devel] [PATCH] ASoC: fsl_asrc: sound is wrong after suspend/resume
by Zidan Wang 12 Dec '15

12 Dec '15

The register ASRCFG is volatile, but some bits need to be recovered after suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_asrc.c | 6 ++++++ sound/soc/fsl/fsl_asrc.h | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 3e404ba..942e74bc 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -990,6 +990,9 @@ static int fsl_asrc_suspend(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + regmap_read(asrc_priv->regmap, REG_ASRCFG, + &asrc_priv->regcache_cfg); + regcache_cache_only(asrc_priv->regmap, true); regcache_mark_dirty(asrc_priv->regmap); @@ -1010,6 +1013,9 @@ static int fsl_asrc_resume(struct device *dev) regcache_cache_only(asrc_priv->regmap, false); regcache_sync(asrc_priv->regmap); + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + 0x1FFFC0, asrc_priv->regcache_cfg); + /* Restart enabled pairs */ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 68802cd..3695ca9 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -453,6 +453,8 @@ struct fsl_asrc { int asrc_rate; int asrc_width; + + u32 regcache_cfg; }; extern struct snd_soc_platform_driver fsl_asrc_platform; -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 12 Dec '15

12 Dec '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. This is a hardware bug/errata and reset is the only option. According to the reference manual, the software reset doesn't reset any control register but only internal hardware logics such as bit clock generator, status flags, and FIFO pointers. (Our purpose is just to reset the clock generator while the software reset is the only way to do that.) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. For asynchronous mode, TX will not be reset when RX is still running. In this case, i can't reproduce this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1) refine the commit log and comments 2) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. sound/soc/fsl/fsl_sai.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8832901..a55a7de 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -564,6 +564,17 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: -- 1.9.1

2 1

[alsa-devel] Available frames > buffer size
by Rob Duncan 11 Dec '15

11 Dec '15

Is it ever correct for snd_pcm_avail() to return a value that’s greater than that returned by snd_pcm_hw_params_get_buffer_size()? I’m trying to track down some unexpected underruns in my application and I notice that when they occur I also see that the available frames count is wrong. For example here are a couple of lines from my logging: 2015-12-08T23:17:23.118817-08:00 X ? 2080 + 2080 2015-12-08T23:17:23.119524-08:00 X = 2080 ? 4126 The first log line: here I asked the device how many frames are available and it responds with 2080; I accordingly write 2080 frames. The second log line: the writei() call returns 2080. I immediately ask how many frames are available now (I usually get a small number in response ~ 20-30) but this time I get an XRUN. I call snd_pcm_recover() and then snd_pcm_avail() again and now I get 4126 frames. As you can see the time between these two logs is less than 1 millisecond, so I don’t understand how the PCM can go from 2080 frames available to >4096 frames available so quickly. Is this evidence of a bug in the device driver? My application is using a single channel dshare PCM (rt) that has an 8-channel device underneath. Here’s the configuration fragment: pcm_slave.cid_dsp { pcm "hw:0,0" channels 8 rate 44100 } pcm.rt_channel { type dshare ipc_key 1025 ipc_gid audio ipc_perm 0660 slave cid_dsp bindings { 0 5 } } pcm.rt { type plug slave.pcm rt_channel } The underlying hardware looks like this: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 8 rate: 44100 (44100/1) period_size: 1024 buffer_size: 4096 tstamp_mode: ENABLE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 1073741824 silence_threshold: 0 silence_size: 0 boundary: 1073741824 Thanks for any insight and help, Rob.

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: constify usb_protocol_ops structures
by Julia Lawall 11 Dec '15

11 Dec '15

The usb_protocol_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/usb/midi.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index ee212e7..cc39f63 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -112,7 +112,7 @@ struct snd_usb_midi { struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; - struct usb_protocol_ops *usb_protocol_ops; + const struct usb_protocol_ops *usb_protocol_ops; struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; @@ -671,31 +671,32 @@ static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep, } } -static struct usb_protocol_ops snd_usbmidi_standard_ops = { +static const struct usb_protocol_ops snd_usbmidi_standard_ops = { .input = snd_usbmidi_standard_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_midiman_ops = { +static const struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; -static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { +static const +struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_cme_ops = { +static const struct usb_protocol_ops snd_usbmidi_cme_ops = { .input = snd_usbmidi_cme_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { +static const struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { .input = ch345_broken_sysex_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, @@ -795,7 +796,7 @@ static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, } } -static struct usb_protocol_ops snd_usbmidi_akai_ops = { +static const struct usb_protocol_ops snd_usbmidi_akai_ops = { .input = snd_usbmidi_akai_input, .output = snd_usbmidi_akai_output, }; @@ -835,7 +836,7 @@ static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = 2 + count; } -static struct usb_protocol_ops snd_usbmidi_novation_ops = { +static const struct usb_protocol_ops snd_usbmidi_novation_ops = { .input = snd_usbmidi_novation_input, .output = snd_usbmidi_novation_output, }; @@ -867,7 +868,7 @@ static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = count; } -static struct usb_protocol_ops snd_usbmidi_raw_ops = { +static const struct usb_protocol_ops snd_usbmidi_raw_ops = { .input = snd_usbmidi_raw_input, .output = snd_usbmidi_raw_output, }; @@ -883,7 +884,7 @@ static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint *ep, snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2); } -static struct usb_protocol_ops snd_usbmidi_ftdi_ops = { +static const struct usb_protocol_ops snd_usbmidi_ftdi_ops = { .input = snd_usbmidi_ftdi_input, .output = snd_usbmidi_raw_output, }; @@ -927,7 +928,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = ep->max_transfer; } -static struct usb_protocol_ops snd_usbmidi_122l_ops = { +static const struct usb_protocol_ops snd_usbmidi_122l_ops = { .input = snd_usbmidi_us122l_input, .output = snd_usbmidi_us122l_output, }; @@ -1060,7 +1061,7 @@ static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = ep->max_transfer - buf_free; } -static struct usb_protocol_ops snd_usbmidi_emagic_ops = { +static const struct usb_protocol_ops snd_usbmidi_emagic_ops = { .input = snd_usbmidi_emagic_input, .output = snd_usbmidi_emagic_output, .init_out_endpoint = snd_usbmidi_emagic_init_out,

3 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx
by Peter Ujfalusi 11 Dec '15

11 Dec '15

The condition for checking for XDAT being cleared was not correct. Fixes: 36bcecd0a73eb ("ASoC: davinci-mcasp: Correct TX start sequence") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c1c9c2e3525b..2ccb8bccc9d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -223,8 +223,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for XDATA to be cleared */ cnt = 0; - while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & - ~XRDATA) && (cnt < 100000)) + while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) && + (cnt < 100000)) cnt++; /* Release TX state machine */ -- 2.6.4

1 0

Re: [alsa-devel] [Intel-gfx] [PATCH V4 1/2] dp/mst: add SDP stream support
by Libin Yang 11 Dec '15

11 Dec '15

Add Takashi and ALSA mailing list Best Regards, Libin On 12/02/2015 02:09 PM, libin.yang(a)linux.intel.com wrote: > From: Libin Yang <libin.yang(a)linux.intel.com> > > This adds code to initialise the SDP streams > for a sink in the simplest ordering. > > I've no idea how you'd want to control the > ordering at this level, so don't bother > until someone comes up with a use case. > > Reviewed-by: Ander Conselvan de Oliveira <conselvan2(a)gmail.com> > Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> > Signed-off-by: Dave Airlie <airlied(a)redhat.com> > --- > drivers/gpu/drm/drm_dp_mst_topology.c | 38 ++++++++++++++++++++++++++++++++--- > include/drm/drm_dp_mst_helper.h | 7 +++++-- > 2 files changed, 40 insertions(+), 5 deletions(-) > > diff --git a/drivers/gpu/drm/drm_dp_mst_topology.c b/drivers/gpu/drm/drm_dp_mst_topology.c > index 809959d..f50eb7b 100644 > --- a/drivers/gpu/drm/drm_dp_mst_topology.c > +++ b/drivers/gpu/drm/drm_dp_mst_topology.c > @@ -666,7 +666,9 @@ static int build_enum_path_resources(struct drm_dp_sideband_msg_tx *msg, int por > } > > static int build_allocate_payload(struct drm_dp_sideband_msg_tx *msg, int port_num, > - u8 vcpi, uint16_t pbn) > + u8 vcpi, uint16_t pbn, > + u8 number_sdp_streams, > + u8 *sdp_stream_sink) > { > struct drm_dp_sideband_msg_req_body req; > memset(&req, 0, sizeof(req)); > @@ -674,6 +676,9 @@ static int build_allocate_payload(struct drm_dp_sideband_msg_tx *msg, int port_n > req.u.allocate_payload.port_number = port_num; > req.u.allocate_payload.vcpi = vcpi; > req.u.allocate_payload.pbn = pbn; > + req.u.allocate_payload.number_sdp_streams = number_sdp_streams; > + memcpy(req.u.allocate_payload.sdp_stream_sink, sdp_stream_sink, > + number_sdp_streams); > drm_dp_encode_sideband_req(&req, msg); > msg->path_msg = true; > return 0; > @@ -1562,6 +1567,8 @@ static int drm_dp_payload_send_msg(struct drm_dp_mst_topology_mgr *mgr, > struct drm_dp_sideband_msg_tx *txmsg; > struct drm_dp_mst_branch *mstb; > int len, ret; > + u8 sinks[DRM_DP_MAX_SDP_STREAMS]; > + int i; > > mstb = drm_dp_get_validated_mstb_ref(mgr, port->parent); > if (!mstb) > @@ -1573,10 +1580,13 @@ static int drm_dp_payload_send_msg(struct drm_dp_mst_topology_mgr *mgr, > goto fail_put; > } > > + for (i = 0; i < port->num_sdp_streams; i++) > + sinks[i] = i; > + > txmsg->dst = mstb; > len = build_allocate_payload(txmsg, port->port_num, > id, > - pbn); > + pbn, port->num_sdp_streams, sinks); > > drm_dp_queue_down_tx(mgr, txmsg); > > @@ -2259,6 +2269,27 @@ out: > EXPORT_SYMBOL(drm_dp_mst_detect_port); > > /** > + * drm_dp_mst_port_has_audio() - Check whether port has audio capability or not > + * @mgr: manager for this port > + * @port: unverified pointer to a port. > + * > + * This returns whether the port supports audio or not. > + */ > +bool drm_dp_mst_port_has_audio(struct drm_dp_mst_topology_mgr *mgr, > + struct drm_dp_mst_port *port) > +{ > + bool ret = false; > + > + port = drm_dp_get_validated_port_ref(mgr, port); > + if (!port) > + return ret; > + ret = port->has_audio; > + drm_dp_put_port(port); > + return ret; > +} > +EXPORT_SYMBOL(drm_dp_mst_port_has_audio); > + > +/** > * drm_dp_mst_get_edid() - get EDID for an MST port > * @connector: toplevel connector to get EDID for > * @mgr: manager for this port > @@ -2283,6 +2314,7 @@ struct edid *drm_dp_mst_get_edid(struct drm_connector *connector, struct drm_dp_ > edid = drm_get_edid(connector, &port->aux.ddc); > drm_mode_connector_set_tile_property(connector); > } > + port->has_audio = drm_detect_monitor_audio(edid); > drm_dp_put_port(port); > return edid; > } > @@ -2566,7 +2598,7 @@ static void drm_dp_mst_dump_mstb(struct seq_file *m, > > seq_printf(m, "%smst: %p, %d\n", prefix, mstb, mstb->num_ports); > list_for_each_entry(port, &mstb->ports, next) { > - seq_printf(m, "%sport: %d: ddps: %d ldps: %d, %p, conn: %p\n", prefix, port->port_num, port->ddps, port->ldps, port, port->connector); > + seq_printf(m, "%sport: %d: ddps: %d ldps: %d, sdp: %d/%d, %p, conn: %p\n", prefix, port->port_num, port->ddps, port->ldps, port->num_sdp_streams, port->num_sdp_stream_sinks, port, port->connector); > if (port->mstb) > drm_dp_mst_dump_mstb(m, port->mstb); > } > diff --git a/include/drm/drm_dp_mst_helper.h b/include/drm/drm_dp_mst_helper.h > index 5340099..74b5888 100644 > --- a/include/drm/drm_dp_mst_helper.h > +++ b/include/drm/drm_dp_mst_helper.h > @@ -94,6 +94,7 @@ struct drm_dp_mst_port { > struct drm_dp_mst_topology_mgr *mgr; > > struct edid *cached_edid; /* for DP logical ports - make tiling work */ > + bool has_audio; > }; > > /** > @@ -215,13 +216,13 @@ struct drm_dp_sideband_msg_rx { > struct drm_dp_sideband_msg_hdr initial_hdr; > }; > > - > +#define DRM_DP_MAX_SDP_STREAMS 16 > struct drm_dp_allocate_payload { > u8 port_number; > u8 number_sdp_streams; > u8 vcpi; > u16 pbn; > - u8 sdp_stream_sink[8]; > + u8 sdp_stream_sink[DRM_DP_MAX_SDP_STREAMS]; > }; > > struct drm_dp_allocate_payload_ack_reply { > @@ -484,6 +485,8 @@ int drm_dp_mst_hpd_irq(struct drm_dp_mst_topology_mgr *mgr, u8 *esi, bool *handl > > enum drm_connector_status drm_dp_mst_detect_port(struct drm_connector *connector, struct drm_dp_mst_topology_mgr *mgr, struct drm_dp_mst_port *port); > > +bool drm_dp_mst_port_has_audio(struct drm_dp_mst_topology_mgr *mgr, > + struct drm_dp_mst_port *port); > struct edid *drm_dp_mst_get_edid(struct drm_connector *connector, struct drm_dp_mst_topology_mgr *mgr, struct drm_dp_mst_port *port); > > >

1 0

[alsa-devel] [PATCH v5] sun4i-codec: add inputs
by Danny Milosavljevic 11 Dec '15

11 Dec '15

Hi, this is the fifth version of the patch that adds inputs to sun4i-codec. (Previous versions have been on the linux-sunxi mailing list only) The inputs added are: - FM-In Left and Right - Line-In Left and Right - Mic1-In - Mic2-In Changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. I successfully tested it on an A20 board using alsamixer, headphones, a radio and my ears. Note that because of missing capturing support I tested only the mixing, for Mic, Line, and FM. The patches are on top of <git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git>, branch "sunxi/for-next". Regards, Danny Danny (1): b/sound/soc/sunxi/sun4i-codec.c | 153 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 150 insertions(+), 3 deletions(-) Signed-off-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> --- diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..106b568 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -57,9 +57,20 @@ #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) #define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LNG (26) +#define SUN4I_CODEC_DAC_ACTL_FMG (23) +#define SUN4I_CODEC_DAC_ACTL_MICG (20) +#define SUN4I_CODEC_DAC_ACTL_LLNS (19) +#define SUN4I_CODEC_DAC_ACTL_RLNS (18) +#define SUN4I_CODEC_DAC_ACTL_LFMS (17) +#define SUN4I_CODEC_DAC_ACTL_RFMS (16) #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -84,8 +95,11 @@ #define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) #define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) #define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_PREG1_A10 (25) +#define SUN4I_CODEC_ADC_ACTL_PREG2_A10 (23) #define SUN4I_CODEC_ADC_ACTL_VADCG (20) #define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_LNRDF (16) #define SUN4I_CODEC_ADC_ACTL_PA_EN (4) #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) @@ -94,7 +108,18 @@ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) #define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +/* MIC_PHONE_CAL register offsets and bit fields (A20 only) */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1 (29) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2 (26) +/* note: no idea where the output pins for the following are. */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTG (5) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTEN (4) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS3 (3) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS2 (2) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS1 (1) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS0 (0) struct sun4i_codec { struct device *dev; @@ -402,16 +427,94 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(sun4i_codec_linein_loopback_gain_scale, -150, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_fmin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_micin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_RANGE(sun7i_codec_micin_preamp_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0) +); +static DECLARE_TLV_DB_RANGE(sun4i_codec_micin_preamp_gain_scale_a10, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(3500, 300, 0) +); + +struct snd_kcontrol_new sun7ionly_codec_widgets[] = { + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), +}; -static const struct snd_kcontrol_new sun4i_codec_widgets[] = { +/* Note: this is modified by the sun4i_codec_probe() function. */ +static struct snd_kcontrol_new sun4i_codec_widgets[] = { SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), + /* Line-In, FM-In, Mic1-In, Mic2-In */ + SOC_SINGLE_TLV("Line-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LNG, + 1, + 0, + sun4i_codec_linein_loopback_gain_scale), + SOC_SINGLE_TLV("FM-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_FMG, + 3, + 0, + sun4i_codec_fmin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MICG, + 7, + 0, + sun4i_codec_micin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic1-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1EN, + 1, + 0, + NULL), + SOC_SINGLE_TLV("Mic2-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2EN, + 1, + 0, + NULL), }; static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), + SOC_DAPM_SINGLE("Left Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LLNS, 1, 0), + SOC_DAPM_SINGLE("Left FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { @@ -419,6 +522,14 @@ static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Right Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RLNS, 1, 0), + SOC_DAPM_SINGLE("Right FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -462,8 +573,16 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), + + SND_SOC_DAPM_INPUT("Line-In Right"), + SND_SOC_DAPM_INPUT("Line-In Left"), + SND_SOC_DAPM_INPUT("FM-In Right"), + SND_SOC_DAPM_INPUT("FM-In Left"), + SND_SOC_DAPM_INPUT("Mic1-In"), + SND_SOC_DAPM_INPUT("Mic2-In"), }; +/* {sink, control, source} */ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { /* Left DAC Routes */ { "Left DAC", NULL, "DAC" }, @@ -490,6 +609,20 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Line-In, FM-In */ + { "Right Mixer", "Right Line-In Playback Switch", "Line-In Right" }, + { "Left Mixer", "Left Line-In Playback Switch", "Line-In Left" }, + { "Right Mixer", "Right FM-In Playback Switch", "FM-In Right" }, + { "Left Mixer", "Left FM-In Playback Switch", "FM-In Left" }, + + /* Mic1-In */ + { "Right Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + { "Left Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + + /* Mic2-In */ + { "Right Mixer", "Mic2-In Playback Switch", "Mic2-In" }, + { "Left Mixer", "Mic2-In Playback Switch", "Mic2-In" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { @@ -537,7 +670,7 @@ static const struct regmap_config sun4i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, }; static const struct of_device_id sun4i_codec_of_match[] = { @@ -594,6 +727,20 @@ static int sun4i_codec_probe(struct platform_device *pdev) void __iomem *base; int ret; + if (of_device_is_compatible(pdev->dev.of_node, "allwinner,sun7i-a20-codec")) { + size_t i, j; + for (i = 0; i < ARRAY_SIZE(sun7ionly_codec_widgets); ++i) { + struct snd_kcontrol_new *new_kcontrol; + new_kcontrol = &sun7ionly_codec_widgets[i]; + for (j = 0; j < ARRAY_SIZE(sun4i_codec_widgets); ++j) { + struct snd_kcontrol_new *kcontrol; + kcontrol = &sun4i_codec_widgets[j]; + if (strcmp(kcontrol->name, new_kcontrol->name) == 0) + memcpy(kcontrol, new_kcontrol, sizeof(*kcontrol)); + } + } + } + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); if (!scodec) return -ENOMEM;

3 3

[alsa-devel] HDMI Repoll after pm?
by David Henningsson 10 Dec '15

10 Dec '15

Hi Takashi, A recent change in sound.git seems to have introduced some repoll after pm. Is this an intentional change? I don't think it's wrong per se (it stops after 6 repolls), but it breaks my hda-emu tester when "list" does not return the expected input. To reproduce, try running: ./hda-emu codecs/canonical/92hd91bxx-dell-inspiron-5423-ccert-201208-11456 -i 1 > pm .... lots of data .... ** TEST_PM DONE ** HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > list 1: HDMI/DP Jack:0 (1) 2: IEC958 Playback Con Mask:0 (1) 3: IEC958 Playback Pro Mask:0 (1) 4: IEC958 Playback Default:0 (1) 5: IEC958 Playback Switch:0 (1) 6: ELD:0 (1) 7: Playback Channel Map:0 (1) HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > ...notice how the last two lines are somewhat unexpected output for a "list" command, but I can make my tool more robust in case you confirm this is an expected change. // David

2 5

[alsa-devel] [PATCH 1/2] ALSA: hda - Optimize audio component check in patch_hdmi.c
by Takashi Iwai 10 Dec '15

10 Dec '15

The audio component is enabled only when CONFIG_SND_HDA_I915 is set. Give a dummy macro for allowing the compiler optimize out the relevant codes when this Kconfig isn't set. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85342d261043..44d0d2374bb6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -152,8 +152,12 @@ struct hdmi_spec { struct i915_audio_component_audio_ops i915_audio_ops; }; +#ifdef CONFIG_SND_HDA_I915 #define codec_has_acomp(codec) \ ((codec)->bus->core.audio_component != NULL) +#else +#define codec_has_acomp(codec) false +#endif struct hdmi_audio_infoframe { u8 type; /* 0x84 */ -- 2.6.3

1 1

[alsa-devel] [RFC][PATCH 0/3] ALSA: fireface: new driver for RME Fireface series (MIDI only)
by Takashi Sakamoto 10 Dec '15

10 Dec '15

Hi, This patchset is to add a new driver for RME Fireface series. This is still working in progress but currently the new driver support MIDI functionality. Unfortunately, ffado library can disturb this functionality. In RME::Device::init_hardware() function, the library sends a write transaction to 0x0000801003f4 with invalid value as higher part of address in IEEE 1212 or ISO/IEC 13213. This is a worst case I describe in patch 03. A workaround is to load this module after running ffado library. This module sends write transactions with valid values and regain MIDI functionality. I think it better that FFADO developers fixes the bug as long as they doesn't support MIDI functionality. Takashi Sakamoto (3): fireface: add skeleton for RME Fireface series fireface: add transaction support fireface: add support for MIDI functionality sound/firewire/Kconfig | 7 + sound/firewire/Makefile | 1 + sound/firewire/fireface/Makefile | 2 + sound/firewire/fireface/fireface-midi.c | 129 ++++++++ sound/firewire/fireface/fireface-transaction.c | 388 +++++++++++++++++++++++++ sound/firewire/fireface/fireface.c | 151 ++++++++++ sound/firewire/fireface/fireface.h | 73 +++++ 7 files changed, 751 insertions(+) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/fireface-midi.c create mode 100644 sound/firewire/fireface/fireface-transaction.c create mode 100644 sound/firewire/fireface/fireface.c create mode 100644 sound/firewire/fireface/fireface.h -- 2.5.0

4 23

[alsa-devel] A root cause of "unable to create IPC shm instance"
by IceBsi 10 Dec '15

10 Dec '15

Dear ALSA developers: I found a root cause of the following issue by using strace: unable to create IPC shm instance Firstly, please see the test code below: 1 #include <stdio.h> 2 #include <unistd.h> 3 #include <sys/ipc.h> 4 #include <sys/shm.h> 5 #include <errno.h> 6 #include <string.h> 7 8 int main(int argc, char *argv[]) 9 { 10 int r = 0; 11 struct shmid_ds buf; 12 key_t ipc_key = (key_t)325559; 13 size_t shmsize = 488; 14 int shmid = -1; 15 mode_t ipc_perm = 0666; 16 void *shmptr = NULL; 17 pid_t pid; 18 19 printf("___%d:%d call shmget(%d,%u,%d)\n", __LINE__, getpid(), (int)(ipc_key), shmsize, IPC_CREAT | ipc_perm); 20 shmid = shmget(ipc_key, shmsize, IPC_CREAT | ipc_perm); 21 printf("___%d:%d ret shmget=%d\n", __LINE__, getpid(), shmid); 22 23 printf("___%d:%d call shmat(%d,0,0)\n", __LINE__, getpid(), shmid); 24 shmptr = shmat(shmid, 0, 0); 25 printf("___%d:%d ret shmat=%p\n", __LINE__, getpid(), shmptr); 26 27 printf("+++%d:%d call fork\n", __LINE__, getpid()); 28 pid = fork(); 29 printf("+++%d:%d ret fork=%d\n", __LINE__, getpid(), (int)pid); 30 31 printf("___%d:%d call shmctl(%d,IPC_STAT,%p)\n", __LINE__, getpid(), shmid, &buf); 32 r = shmctl(shmid, IPC_STAT, &buf); 33 printf("___%d:%d ret shmctl=%d\n", __LINE__, getpid(), r); 34 printf("___%d:%d nattch=%lu\n", __LINE__, getpid(), buf.shm_nattch); 35 36 printf("___%d:%d call shmdt(%p)\n", __LINE__, getpid(), shmptr); 37 r = shmdt(shmptr); 38 printf("___%d:%d ret shmdt=%d\n", __LINE__, getpid(), r); 39 40 printf("___%d:%d call shmctl(%d,IPC_RMID,NULL)\n", __LINE__, getpid(), shmid); 41 r = shmctl(shmid, IPC_RMID, NULL); 42 printf("___%d:%d ret shmctl=%d, %s\n", __LINE__, getpid(), r, strerror(errno)); 43 44 if ((pid_t)0 != pid) { 45 printf("+++%d:%d call wait(NULL)\n", __LINE__, getpid()); 46 pid = wait(NULL); 47 printf("+++%d:%d ret wait=%d\n", __LINE__, getpid(), (int)pid); 48 } 49 50 return 0; 51 } The output of the test code is: ___19:11782 call shmget(325559,488,950) ___21:11782 ret shmget=2850820 ___23:11782 call shmat(2850820,0,0) ___25:11782 ret shmat=0x7f452c223000 +++27:11782 call fork +++29:11782 ret fork=11783 ___31:11782 call shmctl(2850820,IPC_STAT,0x7fffdaf0a750) ___33:11782 ret shmctl=0 ___34:11782 nattch=2 ___36:11782 call shmdt(0x7f452c223000) +++29:11783 ret fork=0 ___38:11782 ret shmdt=0 ___40:11782 call shmctl(2850820,IPC_RMID,NULL) ___31:11783 call shmctl(2850820,IPC_STAT,0x7fffdaf0a750) ___42:11782 ret shmctl=0, Success +++45:11782 call wait(NULL) ___33:11783 ret shmctl=0 ___34:11783 nattch=1 ___36:11783 call shmdt(0x7f452c223000) ___38:11783 ret shmdt=0 ___40:11783 call shmctl(2850820,IPC_RMID,NULL) ___42:11783 ret shmctl=-1, Invalid argument +++47:11782 ret wait=11783 And sometimes output is: ___19:11798 call shmget(325559,488,950) ___21:11798 ret shmget=3112964 ___23:11798 call shmat(3112964,0,0) ___25:11798 ret shmat=0x7f6b39378000 +++27:11798 call fork +++29:11798 ret fork=11799 ___31:11798 call shmctl(3112964,IPC_STAT,0x7fff3a6472a0) ___33:11798 ret shmctl=0 +++29:11799 ret fork=0 ___31:11799 call shmctl(3112964,IPC_STAT,0x7fff3a6472a0) ___34:11798 nattch=2 ___36:11798 call shmdt(0x7f6b39378000) ___33:11799 ret shmctl=0 ___38:11798 ret shmdt=0 ___34:11799 nattch=2 ___40:11798 call shmctl(3112964,IPC_RMID,NULL) ___36:11799 call shmdt(0x7f6b39378000) ___42:11798 ret shmctl=0, Success +++45:11798 call wait(NULL) ___38:11799 ret shmdt=0 ___40:11799 call shmctl(3112964,IPC_RMID,NULL) ___42:11799 ret shmctl=-1, Invalid argument +++47:11798 ret wait=11799 >From the outputs the parent process always get nattch=2, the child process get nattch=1 or sometimes get nattch=2. Then please see the following lines of src/pcm/pcm_direct.c (from alsa-lib-1.1.0.tar.bz2): 110 dmix->shmptr = shmat(dmix->shmid, 0, 0); 111 if (dmix->shmptr == (void *) -1) { 112 err = -errno; 113 snd_pcm_direct_shm_discard(dmix); 114 return err; 115 } 116 mlock(dmix->shmptr, sizeof(snd_pcm_direct_share_t)); 117 if (shmctl(dmix->shmid, IPC_STAT, &buf) < 0) { 118 err = -errno; 119 snd_pcm_direct_shm_discard(dmix); 120 return err; 121 } 122 if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ 123 memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); 124 if (dmix->ipc_gid>= 0) { 125 buf.shm_perm.gid = dmix->ipc_gid; 126 shmctl(dmix->shmid, IPC_SET, &buf); 127 } 128 dmix->shmptr->magic = SND_PCM_DIRECT_MAGIC; 129 return 1; 130 } else { 131 if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { 132 snd_pcm_direct_shm_discard(dmix); 133 return -EINVAL; 134 } 135 } If an applicatioin that uses alsa-lib invokes fork() and the fork() is invoked just after shmat(line 110) and just before shmctl(line 117), buf.shm_nattch will be 2, so "buf.shm_nattch == 1" will be false. But while buf.shm_nattch is 2, we're still the first user this time, and we should executes line 123 to line 129 of the codes above. There is the following statement in `man shmget`: When a new shared memory segment is created, its contents are initialized to zero values According to the statement above, dmix->shmptr->magic is 0 on shm first created. So, to fix this issue, we should change the following line: 122 if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ to: 122 if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { /* we're the first user, clear the segment */ Sincerely, Sisrnb

2 1

[alsa-devel] [PATCH v3 0/4] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 10 Dec '15

10 Dec '15

Hi, this is the third revision of the patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. A few patches have been dropped from this patchset as they were already applied individually beforehand. Also, two cleanup patches for i915 were squashed into one patch. The current patchset is found in sound git tree test/hdmi-jack branch. thanks, Takashi === v2->v3: * Track drm_connector for easier ELD retrieval, remove superfluous audio_enabled flag instead * Back to av_mutex * Adapt the earlier applied preliminary patches * Direct invocation of get_eld now instead of deferred call, as we have no longer deadlocks * Rebase to drm-next, update get_eld documentation for new kernel doc v1->v2: * Use modeset lock for get_eld lock, drop av mutex * Return the expected size from get_eld, not the copied size * Add reverse map from a port number to the encoder * Deferred invocation of get_eld from the eld_notify in HDA side * A bit more code refactoring in HDA side * Move audio component accessors to hdac_i915.c Takashi Iwai (4): drm/i915: Add get_eld audio component drm/i915: Add reverse mapping between port and intel_encoder ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Move audio component accesses to hdac_i915.c drivers/gpu/drm/i915/i915_drv.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 59 +++++++++++++++------ drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 1 + drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_hdmi.c | 2 + include/drm/i915_component.h | 14 +++++ include/sound/hda_i915.h | 14 +++++ sound/hda/hdac_i915.c | 66 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 104 ++++++++++++++++++++++++++++++------- 10 files changed, 229 insertions(+), 36 deletions(-) -- 2.6.3

4 14

[alsa-devel] using all channels of RME Babyface Pro in class compliant mode
by Be 10 Dec '15

10 Dec '15

Hello, I just got an RME Babyface Pro USB sound card. It can operate either as a USB audio class compliant device (with mixing controlled by the device's physical controls) or make use of a proprietary driver on Windows and Mac OS X. The proprietary driver allows for control of the mixing capabilities of the device and using onboard DSP EQ. It would be great if someone wrote a Linux driver to make use of these. I'd be happy to send debugging info or whatever about the device and test kernel patches. I could set up a Windows VM to intercept USB communication. In the meantime, I am trying to use its class compliant mode to use with Mixxx for DJing. However, I cannot figure out how to use channels 3-4 for my headphones. The device has 4 analog channels out and in plus 8 digital out and in (via ADAT). Its manual says, regarding class compliant mode, "in Mac OS X, either the first two or all channels will work, depending on the choice in the Audio MIDI Setup." I believe the two channel operation exists so when the Babyface Pro is plugged into an iPad, the iPad sends stereo output from iTunes and other programs to the Babyface Pro's main output. However, I do not know how to select between this 2 channel mode and the 12 channel mode with Linux. PulseAudio only shows it as a stereo device. Mixxx, which uses the PortAudio library, displays all 12 channels as available but doesn't work when any channels other than 1-2 are selected. Here is debugging output of Mixxx: Debug [Main]: SoundManager::setupDevices() Debug [Main]: SoundDevicePortAudio::open() "2, Babyface Pro (70785713): USB Audio (hw:1,0)" Debug [Main]: framesPerBuffer: 256 Debug [Main]: Requested sample rate: 44100 Hz, latency: 5.80499 ms Debug [Main]: Output channels: 4 | Input channels: 0 Debug [Main]: Opening stream with id 2 Expression 'alsa_snd_pcm_hw_params_set_channels( pcm, hwParams, self->numHostChannels )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2051 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838 Warning [Main]: Error opening stream: Invalid number of channels aplay -l output: **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC272 Analog [ALC272 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Pro70785713 [Babyface Pro (70785713)], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L output: null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default ALSA Output (currently PulseAudio Sound Server) sysdefault:CARD=PCH HDA Intel PCH, ALC272 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output sysdefault:CARD=Pro70785713 Babyface Pro (70785713), USB Audio Default Audio Device front:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio Front speakers surround21:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio IEC958 (S/PDIF) Digital Audio Output /proc/asound/card1/stream0 when playing to channels 1-2 with PulseAudio: RME Babyface Pro (70785713) at usb-0000:00:1d.0-1.2, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 47 Momentary freq = 44118 Hz (0x5.83c9) Feedback Format = 16.16 Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Interface 1 Altset 2 Format: S24_3LE Channels: 12 Endpoint: 3 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Capture: Status: Stop Interface 2 Altset 1 Format: S24_3LE Channels: 12 Endpoint: 4 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Interface 2 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 4 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us So I guess the question is how to get PortAudio to use the "Altset 2" option. Could this be accomplished with some asoundrc sorcery? Does something need to change in snd-usb-audio? Is the issue with PortAudio and PulseAudio? How can I test channels 3-4 with aplay or another program?

2 2

[alsa-devel] [PATCH] ALSA: hda - Implement loopback control switch for Realtek and other codecs
by Takashi Iwai 10 Dec '15

10 Dec '15

Many codecs, typically found on Realtek codecs, have the analog loopback path merged to the secondary input of the middle of the output paths. Currently, we don't offer the dynamic switching in such configuration but let each loopback path mute by itself. This should work well in theory, but in reality, we often see that such a dead loopback path causes some background noises even if all the elements get muted. Such a problem has been fixed by adding the quirk accordingly to disable aamix, and it's the right fix, per se. The only problem is that it's not so trivial to achieve it; user needs to pass a hint string via patch module option or sysfs. This patch gives a bit improvement on the situation: it adds "Loopback Mixing" control element for such codecs like other codecs (e.g. IDT or VIA codecs) with the individual loopback paths. User can turn on/off the loopback path simply via a mixer app. For keeping the compatibility, the loopback is still enabled on these codecs. But user can try to turn it off if experiencing a suspicious background or click noise on the fly, then build a static fixup later once after the problem is addressed. Other than the addition of the loopback enable/disablement control, there should be no changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 87 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 68 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f3c058f6c831..30c8efe0f80a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -754,9 +754,6 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps; unsigned int mask, val; - if (!enable && is_active_nid(codec, nid, dir, idx_to_check)) - return; - caps = query_amp_caps(codec, nid, dir); val = get_amp_val_to_activate(codec, nid, dir, caps, enable); mask = get_amp_mask_to_modify(codec, nid, dir, idx_to_check, caps); @@ -767,12 +764,22 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, update_amp(codec, nid, dir, idx, mask, val); } +static void check_and_activate_amp(struct hda_codec *codec, hda_nid_t nid, + int dir, int idx, int idx_to_check, + bool enable) +{ + /* check whether the given amp is still used by others */ + if (!enable && is_active_nid(codec, nid, dir, idx_to_check)) + return; + activate_amp(codec, nid, dir, idx, idx_to_check, enable); +} + static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, int i, bool enable) { hda_nid_t nid = path->path[i]; init_amp(codec, nid, HDA_OUTPUT, 0); - activate_amp(codec, nid, HDA_OUTPUT, 0, 0, enable); + check_and_activate_amp(codec, nid, HDA_OUTPUT, 0, 0, enable); } static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, @@ -800,9 +807,16 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, * when aa-mixer is available, we need to enable the path as well */ for (n = 0; n < nums; n++) { - if (n != idx && (!add_aamix || conn[n] != spec->mixer_merge_nid)) - continue; - activate_amp(codec, nid, HDA_INPUT, n, idx, enable); + if (n != idx) { + if (conn[n] != spec->mixer_merge_nid) + continue; + /* when aamix is disabled, force to off */ + if (!add_aamix) { + activate_amp(codec, nid, HDA_INPUT, n, n, false); + continue; + } + } + check_and_activate_amp(codec, nid, HDA_INPUT, n, idx, enable); } } @@ -1563,6 +1577,12 @@ static bool map_singles(struct hda_codec *codec, int outs, return found; } +static inline bool has_aamix_out_paths(struct hda_gen_spec *spec) +{ + return spec->aamix_out_paths[0] || spec->aamix_out_paths[1] || + spec->aamix_out_paths[2]; +} + /* create a new path including aamix if available, and return its index */ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) { @@ -2405,25 +2425,51 @@ static void update_aamix_paths(struct hda_codec *codec, bool do_mix, } } +/* re-initialize the output paths; only called from loopback_mixing_put() */ +static void update_output_paths(struct hda_codec *codec, int num_outs, + const int *paths) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + int i; + + for (i = 0; i < num_outs; i++) { + path = snd_hda_get_path_from_idx(codec, paths[i]); + if (path) + snd_hda_activate_path(codec, path, path->active, + spec->aamix_mode); + } +} + static int loopback_mixing_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int val = ucontrol->value.enumerated.item[0]; if (val == spec->aamix_mode) return 0; spec->aamix_mode = val; - update_aamix_paths(codec, val, spec->out_paths[0], - spec->aamix_out_paths[0], - spec->autocfg.line_out_type); - update_aamix_paths(codec, val, spec->hp_paths[0], - spec->aamix_out_paths[1], - AUTO_PIN_HP_OUT); - update_aamix_paths(codec, val, spec->speaker_paths[0], - spec->aamix_out_paths[2], - AUTO_PIN_SPEAKER_OUT); + if (has_aamix_out_paths(spec)) { + update_aamix_paths(codec, val, spec->out_paths[0], + spec->aamix_out_paths[0], + cfg->line_out_type); + update_aamix_paths(codec, val, spec->hp_paths[0], + spec->aamix_out_paths[1], + AUTO_PIN_HP_OUT); + update_aamix_paths(codec, val, spec->speaker_paths[0], + spec->aamix_out_paths[2], + AUTO_PIN_SPEAKER_OUT); + } else { + update_output_paths(codec, cfg->line_outs, spec->out_paths); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + update_output_paths(codec, cfg->hp_outs, spec->hp_paths); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + update_output_paths(codec, cfg->speaker_outs, + spec->speaker_paths); + } return 1; } @@ -2441,12 +2487,13 @@ static int create_loopback_mixing_ctl(struct hda_codec *codec) if (!spec->mixer_nid) return 0; - if (!(spec->aamix_out_paths[0] || spec->aamix_out_paths[1] || - spec->aamix_out_paths[2])) - return 0; if (!snd_hda_gen_add_kctl(spec, NULL, &loopback_mixing_enum)) return -ENOMEM; spec->have_aamix_ctl = 1; + /* if no explicit aamix path is present (e.g. for Realtek codecs), + * enable aamix as default -- just for compatibility + */ + spec->aamix_mode = !has_aamix_out_paths(spec); return 0; } @@ -5647,6 +5694,8 @@ static void init_aamix_paths(struct hda_codec *codec) if (!spec->have_aamix_ctl) return; + if (!has_aamix_out_paths(spec)) + return; update_aamix_paths(codec, spec->aamix_mode, spec->out_paths[0], spec->aamix_out_paths[0], spec->autocfg.line_out_type); -- 2.6.3

2 2

[alsa-devel] [PATCH] ASoC: es8328: Fix deemphasis values
by John Keeping 09 Dec '15

09 Dec '15

This is using completely the wrong mask and value when updating the register. Since the correct values are already defined in the header, switch to using a table with explicit constants rather than shifting the array index. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + 2 files changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 84f5eb0..afa6c5d 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -85,7 +85,15 @@ static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); -static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; +static const struct { + int rate; + unsigned int val; +} deemph_settings[] = { + { 0, ES8328_DACCONTROL6_DEEMPH_OFF }, + { 32000, ES8328_DACCONTROL6_DEEMPH_32k }, + { 44100, ES8328_DACCONTROL6_DEEMPH_44_1k }, + { 48000, ES8328_DACCONTROL6_DEEMPH_48k }, +}; static int es8328_set_deemph(struct snd_soc_codec *codec) { @@ -97,21 +105,22 @@ static int es8328_set_deemph(struct snd_soc_codec *codec) * rate. */ if (es8328->deemph) { - best = 1; - for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - es8328->playback_fs) < - abs(deemph_settings[best] - es8328->playback_fs)) + best = 0; + for (i = 1; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i].rate - es8328->playback_fs) < + abs(deemph_settings[best].rate - es8328->playback_fs)) best = i; } - val = best << 1; + val = deemph_settings[best].val; } else { - val = 0; + val = ES8328_DACCONTROL6_DEEMPH_OFF; } dev_dbg(codec->dev, "Set deemphasis %d\n", val); - return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, + ES8328_DACCONTROL6_DEEMPH_MASK, val); } static int es8328_get_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index cb36afe..156c748 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -153,6 +153,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL6_CLICKFREE (1 << 3) #define ES8328_DACCONTROL6_DAC_INVR (1 << 4) #define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_MASK (3 << 6) #define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) #define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) #define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) -- 2.6.3.462.gbe2c914

2 1

[alsa-devel] [PATCH V3 0/2] Add support for Pistachio internal DAC
by Damien Horsley 09 Dec '15

09 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Changes in V3: Mute control definition changed to SOC_SINGLE Codec driver probe function added to associate regmap with codec Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 287 +++++++++++++++++++++ 4 files changed, 315 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

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[alsa-devel] [asoc:topic/intel 75/78] sound/soc/intel/boards/skl_rt286.c:55:2: error: implicit declaration of function 'SND_SOC_DAPM_SINK'
by kbuild test robot 09 Dec '15

09 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 9ec2053b13f75d7ad9c0e6db9763954bd1a1b9ae commit: b34e24d2406f123d5dbbff4fdeebbc2af76b0acd [75/78] ASoC: Intel: Skylake: add wov as int sink config: i386-allmodconfig (attached as .config) reproduce: git checkout b34e24d2406f123d5dbbff4fdeebbc2af76b0acd # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): >> sound/soc/intel/boards/skl_rt286.c:55:2: error: implicit declaration of function 'SND_SOC_DAPM_SINK' [-Werror=implicit-function-declaration] SND_SOC_DAPM_SINK("WoV Sink"), ^ >> sound/soc/intel/boards/skl_rt286.c:55:2: error: initializer element is not constant sound/soc/intel/boards/skl_rt286.c:55:2: note: (near initialization for 'skylake_widgets[5].id') >> sound/soc/intel/boards/skl_rt286.c:49:61: warning: missing braces around initializer [-Wmissing-braces] static const struct snd_soc_dapm_widget skylake_widgets[] = { ^ sound/soc/intel/boards/skl_rt286.c:49:61: note: (near initialization for 'skylake_widgets') cc1: some warnings being treated as errors vim +/SND_SOC_DAPM_SINK +55 sound/soc/intel/boards/skl_rt286.c 43 static const struct snd_kcontrol_new skylake_controls[] = { 44 SOC_DAPM_PIN_SWITCH("Speaker"), 45 SOC_DAPM_PIN_SWITCH("Headphone Jack"), 46 SOC_DAPM_PIN_SWITCH("Mic Jack"), 47 }; 48 > 49 static const struct snd_soc_dapm_widget skylake_widgets[] = { 50 SND_SOC_DAPM_HP("Headphone Jack", NULL), 51 SND_SOC_DAPM_SPK("Speaker", NULL), 52 SND_SOC_DAPM_MIC("Mic Jack", NULL), 53 SND_SOC_DAPM_MIC("DMIC2", NULL), 54 SND_SOC_DAPM_MIC("SoC DMIC", NULL), > 55 SND_SOC_DAPM_SINK("WoV Sink"), 56 }; 57 58 static const struct snd_soc_dapm_route skylake_rt286_map[] = { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Intel HDA / ca0132: quirk for Alienware 17 2015
by Gabriele Martino 09 Dec '15

09 Dec '15

The Alienware 17 (2015) has the same card and pin configuration of the Alienware 15, so the same quirks must be applied. Signed-off-by: Gabriele Martino <g.martino(a)gmx.com>

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[alsa-devel] [PATCH] ALSA: hda - Fix noise problems on Thinkpad T440s
by Takashi Iwai 09 Dec '15

09 Dec '15

Lenovo Thinkpad T440s suffers from constant background noises, and it seems to be a generic hardware issue on this model: https://forums.lenovo.com/t5/ThinkPad-T400-T500-and-newer-T/T440s-speaker-n… As the noise comes from the analog loopback path, disabling the path is the easy workaround. Also, the machine gives significant cracking noises at PM suspend. A workaround found by trial-and-error is to disable the shutup callback currently used for ALC269-variant. This patch addresses these noise issues by introducing a new fixup chain. Although the same workaround might be applicable to other Thinkpad models, it's applied only to T440s (17aa:220c) in this patch, so far, just to be safe (you chicken!). As a compromise, a new model option string "tp440" is provided now, though, so that owners of other Thinkpad models can test it more easily. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=958504 Reported-and-tested-by: Tim Hardeck <thardeck(a)suse.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 49b1d366f82a..8dd2ac13b3af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4204,6 +4204,18 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, } } +/* additional fixup for Thinkpad T440s noise problem */ +static void alc_fixup_tpt440(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->shutup = alc_no_shutup; /* reduce click noise */ + spec->gen.mixer_nid = 0; /* reduce background noise */ + } +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4578,6 +4590,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPT440, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -5051,6 +5064,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPT440] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5332,7 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), @@ -5432,6 +5451,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, + {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {} }; -- 2.6.3

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[alsa-devel] [PATCH] sound/oss: remove VIRT_TO_BUS dependency
by Arnd Bergmann 09 Dec '15

09 Dec '15

The OSS sound drivers used to rely on virt_to_bus(), but don't any more, so we can remove the Kconfig dependency. As a lot of architectures don't provide VIRT_TO_BUS any more, removing the dependency in sounds/oss/ would make the deprecated drivers appear there, which we probably don't want. Instead I'm replacing the simple dependency with 'VIRT_TO_BUS || RPC || NETWINDER' so we can still build these sound drivers for the platforms that need them, but don't change anything on other architectures. As a follow-up, we can remove the virt_to_bus() implementation and Kconfig symbol in the ARM architecture. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 48568fdf847f..4033fe58f0cf 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -240,7 +240,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" - depends on ISA_DMA_API && VIRT_TO_BUS + depends on ISA_DMA_API && (VIRT_TO_BUS || ARCH_RPC || ARCH_NETWINDER) depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make

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[alsa-devel] [PATCH 0/3] ASoC: sti: correction for HBRA support
by Moise Gergaud 09 Dec '15

09 Dec '15

This patchset is V2 of "[PATCH 2/2] ASoC: sti: correction for HBRA support" Moise Gergaud (3): ALSA: pcm: add IEC958 channel status update ASoC: sti: use pcm iec958 channel status helper ASoC: sti: correction for HBRA (High Bit Rate Audio) support include/sound/pcm_iec958.h | 19 ++++++++- sound/core/pcm_iec958.c | 78 +++++++++++++++++++++++++------------ sound/soc/sti/Kconfig | 1 + sound/soc/sti/uniperif_player.c | 86 +++++++++++++++++------------------------ 4 files changed, 107 insertions(+), 77 deletions(-) -- 1.9.1

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[alsa-devel] [RFC] ARM: remove VIRT_TO_BUS
by Arnd Bergmann 09 Dec '15

09 Dec '15

All drivers that are relevant for rpc or footbridge have stopped using virt_to_bus a while ago, so we can remove it and avoid some harmless randconfig warnings for drivers that we do not care about: drivers/atm/zatm.c: In function 'poll_rx': drivers/atm/zatm.c:401:18: warning: 'bus_to_virt' is deprecated [-Wdeprecated-declarations] skb = ((struct rx_buffer_head *) bus_to_virt(here[2]))->skb; FWIW, the remaining drivers using this are: ATM: firestream, zatm, ambassador, horizon ISDN: hisax/netjet V4L: STA2X11, zoran Net: Appletalk LTPC, Tulip DE4x5, Toshiba IrDA WAN: comtrol sv11, cosa, lanmedia, sealevel SCSI: DPT_I2O, buslogic VME: CA91C142 My best guess is that all of the above are so hopelessly obsolete that we are best off removing all of them form the kernel, but that can be done another time. The sound drivers used to rely on virt_to_bus, but don't any more. As a lot of architectures don't provide VIRT_TO_BUS any more, removing the dependency in sounds/oss/ would make the deprecated drivers appear there, which we probably don't want. Instead I'm replacing the simple dependency with 'VIRT_TO_BUS || RPC || NETWINDER' so we can still build these sound drivers for the platforms that need them, but don't change anything on other architectures. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- No hurry for this one. If we can agree that it's a good idea, we can probably do the sound/oss change first, and then do the rest for the following merge window. If nobody cares about picking up the sound/oss patches, we can also just do it as one patch. diff --git a/arch/arm/Kconfig b/arch/arm/Kconfig index a077f59350a2..b03892cecd65 100644 --- a/arch/arm/Kconfig +++ b/arch/arm/Kconfig @@ -556,7 +556,6 @@ config ARCH_RPC select NEED_MACH_IO_H select NEED_MACH_MEMORY_H select NO_IOPORT_MAP - select VIRT_TO_BUS help On the Acorn Risc-PC, Linux can support the internal IDE disk and CD-ROM interface, serial and parallel port, and the floppy drive. diff --git a/arch/arm/include/asm/memory.h b/arch/arm/include/asm/memory.h index c79b57bf71c4..8f8c3a0571a6 100644 --- a/arch/arm/include/asm/memory.h +++ b/arch/arm/include/asm/memory.h @@ -303,20 +303,6 @@ static inline phys_addr_t __virt_to_idmap(unsigned long x) #define __bus_to_pfn(x) __phys_to_pfn(x) #endif -#ifdef CONFIG_VIRT_TO_BUS -#define virt_to_bus virt_to_bus -static inline __deprecated unsigned long virt_to_bus(void *x) -{ - return __virt_to_bus((unsigned long)x); -} - -#define bus_to_virt bus_to_virt -static inline __deprecated void *bus_to_virt(unsigned long x) -{ - return (void *)__bus_to_virt(x); -} -#endif - /* * Conversion between a struct page and a physical address. * diff --git a/arch/arm/mach-footbridge/Kconfig b/arch/arm/mach-footbridge/Kconfig index 735909f90349..7b87bac114a2 100644 --- a/arch/arm/mach-footbridge/Kconfig +++ b/arch/arm/mach-footbridge/Kconfig @@ -68,7 +68,6 @@ config ARCH_NETWINDER select ISA select ISA_DMA select PCI - select VIRT_TO_BUS help Say Y here if you intend to run this kernel on the Rebel.COM NetWinder. Information about this machine can be found at: diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 48568fdf847f..4033fe58f0cf 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -240,7 +240,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" - depends on ISA_DMA_API && VIRT_TO_BUS + depends on ISA_DMA_API && (VIRT_TO_BUS || ARCH_RPC || ARCH_NETWINDER) depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make

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[alsa-devel] [PATCH 1/2] ALSA: hda - Less grumbling about lack of i915 binding
by Takashi Iwai 09 Dec '15

09 Dec '15

The recent commit [6603249dcdbb: ALSA: hda - Enable audio component for old Intel PCH devices] enabled the i915 binding for HDMI/DP on old Intel PCHs. But many boards are without HDMI/DP, and they actually don't need i915 binding, and yet the driver has a check of i915 binding and complains like Haswell must be built with CONFIG_SND_HDA_I915 This error is false-positive, and it should be put only for HSW/BDW, instead of all devices that may be bound with i915. This patch fixes the condition to check, as well as rephrasing the message specific to HSW/BDW HDMI/DP. Fixes: 6603249dcdbb ('ALSA: hda - Enable audio component for old Intel PCH devices') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ee0e316401f9..b49547f3c2e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1954,8 +1954,8 @@ static int azx_probe(struct pci_dev *pci, #endif /* CONFIG_SND_HDA_PATCH_LOADER */ #ifndef CONFIG_SND_HDA_I915 - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - dev_err(card->dev, "Haswell must build in CONFIG_SND_HDA_I915\n"); + if (CONTROLLER_IN_GPU(pci)) + dev_err(card->dev, "Haswell/Broadwell HDMI/DP must build in CONFIG_SND_HDA_I915\n"); #endif if (schedule_probe) -- 2.6.3

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[alsa-devel] next-20151208 - CONFIG_SND_HDA_I915
by Valdis Kletnieks 09 Dec '15

09 Dec '15

I'm experiencing a spot of trouble with this commit: commit aad730d0704545ad97654bd929b0aba05adb1436 Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon Dec 2 13:33:57 2013 +0100 ALSA: hda - Always do delayed probes for HD-audio devices That adds some code: #ifndef CONFIG_SND_HDA_I915 if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) dev_err(card->dev, "Haswell must build in CONFIG_SND_HDA_I915\n"); "Oh, bother" -- Winnie the Pooh So I go looking how to fix this, and 'make menuconfig' tells me: Symbol: SND_HDA_I915 [=n] x x Type : boolean x x Defined at sound/hda/Kconfig:8 x x Depends on: SOUND [=y] && !M68K && !UML && SND [=y] && DRM_I915 [=n] && SND_HDA_CORE [=y] And why is DRM_I915=n? # lspci | grep NV 01:00.0 VGA compatible controller: NVIDIA Corporation GF108GLM [NVS 5200M] (rev a1) Because this Dell laptop has NVidia graphics onboard, and no sign of I915 graphics that needs I915 DRM. I think the infrastructure that you're trying to re-use from DRM_I915 needs to be moved to another symbol which can be SELECT/DEPEND by both DRM_I915 and SND_HSA_I915. Comments/suggestions?

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[alsa-devel] [PATCH] ASoC: rsnd: tidyup data align position for capture
by Kuninori Morimoto 08 Dec '15

08 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> L/R channel data has been treated as inverted on R-Car sound 16bit mode, Thus, 4689032b1("ASoC: rsnd: tidyup data align position") tidyuped data align position. But it couldn't care about capture case. This patch cares both playback/capture Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/cmd.c | 1 + sound/soc/sh/rcar/core.c | 13 +++++++++++-- sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 2 ++ 4 files changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index ab904c3..cd1f064 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -80,6 +80,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, dev_dbg(dev, "ctu/mix path = 0x%08x", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); + rsnd_mod_write(mod, CMD_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); rsnd_adg_set_cmd_timsel_gen2(mod, io); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bf596ff..0945cff 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -273,13 +273,22 @@ u32 rsnd_get_adinr_chan(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); - struct rsnd_mod *target = src ? src : ssi; + struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 val = 0x76543210; u32 mask = ~0; + if (rsnd_io_is_play(io)) { + struct rsnd_mod *src = rsnd_io_to_mod_src(io); + + target = src ? src : ssi; + } else { + struct rsnd_mod *cmd = rsnd_io_to_mod_cmd(io); + + target = cmd ? cmd : ssi; + } + mask <<= runtime->channels * 4; val = val & mask; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 50fc730..7c5485e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -242,6 +242,7 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20), + RSND_GEN_M_REG(CMD_BUSIF_DALIGN,0x188, 0x20), RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 970e130..ad854d6 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -81,6 +81,7 @@ enum rsnd_reg { RSND_REG_SCU_SYS_INT_EN0, RSND_REG_SCU_SYS_INT_EN1, /* Gen2 only */ RSND_REG_CMD_CTRL, /* Gen2 only */ + RSND_REG_CMD_BUSIF_DALIGN, /* Gen2 only */ RSND_REG_CMD_ROUTE_SLCT, RSND_REG_CMDOUT_TIMSEL, /* Gen2 only */ RSND_REG_CTU_CTUIR, @@ -319,6 +320,7 @@ struct rsnd_dai_stream { #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) #define rsnd_io_to_mod_mix(io) rsnd_io_to_mod((io), RSND_MOD_MIX) #define rsnd_io_to_mod_dvc(io) rsnd_io_to_mod((io), RSND_MOD_DVC) +#define rsnd_io_to_mod_cmd(io) rsnd_io_to_mod((io), RSND_MOD_CMD) #define rsnd_io_to_rdai(io) ((io)->rdai) #define rsnd_io_to_priv(io) (rsnd_rdai_to_priv(rsnd_io_to_rdai(io))) #define rsnd_io_is_play(io) (&rsnd_io_to_rdai(io)->playback == io) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: mediatek: Use current HW pointer for pointer callback
by Koro Chen 08 Dec '15

08 Dec '15

Previously we recorded "last interrupt position" and used it in pointer callback. This is not correct implementation, and it causes underruns when user space monitors buffer level to decide when to send next data chunk in low latency application. Remove position recording in IRQ handler and also hw_ptr in struct mtk_afe_memif used to record that, and let pointer callback reports current HW pointer instead. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-common.h | 1 - sound/soc/mediatek/mtk-afe-pcm.c | 22 +++++++++++----------- 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-common.h b/sound/soc/mediatek/mtk-afe-common.h index cc4393c..9b1af1a 100644 --- a/sound/soc/mediatek/mtk-afe-common.h +++ b/sound/soc/mediatek/mtk-afe-common.h @@ -92,7 +92,6 @@ struct mtk_afe_memif_data { struct mtk_afe_memif { unsigned int phys_buf_addr; int buffer_size; - unsigned int hw_ptr; /* Previous IRQ's HW ptr */ struct snd_pcm_substream *substream; const struct mtk_afe_memif_data *data; const struct mtk_afe_irq_data *irqdata; diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 7f71343..5399a0e 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -175,8 +175,17 @@ static snd_pcm_uframes_t mtk_afe_pcm_pointer struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_afe *afe = snd_soc_platform_get_drvdata(rtd->platform); struct mtk_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + unsigned int hw_ptr; + int ret; + + ret = regmap_read(afe->regmap, memif->data->reg_ofs_cur, &hw_ptr); + if (ret || hw_ptr == 0) { + dev_err(afe->dev, "%s hw_ptr err\n", __func__); + hw_ptr = memif->phys_buf_addr; + } - return bytes_to_frames(substream->runtime, memif->hw_ptr); + return bytes_to_frames(substream->runtime, + hw_ptr - memif->phys_buf_addr); } static const struct snd_pcm_ops mtk_afe_pcm_ops = { @@ -602,7 +611,6 @@ static int mtk_afe_dais_hw_params(struct snd_pcm_substream *substream, memif->phys_buf_addr = substream->runtime->dma_addr; memif->buffer_size = substream->runtime->dma_bytes; - memif->hw_ptr = 0; /* start */ regmap_write(afe->regmap, @@ -737,7 +745,6 @@ static int mtk_afe_dais_trigger(struct snd_pcm_substream *substream, int cmd, /* and clear pending IRQ */ regmap_write(afe->regmap, AFE_IRQ_CLR, 1 << memif->data->irq_clr_shift); - memif->hw_ptr = 0; return 0; default: return -EINVAL; @@ -1081,7 +1088,7 @@ static const struct regmap_config mtk_afe_regmap_config = { static irqreturn_t mtk_afe_irq_handler(int irq, void *dev_id) { struct mtk_afe *afe = dev_id; - unsigned int reg_value, hw_ptr; + unsigned int reg_value; int i, ret; ret = regmap_read(afe->regmap, AFE_IRQ_STATUS, &reg_value); @@ -1097,13 +1104,6 @@ static irqreturn_t mtk_afe_irq_handler(int irq, void *dev_id) if (!(reg_value & (1 << memif->data->irq_clr_shift))) continue; - ret = regmap_read(afe->regmap, memif->data->reg_ofs_cur, - &hw_ptr); - if (ret || hw_ptr == 0) { - dev_err(afe->dev, "%s hw_ptr err\n", __func__); - hw_ptr = memif->phys_buf_addr; - } - memif->hw_ptr = hw_ptr - memif->phys_buf_addr; snd_pcm_period_elapsed(memif->substream); } -- 1.7.9.5

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[alsa-devel] [PATCH v2 00/12] ASoC: Intel: Skylake: Add support for loadable modules
by Vinod Koul 08 Dec '15

08 Dec '15

This series adds first the infrastructure for loadable modules in SKL aDSP. The loadable modules are loaded using load/unload IPC to aDSP thru code load DMA and can be configured by user thru TLV byte kcontrols for these modules. This also adds the implementation of new DAPM sinks for phrase detection modules, and adds support for keeping controller active while is suspend to support ignore_suspend DAIs Changes on v2: - squash two load module IPCs and load modules usgae patches - add comments for module loading - add active suspend two patches Dharageswari R (2): ASoC: Intel: Skylake: Add support for Loadable modules ASoC: Intel: Skylake: Add support for Mic Select module Jeeja KP (4): ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture ASoC: Intel: Skylake: Add support for active suspend Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on Praveen Diwakar (1): ASoC: Intel: Skylake: Update ignore suspend for rt286 machine Vinod Koul (2): ASoC: Intel: Skylake: add wov as int sink ASoc: Intel: Skylake: Fix the dapm machine map sound/soc/intel/boards/skl_rt286.c | 73 ++++++++++- sound/soc/intel/skylake/skl-messages.c | 40 ++++++ sound/soc/intel/skylake/skl-pcm.c | 27 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 18 ++- sound/soc/intel/skylake/skl-sst-ipc.c | 106 ++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 9 ++ sound/soc/intel/skylake/skl-sst.c | 175 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.c | 43 ++++++- sound/soc/intel/skylake/skl-topology.h | 12 +- sound/soc/intel/skylake/skl-tplg-interface.h | 19 ++- sound/soc/intel/skylake/skl.c | 28 ++++- sound/soc/intel/skylake/skl.h | 2 + 12 files changed, 533 insertions(+), 19 deletions(-) -- 1.9.1

2 16

[alsa-devel] [PATCH 00/16] ASoC: Intel: Skylake: Add support for loadable modules
by Vinod Koul 08 Dec '15

08 Dec '15

This series adds first the infrastructure for loadable modules in SKL aDSP. The loadable modules are loaded using load/unload IPC to aDSP thru code load DMA and can be configured by user thru TLV byte kcontrols for these modules. This also adds the implementation of new DAPM sinks for phrase detection modules. Dharageswari R (2): ASoC: Intel: Skylake: Add support for Load/Unload IPCs ASoC: Intel: Skylake: Add support for Loadable modules Dharageswari.R (1): ASoC: Intel: Skylake: Add support for Mic Select module Jeeja KP (8): ASoC: Intel: Skylake: Update DMIC DAIs and capabilities ASoC: Intel: Skylake: Add helper routines to handle module params ASoC: Intel: Skylake: Add helper routine to handle Algo parameter ASoC: Intel: Skylake: Add support to configure module params ASoC: Intel: Skylake: Add tlv byte kcontrols ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on Vinod Koul (2): ASoC: Intel: Skylake: add wov as int sink ASoc: Intel: Skylake: Fix the dapm machine map sound/soc/intel/boards/skl_rt286.c | 58 ++++++- sound/soc/intel/skylake/skl-messages.c | 105 +++++++++++- sound/soc/intel/skylake/skl-pcm.c | 23 ++- sound/soc/intel/skylake/skl-sst-dsp.h | 18 +- sound/soc/intel/skylake/skl-sst-ipc.c | 100 +++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 9 + sound/soc/intel/skylake/skl-sst.c | 175 +++++++++++++++++++ sound/soc/intel/skylake/skl-topology.c | 242 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 32 +++- sound/soc/intel/skylake/skl-tplg-interface.h | 23 ++- 10 files changed, 761 insertions(+), 24 deletions(-) -- 2.4.3

2 36

[alsa-devel] [PATCH v2 00/13] ASoC: topology: Add support for PCM, BE & CC links
by mengdong.lin＠linux.intel.com 08 Dec '15

08 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows the topology core to create PCM devices dynamically, as well as configure BE and CC DAI links. Mengdong Lin (12): ASoC: Vendor drivers get a link's runtime by snd_soc_get_pcm_runtime() ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add a link ASoC: Add add_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components ASoC: The soc card can have auxiliary components ASoC: Support adding a DAI dynamically ASoC: topology: Add PCM DAIs dynamically when loading them ASoC: topology: Add support for FE DAI links Vedang Patel (1): ASoC: topology: Add support for BE and CC DAI Links include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 22 +- include/sound/soc.h | 38 +- sound/soc/fsl/fsl-asoc-card.c | 5 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/rsrc-card.c | 6 +- sound/soc/soc-core.c | 629 ++++++++++++++++++--------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 284 ++++++++++-- sound/soc/tegra/tegra_wm8903.c | 3 +- 24 files changed, 879 insertions(+), 324 deletions(-) -- 1.9.1

4 18

[alsa-devel] [PATCH] ASoC: fsl: use correct format string for dma_addr_t
by Arnd Bergmann 08 Dec '15

08 Dec '15

We get a warning for the imx-pcm-fiq driver when CONFIG_LPAE is enabled on ARM, because dma_addr_t is 64-bit then: sound/soc/fsl/imx-pcm-fiq.c: In function 'snd_imx_pcm_mmap': sound/soc/fsl/imx-pcm-fiq.c:223:107: warning: format '%x' expects argument of type 'unsigned int', but argument 6 has type 'dma_addr_t {aka long long unsigned int}' [-Wformat=] This changes the printk to use the correct format string for printing a dma_addr_t. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/imx-pcm-fiq.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 7abf6a079574..49d7513f429e 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,9 +220,9 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, runtime->dma_area, - runtime->dma_addr, + &runtime->dma_addr, runtime->dma_bytes); return ret; } -- 2.1.0.rc2

2 1

[alsa-devel] [PATCH 01/19] ASoC: da7218: avoid 64-bit compile warning
by Arnd Bergmann 08 Dec '15

08 Dec '15

When building the da7218 driver on a 64-bit architecture, we get a harmless warning: sound/soc/codecs/da7218.c: In function 'da7218_of_get_id': sound/soc/codecs/da7218.c:2261:10: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] This changes the code to use uintptr_t to ensure we have an integer type of the same size as a pointer and won't get a warning on any architecture. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4d50934abd22 ("ASoC: da7218: Add da7218 codec driver") --- 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 4fee7aeaadc7..eacde128c4d6 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2258,7 +2258,7 @@ static inline int da7218_of_get_id(struct device *dev) const struct of_device_id *id = of_match_device(da7218_of_match, dev); if (id) - return (int) id->data; + return (uintptr_t)id->data; else return -EINVAL; } -- 2.1.0.rc2

2 2

[alsa-devel] [PATCH] ASoC: Add max98926 codec driver
by anish kumar 08 Dec '15

08 Dec '15

Changes since v1: Made some widgets as supply widgets Changed the names a bit Moved some registers to volatile struct Some coding style changes added the print for diagnostics Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98926.txt | 32 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98926.c | 599 +++++++++++++++ sound/soc/codecs/max98926.h | 848 +++++++++++++++++++++ 5 files changed, 1486 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 sound/soc/codecs/max98926.c create mode 100644 sound/soc/codecs/max98926.h diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt new file mode 100644 index 0000000..0b7f4e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98926.txt @@ -0,0 +1,32 @@ +max98926 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98926" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + + - imon-slot-no : slot number used to send current information + + - interleave-mode : When using two MAX98926 in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98926@1a { + compatible = "maxim,max98926"; + vmon-slot-no = <0>; + imon-slot-no = <2>; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cfdafc4..3fe8bd3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_MAX98926 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C @@ -331,6 +332,10 @@ config SND_SOC_AK4613 config SND_SOC_AK4641 tristate +config SND_SOC_MAX98926 + tristate "MAX98926 CODEC" + depends on I2C + config SND_SOC_AK4642 tristate "AKM AK4642 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f632fc4..a38b536 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-ak4554-objs := ak4554.o snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o +snd-soc-max98926-objs := max98926.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o @@ -222,6 +223,7 @@ obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c new file mode 100644 index 0000000..3ec54ce --- /dev/null +++ b/sound/soc/codecs/max98926.c @@ -0,0 +1,599 @@ +/* + * max98926.c -- ALSA SoC MAX98926 driver + * Copyright 2013-15 Maxim Integrated Products + * 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/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98926.h" + +static const char * const max98926_boost_voltage_txt[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static const char *const max98926_dai_txt[] = { + "Left", "Right", "LeftRight", "LeftRightDiv2", +}; + +static const char *const max98926_pdm_ch_text[] = { + "Current", "Voltage", +}; + +static const char *const max98926_hpf_cutoff_txt[] = { + "Disable", "DC Block", "100Hz", + "200Hz", "400Hz", "800Hz", +}; + +static struct reg_default max98926_reg[] = { + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x04 }, /* DAI Clock Mode 1 */ + { 0x1B, 0x00 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ +}; + +static const struct soc_enum max98926_voltage_enum[] = { + SOC_ENUM_SINGLE(MAX98926_DAI_CLK_DIV_N_LSBS, 0, + ARRAY_SIZE(max98926_pdm_ch_text), + max98926_pdm_ch_text), +}; + +static const struct snd_kcontrol_new max98926_voltage_control = + SOC_DAPM_ENUM("Route", max98926_voltage_enum); + +static const struct soc_enum max98926_current_enum[] = { + SOC_ENUM_SINGLE(MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_SOURCE_1_SHIFT, + ARRAY_SIZE(max98926_pdm_ch_text), + max98926_pdm_ch_text), +}; + +static const struct snd_kcontrol_new max98926_current_control = + SOC_DAPM_ENUM("Route", max98926_current_enum); + +static const struct snd_kcontrol_new max98926_mixer_controls[] = { + SOC_DAPM_SINGLE("PCM", MAX98926_SPK_AMP, + MAX98926_INSELECT_MODE_SHIFT, 0, 0), + SOC_DAPM_SINGLE("PDM", MAX98926_SPK_AMP, + MAX98926_INSELECT_MODE_SHIFT, 1, 0), +}; + +static const struct snd_kcontrol_new max98926_dai_controls[] = { + SOC_DAPM_SINGLE("Left", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 0, 0), + SOC_DAPM_SINGLE("Right", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LeftRight", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 2, 0), + SOC_DAPM_SINGLE("LeftRightDiv2", MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, 3, 0), +}; + +static const struct snd_soc_dapm_widget max98926_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Amp Enable", NULL, MAX98926_BLOCK_ENABLE, + MAX98926_SPK_EN_SHIFT, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98926_GLOBAL_ENABLE, + MAX98926_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VI Enable", MAX98926_BLOCK_ENABLE, + MAX98926_ADC_IMON_EN_WIDTH | + MAX98926_ADC_VMON_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("BST Enable", MAX98926_BLOCK_ENABLE, + MAX98926_BST_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("BE_OUT"), + SND_SOC_DAPM_MIXER("PCM Sel", MAX98926_SPK_AMP, + MAX98926_INSELECT_MODE_SHIFT, 0, + &max98926_mixer_controls[0], + ARRAY_SIZE(max98926_mixer_controls)), + SND_SOC_DAPM_MIXER("DAI Sel", + MAX98926_GAIN, MAX98926_DAC_IN_SEL_SHIFT, 0, + &max98926_dai_controls[0], + ARRAY_SIZE(max98926_dai_controls)), + SND_SOC_DAPM_MUX("PDM CH1 Source", + MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_CURRENT_SHIFT, + 0, &max98926_current_control), + SND_SOC_DAPM_MUX("PDM CH0 Source", + MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_VOLTAGE_SHIFT, + 0, &max98926_voltage_control), +}; + +static const struct snd_soc_dapm_route max98926_audio_map[] = { + {"VI Enable", NULL, "DAI_OUT"}, + {"DAI Sel", "Left", "VI Enable"}, + {"DAI Sel", "Right", "VI Enable"}, + {"DAI Sel", "LeftRight", "VI Enable"}, + {"DAI Sel", "LeftRightDiv2", "VI Enable"}, + {"PCM Sel", "PCM", "DAI Sel"}, + + {"PDM CH1 Source", "Current", "DAI_OUT"}, + {"PDM CH1 Source", "Voltage", "DAI_OUT"}, + {"PDM CH0 Source", "Current", "DAI_OUT"}, + {"PDM CH0 Source", "Voltage", "DAI_OUT"}, + {"PCM Sel", "Analog", "PDM CH1 Source"}, + {"PCM Sel", "Analog", "PDM CH0 Source"}, + {"Amp Enable", NULL, "PCM Sel"}, + + {"BST Enable", NULL, "Amp Enable"}, + {"BE_OUT", NULL, "BST Enable"}, +}; + +static bool max98926_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98926_VBAT_DATA: + case MAX98926_VBST_DATA: + case MAX98926_LIVE_STATUS0: + case MAX98926_LIVE_STATUS1: + case MAX98926_LIVE_STATUS2: + case MAX98926_STATE0: + case MAX98926_STATE1: + case MAX98926_STATE2: + case MAX98926_FLAG0: + case MAX98926_FLAG1: + case MAX98926_FLAG2: + case MAX98926_VERSION: + return true; + default: + return false; + } +} + +static bool max98926_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98926_IRQ_CLEAR0: + case MAX98926_IRQ_CLEAR1: + case MAX98926_IRQ_CLEAR2: + case MAX98926_ALC_HOLD_RLS: + return false; + default: + return true; + } +}; + +DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0); + +static SOC_ENUM_SINGLE_DECL(max98926_dac_hpf_cutoff, + MAX98926_FILTERS, MAX98926_DAC_HPF_SHIFT, + max98926_hpf_cutoff_txt); + +static SOC_ENUM_SINGLE_DECL(max98926_boost_voltage, + MAX98926_CONFIGURATION, MAX98926_BST_VOUT_SHIFT, + max98926_boost_voltage_txt); + +static const struct snd_kcontrol_new max98926_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98926_GAIN, + MAX98926_SPK_GAIN_SHIFT, + (1<<MAX98926_SPK_GAIN_WIDTH)-1, 0, + max98926_spk_tlv), + SOC_SINGLE("Ramp Switch", MAX98926_GAIN_RAMPING, + MAX98926_SPK_RMP_EN_SHIFT, 1, 0), + SOC_SINGLE("ZCD Switch", MAX98926_GAIN_RAMPING, + MAX98926_SPK_ZCD_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Switch", MAX98926_THRESHOLD, + MAX98926_ALC_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Threshold", MAX98926_THRESHOLD, + MAX98926_ALC_TH_SHIFT, + (1<<MAX98926_ALC_TH_WIDTH)-1, 0), + SOC_ENUM("Boost Output Voltage", max98926_boost_voltage), + SOC_ENUM("DAC HPF Cutoff", max98926_dac_hpf_cutoff), + SOC_DOUBLE("PDM Channel One", MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_CHANNEL_1_SHIFT, + MAX98926_PDM_CHANNEL_1_HIZ, 1, 0), + SOC_DOUBLE("PDM Channel Zero", MAX98926_DAI_CLK_DIV_N_LSBS, + MAX98926_PDM_CHANNEL_0_SHIFT, + MAX98926_PDM_CHANNEL_0_HIZ, 1, 0), +}; + +static const struct { + int rate; + int sr; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + }, + { + .rate = 11025, + .sr = 1, + }, + { + .rate = 12000, + .sr = 2, + }, + { + .rate = 16000, + .sr = 3, + }, + { + .rate = 22050, + .sr = 4, + }, + { + .rate = 24000, + .sr = 5, + }, + { + .rate = 32000, + .sr = 6, + }, + { + .rate = 44100, + .sr = 7, + }, + { + .rate = 48000, + .sr = 8, + }, +}; + +static void max98926_set_sense_data(struct max98926_priv *max98926) +{ + if (max98926->interleave_mode) { + /* set VMON slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VMON, + MAX98926_DAI_VMON_EN_MASK, + MAX98926_DAI_VMON_EN_MASK); + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VMON, + MAX98926_DAI_VMON_SLOT_MASK, + max98926->v_slot); + /* set IMON slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_IMON, + MAX98926_DAI_IMON_EN_MASK, + MAX98926_DAI_IMON_EN_MASK); + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_IMON, + MAX98926_DAI_IMON_SLOT_MASK, + max98926->i_slot); + } else { + /* enable interleave mode */ + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_INTERLEAVE_MASK, + MAX98926_DAI_INTERLEAVE_MASK); + /* set interleave slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VBAT, + MAX98926_DAI_INTERLEAVE_SLOT_MASK, + max98926->v_slot); + } +} + +static int max98926_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98926_set_sense_data(max98926); + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = MAX98926_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98926_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = MAX98926_DAI_BCI_MASK | MAX98926_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_write(max98926->regmap, + MAX98926_FORMAT, MAX98926_DAI_DLY_MASK); + regmap_update_bits(max98926->regmap, MAX98926_FORMAT, + MAX98926_DAI_BCI_MASK, invert); + return 0; +} + +static int max98926_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int dai_sr = -EINVAL; + int rate = params_rate(params), i; + struct snd_soc_codec *codec = dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98926->ch_size; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_16); + max98926->ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_24); + max98926->ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_32); + max98926->ch_size = 32; + break; + default: + dev_dbg(codec->dev, "format unsupported %d", + params_format(params)); + return -EINVAL; + } + + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, + MAX98926_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, + MAX98926_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, + MAX98926_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + /* find the closest rate */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + dai_sr = rate_table[i].sr; + break; + } + } + if (dai_sr < 0) + return -EINVAL; + + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_SR_MASK, dai_sr << MAX98926_DAI_SR_SHIFT); + return 0; +} + +#define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98926_dai_ops = { + .set_fmt = max98926_dai_set_fmt, + .hw_params = max98926_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98926_dai[] = { +{ + .name = "max98926-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98926_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98926_FORMATS, + }, + .ops = &max98926_dai_ops, +} +}; + +static int max98926_probe(struct snd_soc_codec *codec) +{ + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + + max98926->codec = codec; + codec->control_data = max98926->regmap; + regmap_write(max98926->regmap, MAX98926_GLOBAL_ENABLE, 0x00); + regmap_write(max98926->regmap, MAX98926_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98926->regmap, MAX98926_FILTERS, 0xD8); + regmap_write(max98926->regmap, MAX98926_ALC_CONFIGURATION, 0xF8); + regmap_write(max98926->regmap, MAX98926_CONFIGURATION, 0xF0); + /* Disable ALC muting */ + regmap_write(max98926->regmap, MAX98926_BOOST_LIMITER, 0xF8); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98926 = { + .probe = max98926_probe, + .controls = max98926_snd_controls, + .num_controls = ARRAY_SIZE(max98926_snd_controls), + .dapm_routes = max98926_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98926_audio_map), + .dapm_widgets = max98926_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98926_dapm_widgets), +}; + +static struct regmap_config max98926_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98926_VERSION, + .reg_defaults = max98926_reg, + .num_reg_defaults = ARRAY_SIZE(max98926_reg), + .volatile_reg = max98926_volatile_register, + .readable_reg = max98926_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98926_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret, reg; + u32 value; + struct max98926_priv *max98926; + + max98926 = devm_kzalloc(&i2c->dev, + sizeof(*max98926), GFP_KERNEL); + if (!max98926) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98926); + max98926->regmap = devm_regmap_init_i2c(i2c, &max98926_regmap); + if (IS_ERR(max98926->regmap)) { + ret = PTR_ERR(max98926->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err_out; + } + if (of_property_read_bool(i2c->dev.of_node, "interleave-mode")) + max98926->interleave_mode = true; + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { + if (value > MAX98926_DAI_VMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "vmon slot number is wrong:\n"); + return -EINVAL; + } + max98926->v_slot = value; + } + if (!of_property_read_u32(i2c->dev.of_node, "imon-slot-no", &value)) { + if (value > MAX98926_DAI_IMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "imon slot number is wrong:\n"); + return -EINVAL; + } + max98926->i_slot = value; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98926, + max98926_dai, ARRAY_SIZE(max98926_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); + ret = regmap_read(max98926->regmap, + MAX98926_VERSION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %x\n", reg); + return -EINVAL; + } + dev_info(&i2c->dev, "device version: %x\n", reg); +err_out: + return ret; +} + +static int max98926_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98926_i2c_id[] = { + { "max98926", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98926_i2c_id); + +static const struct of_device_id max98926_of_match[] = { + { .compatible = "maxim, max98926", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98926_of_match); + +static struct i2c_driver max98926_i2c_driver = { + .driver = { + .name = "max98926", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98926_of_match), + .pm = NULL, + }, + .probe = max98926_i2c_probe, + .remove = max98926_i2c_remove, + .id_table = max98926_i2c_id, +}; + +module_i2c_driver(max98926_i2c_driver) +MODULE_DESCRIPTION("ALSA SoC MAX98926 driver"); +MODULE_AUTHOR("Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98926.h b/sound/soc/codecs/max98926.h new file mode 100644 index 0000000..5243f82 --- /dev/null +++ b/sound/soc/codecs/max98926.h @@ -0,0 +1,848 @@ +/* + * max98926.h -- MAX98926 ALSA SoC Audio driver + * Copyright 2013-2015 Maxim Integrated Products + * 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 _MAX98926_H +#define _MAX98926_H + +#define MAX98926_CHIP_VERSION 0x40 +#define MAX98926_CHIP_VERSION1 0x50 + +#define MAX98926_VBAT_DATA 0x00 +#define MAX98926_VBST_DATA 0x01 +#define MAX98926_LIVE_STATUS0 0x02 +#define MAX98926_LIVE_STATUS1 0x03 +#define MAX98926_LIVE_STATUS2 0x04 +#define MAX98926_STATE0 0x05 +#define MAX98926_STATE1 0x06 +#define MAX98926_STATE2 0x07 +#define MAX98926_FLAG0 0x08 +#define MAX98926_FLAG1 0x09 +#define MAX98926_FLAG2 0x0A +#define MAX98926_IRQ_ENABLE0 0x0B +#define MAX98926_IRQ_ENABLE1 0x0C +#define MAX98926_IRQ_ENABLE2 0x0D +#define MAX98926_IRQ_CLEAR0 0x0E +#define MAX98926_IRQ_CLEAR1 0x0F +#define MAX98926_IRQ_CLEAR2 0x10 +#define MAX98926_MAP0 0x11 +#define MAX98926_MAP1 0x12 +#define MAX98926_MAP2 0x13 +#define MAX98926_MAP3 0x14 +#define MAX98926_MAP4 0x15 +#define MAX98926_MAP5 0x16 +#define MAX98926_MAP6 0x17 +#define MAX98926_MAP7 0x18 +#define MAX98926_MAP8 0x19 +#define MAX98926_DAI_CLK_MODE1 0x1A +#define MAX98926_DAI_CLK_MODE2 0x1B +#define MAX98926_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98926_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98926_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98926_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98926_FORMAT 0x20 +#define MAX98926_TDM_SLOT_SELECT 0x21 +#define MAX98926_DOUT_CFG_VMON 0x22 +#define MAX98926_DOUT_CFG_IMON 0x23 +#define MAX98926_DOUT_CFG_VBAT 0x24 +#define MAX98926_DOUT_CFG_VBST 0x25 +#define MAX98926_DOUT_CFG_FLAG 0x26 +#define MAX98926_DOUT_HIZ_CFG1 0x27 +#define MAX98926_DOUT_HIZ_CFG2 0x28 +#define MAX98926_DOUT_HIZ_CFG3 0x29 +#define MAX98926_DOUT_HIZ_CFG4 0x2A +#define MAX98926_DOUT_DRV_STRENGTH 0x2B +#define MAX98926_FILTERS 0x2C +#define MAX98926_GAIN 0x2D +#define MAX98926_GAIN_RAMPING 0x2E +#define MAX98926_SPK_AMP 0x2F +#define MAX98926_THRESHOLD 0x30 +#define MAX98926_ALC_ATTACK 0x31 +#define MAX98926_ALC_ATTEN_RLS 0x32 +#define MAX98926_ALC_HOLD_RLS 0x33 +#define MAX98926_ALC_CONFIGURATION 0x34 +#define MAX98926_BOOST_CONVERTER 0x35 +#define MAX98926_BLOCK_ENABLE 0x36 +#define MAX98926_CONFIGURATION 0x37 +#define MAX98926_GLOBAL_ENABLE 0x38 +#define MAX98926_BOOST_LIMITER 0x3A +#define MAX98926_VERSION 0xFF + +#define MAX98926_REG_CNT (MAX98926_R03A_BOOST_LIMITER+1) + +#define MAX98926_PDM_CURRENT_MASK (1<<7) +#define MAX98926_PDM_CURRENT_SHIFT 7 +#define MAX98926_PDM_VOLTAGE_MASK (1<<3) +#define MAX98926_PDM_VOLTAGE_SHIFT 3 +#define MAX98926_PDM_CHANNEL_0_MASK (1<<2) +#define MAX98926_PDM_CHANNEL_0_SHIFT 2 +#define MAX98926_PDM_CHANNEL_1_MASK (1<<6) +#define MAX98926_PDM_CHANNEL_1_SHIFT 6 +#define MAX98926_PDM_CHANNEL_1_HIZ 5 +#define MAX98926_PDM_CHANNEL_0_HIZ 1 +#define MAX98926_PDM_SOURCE_0_SHIFT 0 +#define MAX98926_PDM_SOURCE_0_MASK (1<<0) +#define MAX98926_PDM_SOURCE_1_MASK (1<<4) +#define MAX98926_PDM_SOURCE_1_SHIFT 4 + +/* MAX98926 Register Bit Fields */ + +/* MAX98926_R002_LIVE_STATUS0 */ +#define MAX98926_THERMWARN_STATUS_MASK (1<<3) +#define MAX98926_THERMWARN_STATUS_SHIFT 3 +#define MAX98926_THERMWARN_STATUS_WIDTH 1 +#define MAX98926_THERMSHDN_STATUS_MASK (1<<1) +#define MAX98926_THERMSHDN_STATUS_SHIFT 1 +#define MAX98926_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98926_R003_LIVE_STATUS1 */ +#define MAX98926_SPKCURNT_STATUS_MASK (1<<5) +#define MAX98926_SPKCURNT_STATUS_SHIFT 5 +#define MAX98926_SPKCURNT_STATUS_WIDTH 1 +#define MAX98926_WATCHFAIL_STATUS_MASK (1<<4) +#define MAX98926_WATCHFAIL_STATUS_SHIFT 4 +#define MAX98926_WATCHFAIL_STATUS_WIDTH 1 +#define MAX98926_ALCINFH_STATUS_MASK (1<<3) +#define MAX98926_ALCINFH_STATUS_SHIFT 3 +#define MAX98926_ALCINFH_STATUS_WIDTH 1 +#define MAX98926_ALCACT_STATUS_MASK (1<<2) +#define MAX98926_ALCACT_STATUS_SHIFT 2 +#define MAX98926_ALCACT_STATUS_WIDTH 1 +#define MAX98926_ALCMUT_STATUS_MASK (1<<1) +#define MAX98926_ALCMUT_STATUS_SHIFT 1 +#define MAX98926_ALCMUT_STATUS_WIDTH 1 +#define MAX98926_ACLP_STATUS_MASK (1<<0) +#define MAX98926_ACLP_STATUS_SHIFT 0 +#define MAX98926_ACLP_STATUS_WIDTH 1 + +/* MAX98926_R004_LIVE_STATUS2 */ +#define MAX98926_SLOTOVRN_STATUS_MASK (1<<6) +#define MAX98926_SLOTOVRN_STATUS_SHIFT 6 +#define MAX98926_SLOTOVRN_STATUS_WIDTH 1 +#define MAX98926_INVALSLOT_STATUS_MASK (1<<5) +#define MAX98926_INVALSLOT_STATUS_SHIFT 5 +#define MAX98926_INVALSLOT_STATUS_WIDTH 1 +#define MAX98926_SLOTCNFLT_STATUS_MASK (1<<4) +#define MAX98926_SLOTCNFLT_STATUS_SHIFT 4 +#define MAX98926_SLOTCNFLT_STATUS_WIDTH 1 +#define MAX98926_VBSTOVFL_STATUS_MASK (1<<3) +#define MAX98926_VBSTOVFL_STATUS_SHIFT 3 +#define MAX98926_VBSTOVFL_STATUS_WIDTH 1 +#define MAX98926_VBATOVFL_STATUS_MASK (1<<2) +#define MAX98926_VBATOVFL_STATUS_SHIFT 2 +#define MAX98926_VBATOVFL_STATUS_WIDTH 1 +#define MAX98926_IMONOVFL_STATUS_MASK (1<<1) +#define MAX98926_IMONOVFL_STATUS_SHIFT 1 +#define MAX98926_IMONOVFL_STATUS_WIDTH 1 +#define MAX98926_VMONOVFL_STATUS_MASK (1<<0) +#define MAX98926_VMONOVFL_STATUS_SHIFT 0 +#define MAX98926_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98926_R005_STATE0 */ +#define MAX98926_THERMWARN_END_STATE_MASK (1<<3) +#define MAX98926_THERMWARN_END_STATE_SHIFT 3 +#define MAX98926_THERMWARN_END_STATE_WIDTH 1 +#define MAX98926_THERMWARN_BGN_STATE_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_STATE_SHIFT 1 +#define MAX98926_THERMWARN_BGN_STATE_WIDTH 1 +#define MAX98926_THERMSHDN_END_STATE_MASK (1<<1) +#define MAX98926_THERMSHDN_END_STATE_SHIFT 1 +#define MAX98926_THERMSHDN_END_STATE_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_STATE_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_STATE_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98926_R006_STATE1 */ +#define MAX98926_SPRCURNT_STATE_MASK (1<<5) +#define MAX98926_SPRCURNT_STATE_SHIFT 5 +#define MAX98926_SPRCURNT_STATE_WIDTH 1 +#define MAX98926_WATCHFAIL_STATE_MASK (1<<4) +#define MAX98926_WATCHFAIL_STATE_SHIFT 4 +#define MAX98926_WATCHFAIL_STATE_WIDTH 1 +#define MAX98926_ALCINFH_STATE_MASK (1<<3) +#define MAX98926_ALCINFH_STATE_SHIFT 3 +#define MAX98926_ALCINFH_STATE_WIDTH 1 +#define MAX98926_ALCACT_STATE_MASK (1<<2) +#define MAX98926_ALCACT_STATE_SHIFT 2 +#define MAX98926_ALCACT_STATE_WIDTH 1 +#define MAX98926_ALCMUT_STATE_MASK (1<<1) +#define MAX98926_ALCMUT_STATE_SHIFT 1 +#define MAX98926_ALCMUT_STATE_WIDTH 1 +#define MAX98926_ALCP_STATE_MASK (1<<0) +#define MAX98926_ALCP_STATE_SHIFT 0 +#define MAX98926_ALCP_STATE_WIDTH 1 + +/* MAX98926_R007_STATE2 */ +#define MAX98926_SLOTOVRN_STATE_MASK (1<<6) +#define MAX98926_SLOTOVRN_STATE_SHIFT 6 +#define MAX98926_SLOTOVRN_STATE_WIDTH 1 +#define MAX98926_INVALSLOT_STATE_MASK (1<<5) +#define MAX98926_INVALSLOT_STATE_SHIFT 5 +#define MAX98926_INVALSLOT_STATE_WIDTH 1 +#define MAX98926_SLOTCNFLT_STATE_MASK (1<<4) +#define MAX98926_SLOTCNFLT_STATE_SHIFT 4 +#define MAX98926_SLOTCNFLT_STATE_WIDTH 1 +#define MAX98926_VBSTOVFL_STATE_MASK (1<<3) +#define MAX98926_VBSTOVFL_STATE_SHIFT 3 +#define MAX98926_VBSTOVFL_STATE_WIDTH 1 +#define MAX98926_VBATOVFL_STATE_MASK (1<<2) +#define MAX98926_VBATOVFL_STATE_SHIFT 2 +#define MAX98926_VBATOVFL_STATE_WIDTH 1 +#define MAX98926_IMONOVFL_STATE_MASK (1<<1) +#define MAX98926_IMONOVFL_STATE_SHIFT 1 +#define MAX98926_IMONOVFL_STATE_WIDTH 1 +#define MAX98926_VMONOVFL_STATE_MASK (1<<0) +#define MAX98926_VMONOVFL_STATE_SHIFT 0 +#define MAX98926_VMONOVFL_STATE_WIDTH 1 + +/* MAX98926_R008_FLAG0 */ +#define MAX98926_THERMWARN_END_FLAG_MASK (1<<3) +#define MAX98926_THERMWARN_END_FLAG_SHIFT 3 +#define MAX98926_THERMWARN_END_FLAG_WIDTH 1 +#define MAX98926_THERMWARN_BGN_FLAG_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_FLAG_SHIFT 2 +#define MAX98926_THERMWARN_BGN_FLAG_WIDTH 1 +#define MAX98926_THERMSHDN_END_FLAG_MASK (1<<1) +#define MAX98926_THERMSHDN_END_FLAG_SHIFT 1 +#define MAX98926_THERMSHDN_END_FLAG_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_FLAG_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98926_R009_FLAG1 */ +#define MAX98926_SPKCURNT_FLAG_MASK (1<<5) +#define MAX98926_SPKCURNT_FLAG_SHIFT 5 +#define MAX98926_SPKCURNT_FLAG_WIDTH 1 +#define MAX98926_WATCHFAIL_FLAG_MASK (1<<4) +#define MAX98926_WATCHFAIL_FLAG_SHIFT 4 +#define MAX98926_WATCHFAIL_FLAG_WIDTH 1 +#define MAX98926_ALCINFH_FLAG_MASK (1<<3) +#define MAX98926_ALCINFH_FLAG_SHIFT 3 +#define MAX98926_ALCINFH_FLAG_WIDTH 1 +#define MAX98926_ALCACT_FLAG_MASK (1<<2) +#define MAX98926_ALCACT_FLAG_SHIFT 2 +#define MAX98926_ALCACT_FLAG_WIDTH 1 +#define MAX98926_ALCMUT_FLAG_MASK (1<<1) +#define MAX98926_ALCMUT_FLAG_SHIFT 1 +#define MAX98926_ALCMUT_FLAG_WIDTH 1 +#define MAX98926_ALCP_FLAG_MASK (1<<0) +#define MAX98926_ALCP_FLAG_SHIFT 0 +#define MAX98926_ALCP_FLAG_WIDTH 1 + +/* MAX98926_R00A_FLAG2 */ +#define MAX98926_SLOTOVRN_FLAG_MASK (1<<6) +#define MAX98926_SLOTOVRN_FLAG_SHIFT 6 +#define MAX98926_SLOTOVRN_FLAG_WIDTH 1 +#define MAX98926_INVALSLOT_FLAG_MASK (1<<5) +#define MAX98926_INVALSLOT_FLAG_SHIFT 5 +#define MAX98926_INVALSLOT_FLAG_WIDTH 1 +#define MAX98926_SLOTCNFLT_FLAG_MASK (1<<4) +#define MAX98926_SLOTCNFLT_FLAG_SHIFT 4 +#define MAX98926_SLOTCNFLT_FLAG_WIDTH 1 +#define MAX98926_VBSTOVFL_FLAG_MASK (1<<3) +#define MAX98926_VBSTOVFL_FLAG_SHIFT 3 +#define MAX98926_VBSTOVFL_FLAG_WIDTH 1 +#define MAX98926_VBATOVFL_FLAG_MASK (1<<2) +#define MAX98926_VBATOVFL_FLAG_SHIFT 2 +#define MAX98926_VBATOVFL_FLAG_WIDTH 1 +#define MAX98926_IMONOVFL_FLAG_MASK (1<<1) +#define MAX98926_IMONOVFL_FLAG_SHIFT 1 +#define MAX98926_IMONOVFL_FLAG_WIDTH 1 +#define MAX98926_VMONOVFL_FLAG_MASK (1<<0) +#define MAX98926_VMONOVFL_FLAG_SHIFT 0 +#define MAX98926_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98926_R00B_IRQ_ENABLE0 */ +#define MAX98926_THERMWARN_END_EN_MASK (1<<3) +#define MAX98926_THERMWARN_END_EN_SHIFT 3 +#define MAX98926_THERMWARN_END_EN_WIDTH 1 +#define MAX98926_THERMWARN_BGN_EN_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_EN_SHIFT 2 +#define MAX98926_THERMWARN_BGN_EN_WIDTH 1 +#define MAX98926_THERMSHDN_END_EN_MASK (1<<1) +#define MAX98926_THERMSHDN_END_EN_SHIFT 1 +#define MAX98926_THERMSHDN_END_EN_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_EN_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_EN_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98926_R00C_IRQ_ENABLE1 */ +#define MAX98926_SPKCURNT_EN_MASK (1<<5) +#define MAX98926_SPKCURNT_EN_SHIFT 5 +#define MAX98926_SPKCURNT_EN_WIDTH 1 +#define MAX98926_WATCHFAIL_EN_MASK (1<<4) +#define MAX98926_WATCHFAIL_EN_SHIFT 4 +#define MAX98926_WATCHFAIL_EN_WIDTH 1 +#define MAX98926_ALCINFH_EN_MASK (1<<3) +#define MAX98926_ALCINFH_EN_SHIFT 3 +#define MAX98926_ALCINFH_EN_WIDTH 1 +#define MAX98926_ALCACT_EN_MASK (1<<2) +#define MAX98926_ALCACT_EN_SHIFT 2 +#define MAX98926_ALCACT_EN_WIDTH 1 +#define MAX98926_ALCMUT_EN_MASK (1<<1) +#define MAX98926_ALCMUT_EN_SHIFT 1 +#define MAX98926_ALCMUT_EN_WIDTH 1 +#define MAX98926_ALCP_EN_MASK (1<<0) +#define MAX98926_ALCP_EN_SHIFT 0 +#define MAX98926_ALCP_EN_WIDTH 1 + +/* MAX98926_R00D_IRQ_ENABLE2 */ +#define MAX98926_SLOTOVRN_EN_MASK (1<<6) +#define MAX98926_SLOTOVRN_EN_SHIFT 6 +#define MAX98926_SLOTOVRN_EN_WIDTH 1 +#define MAX98926_INVALSLOT_EN_MASK (1<<5) +#define MAX98926_INVALSLOT_EN_SHIFT 5 +#define MAX98926_INVALSLOT_EN_WIDTH 1 +#define MAX98926_SLOTCNFLT_EN_MASK (1<<4) +#define MAX98926_SLOTCNFLT_EN_SHIFT 4 +#define MAX98926_SLOTCNFLT_EN_WIDTH 1 +#define MAX98926_VBSTOVFL_EN_MASK (1<<3) +#define MAX98926_VBSTOVFL_EN_SHIFT 3 +#define MAX98926_VBSTOVFL_EN_WIDTH 1 +#define MAX98926_VBATOVFL_EN_MASK (1<<2) +#define MAX98926_VBATOVFL_EN_SHIFT 2 +#define MAX98926_VBATOVFL_EN_WIDTH 1 +#define MAX98926_IMONOVFL_EN_MASK (1<<1) +#define MAX98926_IMONOVFL_EN_SHIFT 1 +#define MAX98926_IMONOVFL_EN_WIDTH 1 +#define MAX98926_VMONOVFL_EN_MASK (1<<0) +#define MAX98926_VMONOVFL_EN_SHIFT 0 +#define MAX98926_VMONOVFL_EN_WIDTH 1 + +/* MAX98926_R00E_IRQ_CLEAR0 */ +#define MAX98926_THERMWARN_END_CLR_MASK (1<<3) +#define MAX98926_THERMWARN_END_CLR_SHIFT 3 +#define MAX98926_THERMWARN_END_CLR_WIDTH 1 +#define MAX98926_THERMWARN_BGN_CLR_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_CLR_SHIFT 2 +#define MAX98926_THERMWARN_BGN_CLR_WIDTH 1 +#define MAX98926_THERMSHDN_END_CLR_MASK (1<<1) +#define MAX98926_THERMSHDN_END_CLR_SHIFT 1 +#define MAX98926_THERMSHDN_END_CLR_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_CLR_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_CLR_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98926_R00F_IRQ_CLEAR1 */ +#define MAX98926_SPKCURNT_CLR_MASK (1<<5) +#define MAX98926_SPKCURNT_CLR_SHIFT 5 +#define MAX98926_SPKCURNT_CLR_WIDTH 1 +#define MAX98926_WATCHFAIL_CLR_MASK (1<<4) +#define MAX98926_WATCHFAIL_CLR_SHIFT 4 +#define MAX98926_WATCHFAIL_CLR_WIDTH 1 +#define MAX98926_ALCINFH_CLR_MASK (1<<3) +#define MAX98926_ALCINFH_CLR_SHIFT 3 +#define MAX98926_ALCINFH_CLR_WIDTH 1 +#define MAX98926_ALCACT_CLR_MASK (1<<2) +#define MAX98926_ALCACT_CLR_SHIFT 2 +#define MAX98926_ALCACT_CLR_WIDTH 1 +#define MAX98926_ALCMUT_CLR_MASK (1<<1) +#define MAX98926_ALCMUT_CLR_SHIFT 1 +#define MAX98926_ALCMUT_CLR_WIDTH 1 +#define MAX98926_ALCP_CLR_MASK (1<<0) +#define MAX98926_ALCP_CLR_SHIFT 0 +#define MAX98926_ALCP_CLR_WIDTH 1 + +/* MAX98926_R010_IRQ_CLEAR2 */ +#define MAX98926_SLOTOVRN_CLR_MASK (1<<6) +#define MAX98926_SLOTOVRN_CLR_SHIFT 6 +#define MAX98926_SLOTOVRN_CLR_WIDTH 1 +#define MAX98926_INVALSLOT_CLR_MASK (1<<5) +#define MAX98926_INVALSLOT_CLR_SHIFT 5 +#define MAX98926_INVALSLOT_CLR_WIDTH 1 +#define MAX98926_SLOTCNFLT_CLR_MASK (1<<4) +#define MAX98926_SLOTCNFLT_CLR_SHIFT 4 +#define MAX98926_SLOTCNFLT_CLR_WIDTH 1 +#define MAX98926_VBSTOVFL_CLR_MASK (1<<3) +#define MAX98926_VBSTOVFL_CLR_SHIFT 3 +#define MAX98926_VBSTOVFL_CLR_WIDTH 1 +#define MAX98926_VBATOVFL_CLR_MASK (1<<2) +#define MAX98926_VBATOVFL_CLR_SHIFT 2 +#define MAX98926_VBATOVFL_CLR_WIDTH 1 +#define MAX98926_IMONOVFL_CLR_MASK (1<<1) +#define MAX98926_IMONOVFL_CLR_SHIFT 1 +#define MAX98926_IMONOVFL_CLR_WIDTH 1 +#define MAX98926_VMONOVFL_CLR_MASK (1<<0) +#define MAX98926_VMONOVFL_CLR_SHIFT 0 +#define MAX98926_VMONOVFL_CLR_WIDTH 1 + +/* MAX98926_R011_MAP0 */ +#define MAX98926_ER_THERMWARN_EN_MASK (1<<7) +#define MAX98926_ER_THERMWARN_EN_SHIFT 7 +#define MAX98926_ER_THERMWARN_EN_WIDTH 1 +#define MAX98926_ER_THERMWARN_MAP_MASK (0x07<<4) +#define MAX98926_ER_THERMWARN_MAP_SHIFT 4 +#define MAX98926_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98926_R012_MAP1 */ +#define MAX98926_ER_ALCMUT_EN_MASK (1<<7) +#define MAX98926_ER_ALCMUT_EN_SHIFT 7 +#define MAX98926_ER_ALCMUT_EN_WIDTH 1 +#define MAX98926_ER_ALCMUT_MAP_MASK (0x07<<4) +#define MAX98926_ER_ALCMUT_MAP_SHIFT 4 +#define MAX98926_ER_ALCMUT_MAP_WIDTH 3 +#define MAX98926_ER_ALCP_EN_MASK (1<<3) +#define MAX98926_ER_ALCP_EN_SHIFT 3 +#define MAX98926_ER_ALCP_EN_WIDTH 1 +#define MAX98926_ER_ALCP_MAP_MASK (0x07<<0) +#define MAX98926_ER_ALCP_MAP_SHIFT 0 +#define MAX98926_ER_ALCP_MAP_WIDTH 3 + +/* MAX98926_R013_MAP2 */ +#define MAX98926_ER_ALCINFH_EN_MASK (1<<7) +#define MAX98926_ER_ALCINFH_EN_SHIFT 7 +#define MAX98926_ER_ALCINFH_EN_WIDTH 1 +#define MAX98926_ER_ALCINFH_MAP_MASK (0x07<<4) +#define MAX98926_ER_ALCINFH_MAP_SHIFT 4 +#define MAX98926_ER_ALCINFH_MAP_WIDTH 3 +#define MAX98926_ER_ALCACT_EN_MASK (1<<3) +#define MAX98926_ER_ALCACT_EN_SHIFT 3 +#define MAX98926_ER_ALCACT_EN_WIDTH 1 +#define MAX98926_ER_ALCACT_MAP_MASK (0x07<<0) +#define MAX98926_ER_ALCACT_MAP_SHIFT 0 +#define MAX98926_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98926_R014_MAP3 */ +#define MAX98926_ER_SPKCURNT_EN_MASK (1<<7) +#define MAX98926_ER_SPKCURNT_EN_SHIFT 7 +#define MAX98926_ER_SPKCURNT_EN_WIDTH 1 +#define MAX98926_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define MAX98926_ER_SPKCURNT_MAP_SHIFT 4 +#define MAX98926_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98926_R015_MAP4 */ +/* RESERVED */ + +/* MAX98926_R016_MAP5 */ +#define MAX98926_ER_IMONOVFL_EN_MASK (1<<7) +#define MAX98926_ER_IMONOVFL_EN_SHIFT 7 +#define MAX98926_ER_IMONOVFL_EN_WIDTH 1 +#define MAX98926_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define MAX98926_ER_IMONOVFL_MAP_SHIFT 4 +#define MAX98926_ER_IMONOVFL_MAP_WIDTH 3 +#define MAX98926_ER_VMONOVFL_EN_MASK (1<<3) +#define MAX98926_ER_VMONOVFL_EN_SHIFT 3 +#define MAX98926_ER_VMONOVFL_EN_WIDTH 1 +#define MAX98926_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define MAX98926_ER_VMONOVFL_MAP_SHIFT 0 +#define MAX98926_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98926_R017_MAP6 */ +#define MAX98926_ER_VBSTOVFL_EN_MASK (1<<7) +#define MAX98926_ER_VBSTOVFL_EN_SHIFT 7 +#define MAX98926_ER_VBSTOVFL_EN_WIDTH 1 +#define MAX98926_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define MAX98926_ER_VBSTOVFL_MAP_SHIFT 4 +#define MAX98926_ER_VBSTOVFL_MAP_WIDTH 3 +#define MAX98926_ER_VBATOVFL_EN_MASK (1<<3) +#define MAX98926_ER_VBATOVFL_EN_SHIFT 3 +#define MAX98926_ER_VBATOVFL_EN_WIDTH 1 +#define MAX98926_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define MAX98926_ER_VBATOVFL_MAP_SHIFT 0 +#define MAX98926_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98926_R018_MAP7 */ +#define MAX98926_ER_INVALSLOT_EN_MASK (1<<7) +#define MAX98926_ER_INVALSLOT_EN_SHIFT 7 +#define MAX98926_ER_INVALSLOT_EN_WIDTH 1 +#define MAX98926_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define MAX98926_ER_INVALSLOT_MAP_SHIFT 4 +#define MAX98926_ER_INVALSLOT_MAP_WIDTH 3 +#define MAX98926_ER_SLOTCNFLT_EN_MASK (1<<3) +#define MAX98926_ER_SLOTCNFLT_EN_SHIFT 3 +#define MAX98926_ER_SLOTCNFLT_EN_WIDTH 1 +#define MAX98926_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define MAX98926_ER_SLOTCNFLT_MAP_SHIFT 0 +#define MAX98926_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98926_R019_MAP8 */ +#define MAX98926_ER_SLOTOVRN_EN_MASK (1<<3) +#define MAX98926_ER_SLOTOVRN_EN_SHIFT 3 +#define MAX98926_ER_SLOTOVRN_EN_WIDTH 1 +#define MAX98926_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define MAX98926_ER_SLOTOVRN_MAP_SHIFT 0 +#define MAX98926_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98926_R01A_DAI_CLK_MODE1 */ +#define MAX98926_DAI_CLK_SOURCE_MASK (1<<6) +#define MAX98926_DAI_CLK_SOURCE_SHIFT 6 +#define MAX98926_DAI_CLK_SOURCE_WIDTH 1 +#define MAX98926_MDLL_MULT_MASK (0x0F<<0) +#define MAX98926_MDLL_MULT_SHIFT 0 +#define MAX98926_MDLL_MULT_WIDTH 4 + +#define MAX98926_MDLL_MULT_MCLKx8 6 +#define MAX98926_MDLL_MULT_MCLKx16 8 + +/* MAX98926_R01B_DAI_CLK_MODE2 */ +#define MAX98926_DAI_SR_MASK (0x0F<<4) +#define MAX98926_DAI_SR_SHIFT 4 +#define MAX98926_DAI_SR_WIDTH 4 +#define MAX98926_DAI_MAS_MASK (1<<3) +#define MAX98926_DAI_MAS_SHIFT 3 +#define MAX98926_DAI_MAS_WIDTH 1 +#define MAX98926_DAI_BSEL_MASK (0x07<<0) +#define MAX98926_DAI_BSEL_SHIFT 0 +#define MAX98926_DAI_BSEL_WIDTH 3 + +#define MAX98926_DAI_BSEL_32 (0 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_48 (1 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_64 (2 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_256 (6 << MAX98926_DAI_BSEL_SHIFT) + +/* MAX98926_R01C_DAI_CLK_DIV_M_MSBS */ +#define MAX98926_DAI_M_MSBS_MASK (0xFF<<0) +#define MAX98926_DAI_M_MSBS_SHIFT 0 +#define MAX98926_DAI_M_MSBS_WIDTH 8 + +/* MAX98926_R01D_DAI_CLK_DIV_M_LSBS */ +#define MAX98926_DAI_M_LSBS_MASK (0xFF<<0) +#define MAX98926_DAI_M_LSBS_SHIFT 0 +#define MAX98926_DAI_M_LSBS_WIDTH 8 + +/* MAX98926_R01E_DAI_CLK_DIV_N_MSBS */ +#define MAX98926_DAI_N_MSBS_MASK (0x7F<<0) +#define MAX98926_DAI_N_MSBS_SHIFT 0 +#define MAX98926_DAI_N_MSBS_WIDTH 7 + +/* MAX98926_R01F_DAI_CLK_DIV_N_LSBS */ +#define MAX98926_DAI_N_LSBS_MASK (0xFF<<0) +#define MAX98926_DAI_N_LSBS_SHIFT 0 +#define MAX98926_DAI_N_LSBS_WIDTH 8 + +/* MAX98926_R020_FORMAT */ +#define MAX98926_DAI_CHANSZ_MASK (0x03<<6) +#define MAX98926_DAI_CHANSZ_SHIFT 6 +#define MAX98926_DAI_CHANSZ_WIDTH 2 +#define MAX98926_DAI_INTERLEAVE_MASK (1<<5) +#define MAX98926_DAI_INTERLEAVE_SHIFT 5 +#define MAX98926_DAI_INTERLEAVE_WIDTH 1 +#define MAX98926_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define MAX98926_DAI_EXTBCLK_HIZ_SHIFT 4 +#define MAX98926_DAI_EXTBCLK_HIZ_WIDTH 1 +#define MAX98926_DAI_WCI_MASK (1<<3) +#define MAX98926_DAI_WCI_SHIFT 3 +#define MAX98926_DAI_WCI_WIDTH 1 +#define MAX98926_DAI_BCI_MASK (1<<2) +#define MAX98926_DAI_BCI_SHIFT 2 +#define MAX98926_DAI_BCI_WIDTH 1 +#define MAX98926_DAI_DLY_MASK (1<<1) +#define MAX98926_DAI_DLY_SHIFT 1 +#define MAX98926_DAI_DLY_WIDTH 1 +#define MAX98926_DAI_TDM_MASK (1<<0) +#define MAX98926_DAI_TDM_SHIFT 0 +#define MAX98926_DAI_TDM_WIDTH 1 + +#define MAX98926_DAI_CHANSZ_16 (1 << MAX98926_DAI_CHANSZ_SHIFT) +#define MAX98926_DAI_CHANSZ_24 (2 << MAX98926_DAI_CHANSZ_SHIFT) +#define MAX98926_DAI_CHANSZ_32 (3 << MAX98926_DAI_CHANSZ_SHIFT) + +/* MAX98926_R021_TDM_SLOT_SELECT */ +#define MAX98926_DAI_DO_EN_MASK (1<<7) +#define MAX98926_DAI_DO_EN_SHIFT 7 +#define MAX98926_DAI_DO_EN_WIDTH 1 +#define MAX98926_DAI_DIN_EN_MASK (1<<6) +#define MAX98926_DAI_DIN_EN_SHIFT 6 +#define MAX98926_DAI_DIN_EN_WIDTH 1 +#define MAX98926_DAI_INR_SOURCE_MASK (0x07<<3) +#define MAX98926_DAI_INR_SOURCE_SHIFT 3 +#define MAX98926_DAI_INR_SOURCE_WIDTH 3 +#define MAX98926_DAI_INL_SOURCE_MASK (0x07<<0) +#define MAX98926_DAI_INL_SOURCE_SHIFT 0 +#define MAX98926_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98926_R022_DOUT_CFG_VMON */ +#define MAX98926_DAI_VMON_EN_MASK (1<<5) +#define MAX98926_DAI_VMON_EN_SHIFT 5 +#define MAX98926_DAI_VMON_EN_WIDTH 1 +#define MAX98926_DAI_VMON_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_VMON_SLOT_SHIFT 0 +#define MAX98926_DAI_VMON_SLOT_WIDTH 5 + +#define MAX98926_DAI_VMON_SLOT_00_01 (0 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_01_02 (1 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_02_03 (2 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_03_04 (3 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_04_05 (4 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_05_06 (5 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_06_07 (6 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_07_08 (7 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_08_09 (8 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_09_0A (9 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0A_0B (10 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0B_0C (11 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0C_0D (12 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0D_0E (13 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0E_0F (14 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0F_10 (15 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_10_11 (16 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_11_12 (17 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_12_13 (18 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_13_14 (19 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_14_15 (20 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_15_16 (21 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_16_17 (22 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_17_18 (23 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_18_19 (24 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_19_1A (25 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1A_1B (26 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1B_1C (27 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1C_1D (28 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1D_1E (29 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1E_1F (30 << MAX98926_DAI_VMON_SLOT_SHIFT) + +/* MAX98926_R023_DOUT_CFG_IMON */ +#define MAX98926_DAI_IMON_EN_MASK (1<<5) +#define MAX98926_DAI_IMON_EN_SHIFT 5 +#define MAX98926_DAI_IMON_EN_WIDTH 1 +#define MAX98926_DAI_IMON_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_IMON_SLOT_SHIFT 0 +#define MAX98926_DAI_IMON_SLOT_WIDTH 5 + +#define MAX98926_DAI_IMON_SLOT_00_01 (0 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_01_02 (1 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_02_03 (2 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_03_04 (3 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_04_05 (4 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_05_06 (5 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_06_07 (6 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_07_08 (7 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_08_09 (8 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_09_0A (9 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0A_0B (10 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0B_0C (11 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0C_0D (12 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0D_0E (13 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0E_0F (14 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0F_10 (15 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_10_11 (16 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_11_12 (17 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_12_13 (18 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_13_14 (19 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_14_15 (20 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_15_16 (21 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_16_17 (22 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_17_18 (23 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_18_19 (24 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_19_1A (25 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1A_1B (26 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1B_1C (27 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1C_1D (28 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1D_1E (29 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1E_1F (30 << MAX98926_DAI_IMON_SLOT_SHIFT) + +/* MAX98926_R024_DOUT_CFG_VBAT */ +#define MAX98926_DAI_INTERLEAVE_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_INTERLEAVE_SLOT_SHIFT 0 +#define MAX98926_DAI_INTERLEAVE_SLOT_WIDTH 5 + +/* MAX98926_R025_DOUT_CFG_VBST */ +#define MAX98926_DAI_VBST_EN_MASK (1<<5) +#define MAX98926_DAI_VBST_EN_SHIFT 5 +#define MAX98926_DAI_VBST_EN_WIDTH 1 +#define MAX98926_DAI_VBST_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_VBST_SLOT_SHIFT 0 +#define MAX98926_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98926_R026_DOUT_CFG_FLAG */ +#define MAX98926_DAI_FLAG_EN_MASK (1<<5) +#define MAX98926_DAI_FLAG_EN_SHIFT 5 +#define MAX98926_DAI_FLAG_EN_WIDTH 1 +#define MAX98926_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_FLAG_SLOT_SHIFT 0 +#define MAX98926_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98926_R027_DOUT_HIZ_CFG1 */ +#define MAX98926_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98926_R028_DOUT_HIZ_CFG2 */ +#define MAX98926_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98926_R029_DOUT_HIZ_CFG3 */ +#define MAX98926_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98926_R02A_DOUT_HIZ_CFG4 */ +#define MAX98926_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98926_R02B_DOUT_DRV_STRENGTH */ +#define MAX98926_DAI_OUT_DRIVE_MASK (0x03<<0) +#define MAX98926_DAI_OUT_DRIVE_SHIFT 0 +#define MAX98926_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98926_R02C_FILTERS */ +#define MAX98926_ADC_DITHER_EN_MASK (1<<7) +#define MAX98926_ADC_DITHER_EN_SHIFT 7 +#define MAX98926_ADC_DITHER_EN_WIDTH 1 +#define MAX98926_IV_DCB_EN_MASK (1<<6) +#define MAX98926_IV_DCB_EN_SHIFT 6 +#define MAX98926_IV_DCB_EN_WIDTH 1 +#define MAX98926_DAC_DITHER_EN_MASK (1<<4) +#define MAX98926_DAC_DITHER_EN_SHIFT 4 +#define MAX98926_DAC_DITHER_EN_WIDTH 1 +#define MAX98926_DAC_FILTER_MODE_MASK (1<<3) +#define MAX98926_DAC_FILTER_MODE_SHIFT 3 +#define MAX98926_DAC_FILTER_MODE_WIDTH 1 +#define MAX98926_DAC_HPF_MASK (0x07<<0) +#define MAX98926_DAC_HPF_SHIFT 0 +#define MAX98926_DAC_HPF_WIDTH 3 +#define MAX98926_DAC_HPF_DISABLE (0 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_DC_BLOCK (1 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_100 (2 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_200 (3 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_400 (4 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_800 (5 << MAX98926_DAC_HPF_SHIFT) + +/* MAX98926_R02D_GAIN */ +#define MAX98926_DAC_IN_SEL_MASK (0x03<<5) +#define MAX98926_DAC_IN_SEL_SHIFT 5 +#define MAX98926_DAC_IN_SEL_WIDTH 2 +#define MAX98926_SPK_GAIN_MASK (0x1F<<0) +#define MAX98926_SPK_GAIN_SHIFT 0 +#define MAX98926_SPK_GAIN_WIDTH 5 + +#define MAX98926_DAC_IN_SEL_LEFT_DAI (0 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_RIGHT_DAI (1 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_SUMMED_DAI (2 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << MAX98926_DAC_IN_SEL_SHIFT) + +/* MAX98926_R02E_GAIN_RAMPING */ +#define MAX98926_SPK_RMP_EN_MASK (1<<1) +#define MAX98926_SPK_RMP_EN_SHIFT 1 +#define MAX98926_SPK_RMP_EN_WIDTH 1 +#define MAX98926_SPK_ZCD_EN_MASK (1<<0) +#define MAX98926_SPK_ZCD_EN_SHIFT 0 +#define MAX98926_SPK_ZCD_EN_WIDTH 1 + +/* MAX98926_R02F_SPK_AMP */ +#define MAX98926_SPK_MODE_MASK (1<<0) +#define MAX98926_SPK_MODE_SHIFT 0 +#define MAX98926_SPK_MODE_WIDTH 1 +#define MAX98926_INSELECT_MODE_MASK (1<<1) +#define MAX98926_INSELECT_MODE_SHIFT 1 +#define MAX98926_INSELECT_MODE_WIDTH 1 + +/* MAX98926_R030_THRESHOLD */ +#define MAX98926_ALC_EN_MASK (1<<5) +#define MAX98926_ALC_EN_SHIFT 5 +#define MAX98926_ALC_EN_WIDTH 1 +#define MAX98926_ALC_TH_MASK (0x1F<<0) +#define MAX98926_ALC_TH_SHIFT 0 +#define MAX98926_ALC_TH_WIDTH 5 + +/* MAX98926_R031_ALC_ATTACK */ +#define MAX98926_ALC_ATK_STEP_MASK (0x0F<<4) +#define MAX98926_ALC_ATK_STEP_SHIFT 4 +#define MAX98926_ALC_ATK_STEP_WIDTH 4 +#define MAX98926_ALC_ATK_RATE_MASK (0x7<<0) +#define MAX98926_ALC_ATK_RATE_SHIFT 0 +#define MAX98926_ALC_ATK_RATE_WIDTH 3 + +/* MAX98926_R032_ALC_ATTEN_RLS */ +#define MAX98926_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define MAX98926_ALC_MAX_ATTEN_SHIFT 4 +#define MAX98926_ALC_MAX_ATTEN_WIDTH 4 +#define MAX98926_ALC_RLS_RATE_MASK (0x7<<0) +#define MAX98926_ALC_RLS_RATE_SHIFT 0 +#define MAX98926_ALC_RLS_RATE_WIDTH 3 + +/* MAX98926_R033_ALC_HOLD_RLS */ +#define MAX98926_ALC_RLS_TGR_MASK (1<<0) +#define MAX98926_ALC_RLS_TGR_SHIFT 0 +#define MAX98926_ALC_RLS_TGR_WIDTH 1 + +/* MAX98926_R034_ALC_CONFIGURATION */ +#define MAX98926_ALC_MUTE_EN_MASK (1<<7) +#define MAX98926_ALC_MUTE_EN_SHIFT 7 +#define MAX98926_ALC_MUTE_EN_WIDTH 1 +#define MAX98926_ALC_MUTE_DLY_MASK (0x07<<4) +#define MAX98926_ALC_MUTE_DLY_SHIFT 4 +#define MAX98926_ALC_MUTE_DLY_WIDTH 3 +#define MAX98926_ALC_RLS_DBT_MASK (0x07<<0) +#define MAX98926_ALC_RLS_DBT_SHIFT 0 +#define MAX98926_ALC_RLS_DBT_WIDTH 3 + +/* MAX98926_R035_BOOST_CONVERTER */ +#define MAX98926_BST_SYNC_MASK (1<<7) +#define MAX98926_BST_SYNC_SHIFT 7 +#define MAX98926_BST_SYNC_WIDTH 1 +#define MAX98926_BST_PHASE_MASK (0x03<<4) +#define MAX98926_BST_PHASE_SHIFT 4 +#define MAX98926_BST_PHASE_WIDTH 2 +#define MAX98926_BST_SKIP_MODE_MASK (0x03<<0) +#define MAX98926_BST_SKIP_MODE_SHIFT 0 +#define MAX98926_BST_SKIP_MODE_WIDTH 2 + +/* MAX98926_R036_BLOCK_ENABLE */ +#define MAX98926_BST_EN_MASK (1<<7) +#define MAX98926_BST_EN_SHIFT 7 +#define MAX98926_BST_EN_WIDTH 1 +#define MAX98926_WATCH_EN_MASK (1<<6) +#define MAX98926_WATCH_EN_SHIFT 6 +#define MAX98926_WATCH_EN_WIDTH 1 +#define MAX98926_CLKMON_EN_MASK (1<<5) +#define MAX98926_CLKMON_EN_SHIFT 5 +#define MAX98926_CLKMON_EN_WIDTH 1 +#define MAX98926_SPK_EN_MASK (1<<4) +#define MAX98926_SPK_EN_SHIFT 4 +#define MAX98926_SPK_EN_WIDTH 1 +#define MAX98926_ADC_VBST_EN_MASK (1<<3) +#define MAX98926_ADC_VBST_EN_SHIFT 3 +#define MAX98926_ADC_VBST_EN_WIDTH 1 +#define MAX98926_ADC_VBAT_EN_MASK (1<<2) +#define MAX98926_ADC_VBAT_EN_SHIFT 2 +#define MAX98926_ADC_VBAT_EN_WIDTH 1 +#define MAX98926_ADC_IMON_EN_MASK (1<<1) +#define MAX98926_ADC_IMON_EN_SHIFT 1 +#define MAX98926_ADC_IMON_EN_WIDTH 1 +#define MAX98926_ADC_VMON_EN_MASK (1<<0) +#define MAX98926_ADC_VMON_EN_SHIFT 0 +#define MAX98926_ADC_VMON_EN_WIDTH 1 + +/* MAX98926_R037_CONFIGURATION */ +#define MAX98926_BST_VOUT_MASK (0x0F<<4) +#define MAX98926_BST_VOUT_SHIFT 4 +#define MAX98926_BST_VOUT_WIDTH 4 +#define MAX98926_THERMWARN_LEVEL_MASK (0x03<<2) +#define MAX98926_THERMWARN_LEVEL_SHIFT 2 +#define MAX98926_THERMWARN_LEVEL_WIDTH 2 +#define MAX98926_WATCH_TIME_MASK (0x03<<0) +#define MAX98926_WATCH_TIME_SHIFT 0 +#define MAX98926_WATCH_TIME_WIDTH 2 + +/* MAX98926_R038_GLOBAL_ENABLE */ +#define MAX98926_EN_MASK (1<<7) +#define MAX98926_EN_SHIFT 7 +#define MAX98926_EN_WIDTH 1 + +/* MAX98926_R03A_BOOST_LIMITER */ +#define MAX98926_BST_ILIM_MASK (0x1F<<3) +#define MAX98926_BST_ILIM_SHIFT 3 +#define MAX98926_BST_ILIM_WIDTH 5 + +/* MAX98926_R0FF_VERSION */ +#define MAX98926_REV_ID_MASK (0xFF<<0) +#define MAX98926_REV_ID_SHIFT 0 +#define MAX98926_REV_ID_WIDTH 8 + +struct max98926_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + unsigned int sysclk; + unsigned int v_slot; + unsigned int i_slot; + unsigned int ch_size; + unsigned int interleave_mode; +}; +#endif -- 1.9.3

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[alsa-devel] [PATCH V2 0/2] Add support for Pistachio internal DAC
by Damien Horsley 08 Dec '15

08 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 316 +++++++++++++++++++++ 4 files changed, 344 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

2 3

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused snd_hda_get_nid_path()
by Takashi Iwai 08 Dec '15

08 Dec '15

An exported helper function snd_hda_get_nid_path() is nowhere used. Let's remove it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 16 ---------------- sound/pci/hda/hda_generic.h | 2 -- 2 files changed, 18 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c6e8a651cea1..a644fc3302f0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -279,22 +279,6 @@ static struct nid_path *get_nid_path(struct hda_codec *codec, } /** - * snd_hda_get_nid_path - get the path between the given NIDs - * @codec: the HDA codec - * @from_nid: the NID where the path start from - * @to_nid: the NID where the path ends at - * - * Return the found nid_path object or NULL for error. - * Passing 0 to either @from_nid or @to_nid behaves as a wildcard. - */ -struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec, - hda_nid_t from_nid, hda_nid_t to_nid) -{ - return get_nid_path(codec, from_nid, to_nid, 0); -} -EXPORT_SYMBOL_GPL(snd_hda_get_nid_path); - -/** * snd_hda_get_path_idx - get the index number corresponding to the path * instance * @codec: the HDA codec diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139b9032..692510e59365 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -306,8 +306,6 @@ int snd_hda_gen_spec_init(struct hda_gen_spec *spec); int snd_hda_gen_init(struct hda_codec *codec); void snd_hda_gen_free(struct hda_codec *codec); -struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec, - hda_nid_t from_nid, hda_nid_t to_nid); int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path); struct nid_path *snd_hda_get_path_from_idx(struct hda_codec *codec, int idx); bool snd_hda_parse_nid_path(struct hda_codec *codec, hda_nid_t from_nid, -- 2.6.3

1 1

[alsa-devel] [PATCH v2] ALSA: hda - Fixing speaker noise on the two latest thinkpad models
by Hui Wang 08 Dec '15

08 Dec '15

We have two latest thinkpad laptop models which are all based on the Intel skylake platforms, and all of them have the codec alc293 on them. When the machines boot to the desktop, an greeting dialogue shows up with the notification sound. But on these two models, there is noise with the notification sound. We have 3 SKUs for each of the models, all of them have this problem. So far, this problem is only specific to these two thinkpad models, we did not find this problem on the old thinkpad models with the codec alc293 or alc292. A workaround for this problem is disabling the aamix. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523517 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ebc5362..49b1d36 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC293_FIXUP_LENOVO_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5187,6 +5188,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC293_FIXUP_LENOVO_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5334,6 +5341,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5343,6 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.9.1

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[alsa-devel] [PATCH V2 1/1] audio: Fix playback noise with 24/32 bit sample size on BXT
by han.lu＠intel.com 08 Dec '15

08 Dec '15

From: "Lu, Han" <han.lu(a)intel.com> In BXT-P A0, HD-Audio DMA requests is later than expected, and makes an audio stream sensitive to system latencies when 24/32 bits are playing. Adjusting threshold of DMA fifo to force the DMA request sooner to improve latency tolerance at the expense of power. v2: move Intel specific code to hda_intel.c Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 2ae8812..94dc6a9 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -93,6 +93,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_HSW_EM4 0x100c #define AZX_REG_HSW_EM5 0x1010 +/* Skylake/Broxton display HD-A controller Extended Mode registers */ +#define AZX_REG_SKL_EM4L 0x1040 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 963f824..bff5c8b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -355,6 +355,8 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) +#define IS_BROXTON(pci) ((pci)->device == 0x5a98) + static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", @@ -506,15 +508,36 @@ static void azx_init_pci(struct azx *chip) } } +/* + * In BXT-P A0, HD-Audio DMA requests is later than expected, + * and makes an audio stream sensitive to system latencies when + * 24/32 bits are playing. + * Adjusting threshold of DMA fifo to force the DMA request + * sooner to improve latency tolerance at the expense of power. + */ +static void bxt_reduce_dma_latency(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, SKL_EM4L); + val &= (0x3 << 20); + azx_writel(chip, SKL_EM4L, val); +} + static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); + struct pci_dev *pci = chip->pci; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, false); + + /* reduce dma latency to avoid noise */ + if (IS_BROXTON(pci)) + bxt_reduce_dma_latency(chip); } /* calculate runtime delay from LPIB */ -- 2.5.0

3 4

[alsa-devel] [PATCH] ALSA: hda - Fixing speaker noise on two thinkpad machines
by Hui Wang 08 Dec '15

08 Dec '15

Disable the aamix can fix this problem. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523517 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ebc5362..49b1d36 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC293_FIXUP_LENOVO_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5187,6 +5188,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC293_FIXUP_LENOVO_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5334,6 +5341,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5343,6 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.9.1

2 6

[alsa-devel] [PATCH] [RFC] ARM: shmobile: bockw dts: Override #sound-dai-cells to zero
by Geert Uytterhoeven 08 Dec '15

08 Dec '15

The default value of #sound-dai-cells in r8a7778.dtsi is one, while the /sound/simple-audio-card,cpu device node in r8a7778-bockw.dts uses a phandle without any extra cells ("<&rcar_sound>"), causing: /sound/simple-audio-card,cpu: arguments longer than property asoc-simple-card sound: parse error -22 asoc-simple-card: probe of sound failed with error -22 Override #sound-dai-cells to zero to fix this. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- The rcar_sound node is still disabled. So why is this needed?!? --- arch/arm/boot/dts/r8a7778-bockw.dts | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/arch/arm/boot/dts/r8a7778-bockw.dts b/arch/arm/boot/dts/r8a7778-bockw.dts index 3c03e6b8261e9348..158ad94fd4d187a7 100644 --- a/arch/arm/boot/dts/r8a7778-bockw.dts +++ b/arch/arm/boot/dts/r8a7778-bockw.dts @@ -172,6 +172,11 @@ }; }; +&rcar_sound { + /* Single DAI */ + #sound-dai-cells = <0>; +}; + &sdhi0 { pinctrl-0 = <&sdhi0_pins>, <&sdhi0_pup_pins>; pinctrl-names = "default"; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 MC Next Gen] sound/usb: Fix out of bounds access in media_entity_init()
by Shuah Khan 07 Dec '15

07 Dec '15

Fix the out of bounds access in media_entity_init() found by KASan. This is a result of media_mixer_init() failing to allocate memory for all 3 of its pads before calling media_entity_init(). Fix it to allocate memory for the right struct media_mixer_ctl instead of struct media_ctl. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- Changes since v1: Change to address review comment from Takashi Iwai This patch fixes the mixer patch below: https://patchwork.linuxtv.org/patch/31827/ sound/usb/media.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index bebe27b..b0d283f 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -233,8 +233,8 @@ int media_mixer_init(struct snd_usb_audio *chip) if (mixer->media_mixer_ctl) continue; - /* allocate media_ctl */ - mctl = kzalloc(sizeof(struct media_ctl), GFP_KERNEL); + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); if (!mctl) return -ENOMEM; -- 2.5.0

1 0

[alsa-devel] Sound UsbPre : minimal patch for support
by shal＠free.fr 07 Dec '15

07 Dec '15

Hello, I have a Sound UsbPre (the first not the UsbPRe2) sound card. There is no support for this card in alsa. I do a usb packet sniffing on Windows for detect problem. Some control quirk should be added at boot in order to use the card. It works to play and record in 44100 khz. It works only with 44100khz (for example, if you play a 88200 file, it will be played too slow). This patch is not well tested, work on my desktop : Linux Ubuntu with 4.2.19-lowlatency kernel. The button emulation is not done (by a control packet, you can change the card configuration, like phantom power activation). If somebody is interested, I can provide needed information. If Alsa's developer is interested, I can provide support for test patches or develop a second version of this patch in order to integrate the support for this card in the linux kernel Regards, Olivier linux-source-4.2.0/sound/usb$ diff -rub quirks.c.ORIG quirks.c --- quirks.c.ORIG 2015-11-24 22:43:16.878635155 +0100 +++ quirks.c 2015-11-26 20:56:09.492126460 +0100 @@ -601,6 +601,40 @@ return 0; } +/* Add some control messages needed by Sound Devices UsbPre. + * Values here are chosen based on sniffing USB traffic + * under Windows.*/ +static int snd_usb_usbpre_boot_quirk(struct usb_device *dev) +{ + u8 buf = 1; + u8 data1[1] = {0x81}; + u8 data2[1] = {0xc1}; + u8 data3[3] = {0x44,0xac,0x00}; + + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 496, data1, 1); + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x81 , + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 0, &buf, 1); + if (buf == 0) { + return -ENODEV; + } + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 496, data2, 1); + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x81 , + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x00f0, 0, &buf, 1); + if (buf == 0) { + return -ENODEV; + } + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_ENDPOINT, + 0x0100, 133, data3, 3); + return 0; +} + static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) { int err; @@ -985,6 +1019,10 @@ /* SB Audigy 2 NX needs its own boot-up magic, too */ return snd_usb_audigy2nx_boot_quirk(dev); + case USB_ID(0x0926, 0x0100): + /* Sound Devices UsbPre needs its own boot-up magic, too */ + return snd_usb_usbpre_boot_quirk(dev); + case USB_ID(0x10f5, 0x0200): /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */ return snd_usb_cm106_boot_quirk(dev);

2 1

[alsa-devel] [PATCH MC Next Gen] sound/usb: Fix out of bounds access in media_entity_init()
by Shuah Khan 07 Dec '15

07 Dec '15

Fix the out of bounds access in media_entity_init() found by KASan. This is a result of media_mixer_init() failing to allocate memory for all 3 of its pads before calling media_entity_init(). Fix it to allocate memory for the right struct media_mixer_ctl instead of struct media_ctl. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- This patch fixes the mixer patch below: https://patchwork.linuxtv.org/patch/31827/ sound/usb/media.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index bebe27b..0cb44b9 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -233,8 +233,8 @@ int media_mixer_init(struct snd_usb_audio *chip) if (mixer->media_mixer_ctl) continue; - /* allocate media_ctl */ - mctl = kzalloc(sizeof(struct media_ctl), GFP_KERNEL); + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(struct media_mixer_ctl), GFP_KERNEL); if (!mctl) return -ENOMEM; @@ -244,6 +244,7 @@ int media_mixer_init(struct snd_usb_audio *chip) mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, mctl->media_pad); ret = media_device_register_entity(mctl->media_dev, -- 2.5.0

2 2

[alsa-devel] [GIT PULL] AC'97 gpiochip
by Mark Brown 07 Dec '15

07 Dec '15

The following changes since commit 8005c49d9aea74d382f474ce11afbbc7d7130bec: Linux 4.4-rc1 (2015-11-15 17:00:27 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-ac97-gpio for you to fetch changes up to 9bf5c3d11f1fbaf43399d189f05fb20ceb46ee5d: ASoC: ac97: add gpio chip (2015-11-18 18:08:54 +0000) ---------------------------------------------------------------- ASoC: Add a GPIO chip for AC'97 GPIOs are part of the AC'97 spec, enable their use on embedded platforms using AC'97. ---------------------------------------------------------------- Robert Jarzmik (1): ASoC: ac97: add gpio chip include/sound/ac97_codec.h | 3 ++ sound/soc/soc-ac97.c | 125 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 128 insertions(+)

2 1

[alsa-devel] [PATCH] ALSA: hda - Add inverted dmic for Packard Bell DOTS
by David Henningsson 07 Dec '15

07 Dec '15

On the internal mic of the Packard Bell DOTS, one channel has an inverted signal. Add a quirk to fix this up. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523232 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c..ebc5362 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6409,6 +6409,7 @@ static const struct hda_fixup alc662_fixups[] = { static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ALSA: compress: add support for 32bit calls in a 64bit kernel
by Ravindra Lokhande 07 Dec '15

07 Dec '15

Compress offload does not support ioctl calls from a 32bit userspace in a 64 bit kernel. This patch adds support for ioctls from a 32bit userspace in a 64bit kernel Signed-off-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- v1 -> v2: use compat_ptr to convert 32 bit address to 64 bit pointer v2 -> v3: add header file compat.h sound/core/compress_offload.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b123c42..4597820 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -38,6 +38,7 @@ #include <linux/uio.h> #include <linux/uaccess.h> #include <linux/module.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/compress_params.h> @@ -847,6 +848,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) return retval; } +/* support of 32bit userspace on 64bit platforms */ +#ifdef CONFIG_COMPAT +static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_compr_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} +#endif + static const struct file_operations snd_compr_file_ops = { .owner = THIS_MODULE, .open = snd_compr_open, @@ -854,6 +864,9 @@ static const struct file_operations snd_compr_file_ops = { .write = snd_compr_write, .read = snd_compr_read, .unlocked_ioctl = snd_compr_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = snd_compr_ioctl_compat, +#endif .mmap = snd_compr_mmap, .poll = snd_compr_poll, }; -- 2.1.4

3 3

[alsa-devel] [PATCH 1/1] audio: Fix playback noise with 24/32 bit sample size on BXT
by han.lu＠intel.com 07 Dec '15

07 Dec '15

From: "Lu, Han" <han.lu(a)intel.com> In BXT-P A0, HD-Audio DMA requests is later than expected, and makes an audio stream sensitive to system latencies when 24/32 bits are playing. Adjusting threshold of DMA fifo to force the DMA request sooner to improve latency tolerance at the expense of power. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 2ae8812..94dc6a9 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -93,6 +93,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_HSW_EM4 0x100c #define AZX_REG_HSW_EM5 0x1010 +/* Skylake/Broxton display HD-A controller Extended Mode registers */ +#define AZX_REG_SKL_EM4L 0x1040 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 22dbfa5..6c2aa4c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -27,6 +27,7 @@ #include <linux/module.h> #include <linux/pm_runtime.h> #include <linux/slab.h> +#include <linux/pci.h> #include <sound/core.h> #include <sound/initval.h> #include "hda_controller.h" @@ -34,6 +35,8 @@ #define CREATE_TRACE_POINTS #include "hda_controller_trace.h" +#define is_broxton(pci) ((pci)->device == 0x5a98) + /* DSP lock helpers */ #define dsp_lock(dev) snd_hdac_dsp_lock(azx_stream(dev)) #define dsp_unlock(dev) snd_hdac_dsp_unlock(azx_stream(dev)) @@ -880,14 +883,35 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_load_dsp_cleanup); #endif /* CONFIG_SND_HDA_DSP_LOADER */ /* + * In BXT-P A0, HD-Audio DMA requests is later than expected, + * and makes an audio stream sensitive to system latencies when + * 24/32 bits are playing. + * Adjusting threshold of DMA fifo to force the DMA request + * sooner to improve latency tolerance at the expense of power. + */ +static void bxt_reduce_dma_latency(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, SKL_EM4L); + val &= (0x3 << 20); + azx_writel(chip, SKL_EM4L, val); +} + +/* * reset and start the controller registers */ void azx_init_chip(struct azx *chip, bool full_reset) { + struct pci_dev *pci = chip->pci; + if (snd_hdac_bus_init_chip(azx_bus(chip), full_reset)) { /* correct RINTCNT for CXT */ if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) azx_writew(chip, RINTCNT, 0xc0); + /* reduce dma latency to avoid noise */ + if (is_broxton(pci)) + bxt_reduce_dma_latency(chip); } } EXPORT_SYMBOL_GPL(azx_init_chip); -- 2.5.0

3 2

[alsa-devel] [PATCH] crec: Always read whole number of fragments when length limit specified
by Richard Fitzgerald 07 Dec '15

07 Dec '15

This patch removes the lines that adjusted the read size to exactly read to the end of the calculated length in bytes. Doing this could lead to invalid fragment sizes being requested which may then cause a read error on some codecs. Instead we keep reading whole fragments until we have >= length bytes of data captured. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- src/utils/crec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/src/utils/crec.c b/src/utils/crec.c index 258fade..e61e620 100644 --- a/src/utils/crec.c +++ b/src/utils/crec.c @@ -332,9 +332,6 @@ void capture_samples(char *name, unsigned int card, unsigned int device, fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); do { - if (length && size > length - total_read) - size = length - total_read; - read = compress_read(compress, buffer, size); if (read < 0) { fprintf(stderr, "Error reading sample\n"); -- 1.9.1

2 1

[alsa-devel] M-audio 410 : complex procedure to get a sound
by ROUGE ETIENNE 06 Dec '15

06 Dec '15

Hello ! I've just bought a M-audio firewire 410 and I'm very thankfull to takashi's work that allows us to use the device almost like an USB one ! I can have sound and have access to the routing controls. However 2 bugs makes it a bit tedious : NOTE : I use an up to date freshly installed ubuntustudio 15.10. The only modifications I made was 1) deleting pulse 2) select the soundcard as default in asoundrc 1) I do not have sound out of the box, but I can manage to have sound by starting ffado-mixer then force restart alsa. After that everything works fine.... but I gotta do it at every restart which isn't nice 2) ffado-mixer suffers several bugs : - if I close the window I can't restart it again unless I kill ffado-dbus-server (it says : "another instance of ffado-mixer is already running). - I got few errors on startup "invalid samplerate for input"... don't know it it has any impact - when I tweak too much ffado-mixer (AFAIK, it's random, but happens within few minuts) the mouse shows a 'hand closed' cursor and I can't click anything. The system remains fully functionnal, but I can't close anything nor click... If I start an emergency terminal I can kill whatever I want, but the problem remains. I'll try with a fedora this evening to see if it's not related with ubuntustudio, but I'd be glad to have some help with those 2 issues (especially the stalling ffado-mixer) thanks ! Etienne

2 2

[alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module
by Takashi Sakamoto 06 Dec '15

06 Dec '15

Hi, This patchset update my previous post for merging ALSA scs1x module to oxfw module: [alsa-devel] [RFC][PATCH 0/9] ALSA: oxfw: merge scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100534.h… This patchset consists of three parts: * 01-05: code refactoring for OXFW module * 06-09: code refactoring for old firewire-speaker functionalities in OXFW module * 10-17: merge scs1x module Changes: * split some patches for a fine granularity * drop a small framework I proposed in previous patch * rename functions for AV/C Audio Subunit command After this patchset is applied, OXFW module is loaded instead of scs1x module. Then, some names except for rawmidi ID are changed, see patch 16. Currently, I find no advantage to keep them what they were. I'm welcome to receive your oppositions about it. Takashi Sakamoto (17): ALSA: oxfw: rename a file for control elements so that it's for model-specific ALSA: oxfw: rename local functions for control elements so that they represent as local ALSA: oxfw: change function prototype for AV/C Audio Subunit command ALSA: oxfw: reuse driver entry to detect quirk ALSA: oxfw: gather model-dependent conditions to a function ALSA: oxfw: add memory allocation for model-specific structure ALSA: oxfw: move model-specific members from common structure ALSA: oxfw: move model-specific parameters from common structure ALSA: oxfw: rename a structure so that it means backward compatibility to old drivers ALSA: oxfw: add scs1x layer ALSA: oxfw: allocate own address region for SCS.1 series ALSA: oxfw: copy handlers of asynchronous transaction for MIDI capture ALSA: oxfw: add MIDI capture port for SCS.1 models ALSA: oxfw: copy handlers of asynchronous transaction for MIDI playback ALSA: oxfw: add MIDI playback port for SCS.1 models ALSA: oxfw: obsolete scs1x module ALSA: oxfw: add stream format quirk for SCS.1 models sound/firewire/Kconfig | 12 +- sound/firewire/Makefile | 2 - sound/firewire/oxfw/Makefile | 4 +- sound/firewire/oxfw/oxfw-scs1x.c | 409 +++++++++++++++++++++ .../firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} | 142 ++++--- sound/firewire/oxfw/oxfw.c | 110 ++++-- sound/firewire/oxfw/oxfw.h | 23 +- 7 files changed, 586 insertions(+), 116 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-scs1x.c rename sound/firewire/oxfw/{oxfw-control.c => oxfw-spkr.c} (55%) -- 2.5.0

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Re: [alsa-devel] [sound:for-linus 29/29] sound/pci/rme96.c:1035:1-7: preceding lock on line 991
by Julia Lawall 04 Dec '15

04 Dec '15

Looks suspicious. Please check. julia On Sat, 5 Dec 2015, kbuild test robot wrote: > CC: kbuild-all(a)01.org > CC: alsa-devel(a)alsa-project.org > TO: Takashi Iwai <tiwai(a)suse.de> > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus > head: ec20c9c7a63f7b8676eb574406a8bc8f32b7ba7e > commit: ec20c9c7a63f7b8676eb574406a8bc8f32b7ba7e [29/29] ALSA: rme96: Fix unexpected volume reset after rate changes > :::::: branch date: 2 hours ago > :::::: commit date: 2 hours ago > > >> sound/pci/rme96.c:1035:1-7: preceding lock on line 991 > > git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git > git remote update sound > git checkout ec20c9c7a63f7b8676eb574406a8bc8f32b7ba7e > vim +1035 sound/pci/rme96.c > > ^1da177e Linus Torvalds 2005-04-16 985 > 4d23359b Clemens Ladisch 2005-09-05 986 runtime->dma_area = (void __force *)(rme96->iobase + > 4d23359b Clemens Ladisch 2005-09-05 987 RME96_IO_PLAY_BUFFER); > ^1da177e Linus Torvalds 2005-04-16 988 runtime->dma_addr = rme96->port + RME96_IO_PLAY_BUFFER; > ^1da177e Linus Torvalds 2005-04-16 989 runtime->dma_bytes = RME96_BUFFER_SIZE; > ^1da177e Linus Torvalds 2005-04-16 990 > ^1da177e Linus Torvalds 2005-04-16 @991 spin_lock_irq(&rme96->lock); > ^1da177e Linus Torvalds 2005-04-16 992 if (!(rme96->wcreg & RME96_WCR_MASTER) && > ^1da177e Linus Torvalds 2005-04-16 993 snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG && > ^1da177e Linus Torvalds 2005-04-16 994 (rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0) > ^1da177e Linus Torvalds 2005-04-16 995 { > ^1da177e Linus Torvalds 2005-04-16 996 /* slave clock */ > ^1da177e Linus Torvalds 2005-04-16 997 if ((int)params_rate(params) != rate) { > ec20c9c7 Takashi Iwai 2015-12-04 998 err = -EIO; > ec20c9c7 Takashi Iwai 2015-12-04 999 goto error; > ^1da177e Linus Torvalds 2005-04-16 1000 } > ec20c9c7 Takashi Iwai 2015-12-04 1001 } else { > ec20c9c7 Takashi Iwai 2015-12-04 1002 err = snd_rme96_playback_setrate(rme96, params_rate(params)); > ec20c9c7 Takashi Iwai 2015-12-04 1003 if (err < 0) > ec20c9c7 Takashi Iwai 2015-12-04 1004 goto error; > ec20c9c7 Takashi Iwai 2015-12-04 1005 apply_dac_volume = err > 0; /* need to restore volume later? */ > ^1da177e Linus Torvalds 2005-04-16 1006 } > ec20c9c7 Takashi Iwai 2015-12-04 1007 if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) > ec20c9c7 Takashi Iwai 2015-12-04 1008 goto error; > ^1da177e Linus Torvalds 2005-04-16 1009 snd_rme96_setframelog(rme96, params_channels(params), 1); > ^1da177e Linus Torvalds 2005-04-16 1010 if (rme96->capture_periodsize != 0) { > ^1da177e Linus Torvalds 2005-04-16 1011 if (params_period_size(params) << rme96->playback_frlog != > ^1da177e Linus Torvalds 2005-04-16 1012 rme96->capture_periodsize) > ^1da177e Linus Torvalds 2005-04-16 1013 { > ec20c9c7 Takashi Iwai 2015-12-04 1014 err = -EBUSY; > ec20c9c7 Takashi Iwai 2015-12-04 1015 goto error; > ^1da177e Linus Torvalds 2005-04-16 1016 } > ^1da177e Linus Torvalds 2005-04-16 1017 } > ^1da177e Linus Torvalds 2005-04-16 1018 rme96->playback_periodsize = > ^1da177e Linus Torvalds 2005-04-16 1019 params_period_size(params) << rme96->playback_frlog; > ^1da177e Linus Torvalds 2005-04-16 1020 snd_rme96_set_period_properties(rme96, rme96->playback_periodsize); > ^1da177e Linus Torvalds 2005-04-16 1021 /* S/PDIF setup */ > ^1da177e Linus Torvalds 2005-04-16 1022 if ((rme96->wcreg & RME96_WCR_ADAT) == 0) { > ^1da177e Linus Torvalds 2005-04-16 1023 rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP); > ^1da177e Linus Torvalds 2005-04-16 1024 writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER); > ^1da177e Linus Torvalds 2005-04-16 1025 } > ^1da177e Linus Torvalds 2005-04-16 1026 spin_unlock_irq(&rme96->lock); > ec20c9c7 Takashi Iwai 2015-12-04 1027 err = 0; > ^1da177e Linus Torvalds 2005-04-16 1028 > ec20c9c7 Takashi Iwai 2015-12-04 1029 error: > ec20c9c7 Takashi Iwai 2015-12-04 1030 if (apply_dac_volume) { > ec20c9c7 Takashi Iwai 2015-12-04 1031 usleep_range(3000, 10000); > ec20c9c7 Takashi Iwai 2015-12-04 1032 snd_rme96_apply_dac_volume(rme96); > ec20c9c7 Takashi Iwai 2015-12-04 1033 } > ec20c9c7 Takashi Iwai 2015-12-04 1034 > ec20c9c7 Takashi Iwai 2015-12-04 @1035 return err; > ^1da177e Linus Torvalds 2005-04-16 1036 } > ^1da177e Linus Torvalds 2005-04-16 1037 > ^1da177e Linus Torvalds 2005-04-16 1038 static int > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation >

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Re: [alsa-devel] [bug] Volume at maximum when track with different frequency is played with my RME sound card
by Maeda 04 Dec '15

04 Dec '15

Hi ! Someone's here ? Thanks. Le 23/10/2015 18:30, Maeda a écrit : > Hi all ! > > I don't know if the bug is known from your team, but some people > advice me to send an email to the Alsa devel list if nobody answer in > the kernel's bugtracking. > > Here is the bug I have : > https://bugzilla.kernel.org/show_bug.cgi?id=105771 > > What do you think about it ? > > If I was wrong sending this mail to this list, I'm sorry :) > > Thanks. > Have a nice day.

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[alsa-devel] [PATCH v2 0/9] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 04 Dec '15

04 Dec '15

Hi, this is a revised patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. The patch still doesn't follow the recent kernel doc comment changes in i915_component.h. I'll rebase once when I get the steady branch in drm tree. The current patchset is found in sound git tree test/hdmi-jack branch. Takashi === v1->v2: * Use modeset lock for get_eld lock, drop av mutex * Return the expected size from get_eld, not the copied size * Add reverse map from a port number to the encoder * Deferred invocation of get_eld from the eld_notify in HDA side * A bit more code refactoring in HDA side * Move audio component accessors to hdac_i915.c Takashi Iwai (9): drm/i915: Remove superfluous NULL check drm/i915: Add get_eld audio component drm/i915: refactoring audio component functions drm/i915: Add reverse mapping between port and intel_encoder ALSA: hda - Split ELD update code from hdmi_present_sense() ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Do zero-clear in snd_hdmi_parse_eld() itself ALSA: hda - Skip ELD notification during PM process ALSA: hda - Move audio component accesses to hdac_i915.c drivers/gpu/drm/i915/i915_drv.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 59 ++++++--- drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 1 + drivers/gpu/drm/i915/intel_drv.h | 1 + drivers/gpu/drm/i915/intel_hdmi.c | 2 + include/drm/i915_component.h | 6 + include/sound/hda_i915.h | 14 +++ sound/hda/hdac_i915.c | 66 +++++++++++ sound/pci/hda/hda_eld.c | 1 + sound/pci/hda/patch_hdmi.c | 237 +++++++++++++++++++++++++------------ 11 files changed, 294 insertions(+), 96 deletions(-) -- 2.6.3

3 34

[alsa-devel] [PATCH v2] ALSA: compress: add support for 32bit calls in a 64bit kernel
by Ravindra Lokhande 04 Dec '15

04 Dec '15

Compress offload does not support ioctl calls from a 32bit userspace in a 64 bit kernel. This patch adds support for ioctls from a 32bit userspace in a 64bit kernel Signed-off-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- v1 -> v2: use compat_ptr to convert 32 bit address to 64 bit pointer sound/core/compress_offload.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b123c42..4c156a7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -847,6 +847,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) return retval; } +/* support of 32bit userspace on 64bit platforms */ +#ifdef CONFIG_COMPAT +static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_compr_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} +#endif + static const struct file_operations snd_compr_file_ops = { .owner = THIS_MODULE, .open = snd_compr_open, @@ -854,6 +863,9 @@ static const struct file_operations snd_compr_file_ops = { .write = snd_compr_write, .read = snd_compr_read, .unlocked_ioctl = snd_compr_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = snd_compr_ioctl_compat, +#endif .mmap = snd_compr_mmap, .poll = snd_compr_poll, }; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable audio component for old Intel PCH devices
by Takashi Iwai 04 Dec '15

04 Dec '15

As i915 graphics driver provides the notification via audio component, not only the currently implemented HSW+ and VLV+ platforms but also all other PCH-based platforms (e.g. Cougar Point, Panther Point, etc) can use this infrastructure. It'll improve the reliability and the power consumption significantly, especially once when we implement the ELD notification via component. As a preliminary, this patch enables the usage of audio component for all PCH platforms. The HDA controller just needs to set AZX_DCAPS_I915_POWERWELL flag appropriately. The name of the flag is a bit confusing, but this actually works even on the chips without the powerwell but accesses only the other component ops. In the HDMI/DP codec driver side, we just need to register/unregister the notifier for such chips. This can be identified by checking the audio_component field in the assigned hdac_bus. One caveat is that PCH for Haswell and Broadwell must not be bound with i915 audio component, as there are dedicated HD-audio HDMI controllers on these platforms. Ditto for Poulsbo and Oaktrail as they use gma500 graphics, not i915. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 14 ++++++++++---- sound/pci/hda/patch_hdmi.c | 6 ++++-- 2 files changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 963f82430938..ee0e316401f9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -284,13 +284,19 @@ enum { (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) /* quirks for Intel PCH */ -#define AZX_DCAPS_INTEL_PCH_NOPM \ +#define AZX_DCAPS_INTEL_PCH_BASE \ (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH)) +/* PCH up to IVB; bound with i915 audio component for HDMI, no runtime PM */ +#define AZX_DCAPS_INTEL_PCH_NOPM \ + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_POWERWELL) + +/* PCH for HSW/BDW; with runtime PM, but no i915 binding */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) +/* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ @@ -2146,10 +2152,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Poulsbo */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a918377d3e9b..85342d261043 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -152,6 +152,8 @@ struct hdmi_spec { struct i915_audio_component_audio_ops i915_audio_ops; }; +#define codec_has_acomp(codec) \ + ((codec)->bus->core.audio_component != NULL) struct hdmi_audio_infoframe { u8 type; /* 0x84 */ @@ -2218,7 +2220,7 @@ static void generic_hdmi_free(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; int pin_idx; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) + if (codec_has_acomp(codec)) snd_hdac_i915_register_notifier(NULL); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { @@ -2390,7 +2392,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { + if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; -- 2.6.3

1 0

[alsa-devel] Clock Rate Clock Rate Adjustment API
by Chas Leichner 04 Dec '15

04 Dec '15

I would like to be able to make small adjustments the playback clock rate of a PCM device as it is playing in order to accurately synchronize music playback across several computers. Looking at the kernel source, this functionality has been previously implemented in both the aloop driver and ST's uniperif driver where the clock rate adjustment was exposed to the user space as a control element. Although the control interface can be used to communicate this information, the naming and properties of the clock adjustment controls don't appear to be standardized or documented and I don't see anything related to clock rate adjustment in Documentation/sound/alsa/ControlNames.txt. In order to make a standardized interface, do you think it makes more sense to standardize on a clock rate adjustment control or introduce a new interface other than the control interface because the control interface is only for mixer controls rather than arbitrary runtime changes to the driver? Regards, Chas

2 1

[alsa-devel] [PATCH 1/2] ASoC: arizona: Correct types of mixer texts and values
by Charles Keepax 03 Dec '15

03 Dec '15

The core expects "const char * const" and "unsigned int" for enum controls, various places in Arizona use "const char *" and "int". This patch corrects the type of these arrays. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 8 ++++---- sound/soc/codecs/arizona.h | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a23f7d1..d2731cf 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -310,7 +310,7 @@ int arizona_init_gpio(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_gpio); -const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { +const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", "Tone Generator 2", @@ -418,7 +418,7 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { }; EXPORT_SYMBOL_GPL(arizona_mixer_texts); -int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { +unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x00, /* None */ 0x04, /* Tone */ 0x05, @@ -555,12 +555,12 @@ const char *arizona_sample_rate_val_to_name(unsigned int rate_val) } EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name); -const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { +const char * const arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { "SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate", }; EXPORT_SYMBOL_GPL(arizona_rate_text); -const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { +const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { 0, 1, 2, 8, }; EXPORT_SYMBOL_GPL(arizona_rate_val); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 01a367c..b4f1867 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -96,8 +96,8 @@ struct arizona_priv { #define ARIZONA_NUM_MIXER_INPUTS 104 extern const unsigned int arizona_mixer_tlv[]; -extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; -extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; +extern const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern unsigned int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; #define ARIZONA_GAINMUX_CONTROLS(name, base) \ SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1, \ @@ -216,8 +216,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; #define ARIZONA_RATE_ENUM_SIZE 4 #define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14 -extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; -extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; +extern const char * const arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; +extern const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE]; extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE]; -- 2.1.4

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[alsa-devel] mx28: bad unlock balance detected
by Fabio Estevam 03 Dec '15

03 Dec '15

Hi, Running linux-next 20151126 on a mx28evk leads to the following bug: sgtl5000 0-000a: sgtl5000 revision 0x11 sgtl5000 0-000a: Using internal LDO instead of VDDD ===================================== [ BUG: bad unlock balance detected! ] 4.4.0-rc2-next-20151126-00001-g44eb6b2 #156 Not tainted ------------------------------------- kworker/u2:0/6 is trying to release lock (&rdev->mutex) at: [<c02e7858>] regulator_get_voltage+0x30/0x44 but there are no more locks to release! other info that might help us debug this: 5 locks held by kworker/u2:0/6: #0: ("%s""deferwq"){++++.+}, at: [<c002f1f8>] process_one_work+0x10c/0x49c #1: (deferred_probe_work){+.+.+.}, at: [<c002f1f8>] process_one_work+0x10c/0x49c #2: (&dev->mutex){......}, at: [<c03197bc>] __device_attach+0x20/0x108 #3: (client_mutex){+.+.+.}, at: [<c042dc34>] snd_soc_register_card+0x280/0x1108 #4: (&card->mutex){+.+.+.}, at: [<c042dc40>] snd_soc_register_card+0x28c/0x1108 stack backtrace: CPU: 0 PID: 6 Comm: kworker/u2:0 Not tainted 4.4.0-rc2-next-20151126-00001-g44eb6b2 #156 Hardware name: Freescale MXS (Device Tree) Workqueue: deferwq deferred_probe_work_func [<c000eea0>] (unwind_backtrace) from [<c000cfa0>] (show_stack+0x10/0x14) [<c000cfa0>] (show_stack) from [<c0049350>] (print_unlock_imbalance_bug+0xac/0xdc) [<c0049350>] (print_unlock_imbalance_bug) from [<c004c2f8>] (lock_release+0x2a4/0x38c) [<c004c2f8>] (lock_release) from [<c0513e70>] (__mutex_unlock_slowpath+0x7c/0x15c) [<c0513e70>] (__mutex_unlock_slowpath) from [<c02e7858>] (regulator_get_voltage+0x30/0x44) [<c02e7858>] (regulator_get_voltage) from [<c043c200>] (sgtl5000_probe+0x1c0/0x5f0) [<c043c200>] (sgtl5000_probe) from [<c0428a74>] (snd_soc_codec_drv_probe+0x14/0x18) [<c0428a74>] (snd_soc_codec_drv_probe) from [<c042cd54>] (soc_probe_component+0x200/0x330) [<c042cd54>] (soc_probe_component) from [<c042e178>] (snd_soc_register_card+0x7c4/0x1108) [<c042e178>] (snd_soc_register_card) from [<c0439cdc>] (devm_snd_soc_register_card+0x30/0x70) [<c0439cdc>] (devm_snd_soc_register_card) from [<c043d1cc>] (mxs_sgtl5000_probe+0xc8/0x138) [<c043d1cc>] (mxs_sgtl5000_probe) from [<c031b1fc>] (platform_drv_probe+0x50/0xac) [<c031b1fc>] (platform_drv_probe) from [<c0319ab0>] (driver_probe_device+0x1e4/0x2b0) [<c0319ab0>] (driver_probe_device) from [<c03180d8>] (bus_for_each_drv+0x44/0x8c) [<c03180d8>] (bus_for_each_drv) from [<c0319840>] (__device_attach+0xa4/0x108) [<c0319840>] (__device_attach) from [<c0318f84>] (bus_probe_device+0x88/0x90) [<c0318f84>] (bus_probe_device) from [<c0319370>] (deferred_probe_work_func+0x64/0x94) [<c0319370>] (deferred_probe_work_func) from [<c002f290>] (process_one_work+0x1a4/0x49c) [<c002f290>] (process_one_work) from [<c002f5e8>] (worker_thread+0x28/0x468) [<c002f5e8>] (worker_thread) from [<c0034dbc>] (kthread+0xc0/0xdc) [<c0034dbc>] (kthread) from [<c000a2ec>] (ret_from_fork+0x14/0x28) mxs-sgtl5000 sound: sgtl5000 <-> 80042000.saif mapping ok By inspecting the sgtl5000 codec driver I could not find where this "bad unlock balance" is happening. Any ideas? Thanks, Fabio Estevam

2 3

[alsa-devel] [PATCH v2] ASoC: da7218: Enable mic level detection reporting to user-space
by Adam Thomson 03 Dec '15

03 Dec '15

This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to user-space, if the mic level detect feature is enabled, and the audio captured at the chosen mic(s) is above a certain threshold. User-space can then handle the event accordingly (e.g. process audio capture stream). This method was chosen over ALSA control notification for a couple of reasons: 1) There's no requirement here for a control to read state from. The event is the only thing that's required and of interest. 2) tinyalsa support for control notifications does not exist so on platforms using this over alsa-lib there is a need to add code to support this event handling. Another possible option would be to use the standard Jack reporting framework but this really does not fit for this kind of event. Finally, use of the input device framework is not being encouraged, due to difficulties in enabling apps to access input devices, so this has also been avoided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Changes in v2: - More verbose and descriptive commit message added. sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 4fee7ae..752ed04 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 1.9.3

1 0

[alsa-devel] [PATCH] ALSA: Fix compat_ioctl handling for OSS emulations
by Takashi Iwai 03 Dec '15

03 Dec '15

The ALSA PCM, mixer and sequencer OSS emulations provide the 32bit compatible ioctl, but they just call the 64bit native ioctl as is. Although this works in most cases, passing the argument value as-is isn't guaranteed to work on all architectures. We need to convert it via compat_ptr() instead. This patch addresses the missing conversions. Since all relevant ioctls in these functions take the argument as a pointer, we do the pointer conversion in each compat_ioctl and pass it as a 64bit value to the native ioctl. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/mixer_oss.c | 8 +++++++- sound/core/oss/pcm_oss.c | 7 ++++++- sound/core/seq/oss/seq_oss.c | 7 ++++++- 3 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 7a8c79dd9734..2ff9c12d664a 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -24,6 +24,7 @@ #include <linux/time.h> #include <linux/string.h> #include <linux/module.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/control.h> @@ -397,7 +398,12 @@ int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned l #ifdef CONFIG_COMPAT /* all compatible */ -#define snd_mixer_oss_ioctl_compat snd_mixer_oss_ioctl +static long snd_mixer_oss_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_mixer_oss_ioctl1(file->private_data, cmd, + (unsigned long)compat_ptr(arg)); +} #else #define snd_mixer_oss_ioctl_compat NULL #endif diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 58550cc93f28..e557dbe469f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -33,6 +33,7 @@ #include <linux/module.h> #include <linux/math64.h> #include <linux/string.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/pcm.h> @@ -2648,7 +2649,11 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long #ifdef CONFIG_COMPAT /* all compatible */ -#define snd_pcm_oss_ioctl_compat snd_pcm_oss_ioctl +static long snd_pcm_oss_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_pcm_oss_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} #else #define snd_pcm_oss_ioctl_compat NULL #endif diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 7354b8bed860..8db156b207f1 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -23,6 +23,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/mutex.h> +#include <linux/compat.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/initval.h> @@ -189,7 +190,11 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) } #ifdef CONFIG_COMPAT -#define odev_ioctl_compat odev_ioctl +static long odev_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return odev_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); +} #else #define odev_ioctl_compat NULL #endif -- 2.6.3

1 0

[alsa-devel] [PATCH] ALSA: compress: add support for 32bit calls in a 64bit kernel
by Ravindra Lokhande 03 Dec '15

03 Dec '15

Compress offload does not support ioctl calls from a 32bit userspace in a 64 bit kernel. This patch adds support for ioctls from a 32bit userspace in a 64bit kernel Signed-off-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- sound/core/compress_offload.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b123c42..98f6b47 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -847,6 +847,15 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) return retval; } +/* support of 32bit userspace on 64bit platforms */ +#ifdef CONFIG_COMPAT +static long snd_compr_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + return snd_compr_ioctl(file, cmd, arg); +} +#endif + static const struct file_operations snd_compr_file_ops = { .owner = THIS_MODULE, .open = snd_compr_open, @@ -854,6 +863,9 @@ static const struct file_operations snd_compr_file_ops = { .write = snd_compr_write, .read = snd_compr_read, .unlocked_ioctl = snd_compr_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = snd_compr_ioctl_compat, +#endif .mmap = snd_compr_mmap, .poll = snd_compr_poll, }; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable power_save_node for CX20722
by David Henningsson 03 Dec '15

03 Dec '15

I've tested it on one device and it works fine, no clicks. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Alsa-info for this device is available here: https://launchpadlibrarian.net/228090406/AlsaInfo.txt Possibly it works well for other chips too, but this is the only one I have tested. Btw; I noticed another thing which I wonder whether it's correct or not: If I run "hda-emu AlsaInfo.txt", all nodes are in D3. But after running "P 0 p", some nodes (e g the internal speaker pin) stays in D0. Is that expected? I thought all pins would go down to D3 when playback closes. Same behaviour is seen on the real hardware, too. sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ef19890..6122b8c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -901,6 +901,9 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f150f2: + codec->power_save_node = 1; + /* Fall through */ default: codec->pin_amp_workaround = 1; snd_hda_pick_fixup(codec, cxt5066_fixup_models, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: da7218: Enable mic level detection reporting to user-space
by Adam Thomson 03 Dec '15

03 Dec '15

This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 4fee7ae..752ed04 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 1.9.3

3 4

[alsa-devel] [PATCH] hda-emu: Add snd_power_get_state
by David Henningsson 03 Dec '15

03 Dec '15

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/sound/core.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index e556765..1196319 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -113,6 +113,8 @@ int snd_card_register(struct snd_card *card) return 0; } +#define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; }) + #include <linux/list.h> #include <linux/mutex.h> #include <linux/pci.h> -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Rename NAU88L25->NAU8825 to stay consistent with existing driver name
by Anatol Pomozov 03 Dec '15

03 Dec '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/intel/Kconfig | 6 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/skl_nau8825_ssm4567.c | 366 ++++++++++++++++++++++++++ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 366 -------------------------- sound/soc/intel/skylake/skl.c | 2 +- 5 files changed, 372 insertions(+), 372 deletions(-) create mode 100644 sound/soc/intel/boards/skl_nau8825_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2d3b124..7108e13 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -156,8 +156,8 @@ config SND_SOC_INTEL_SKL_RT286_MACH Say Y if you have such a device If unsure select "N". -config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH - tristate "ASoC Audio driver for SKL with NAU88L25 and SSM4567 in I2S Mode" +config SND_SOC_INTEL_SKL_NAU8825_SSM4567_MACH + tristate "ASoC Audio driver for SKL with NAU8825 and SSM4567 in I2S Mode" depends on X86_INTEL_LPSS && I2C select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE @@ -166,6 +166,6 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH select SND_SOC_DMIC help This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for NAU88L25 + SSM4567. + create an alsa sound card for NAU8825 + SSM4567. Say Y if you have such a device If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a59f762..fb50457 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -7,7 +7,7 @@ snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o snd-soc-skl_rt286-objs := skl_rt286.o -snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-skl_nau8825_ssm4567-objs := skl_nau8825_ssm4567.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 @@ -18,4 +18,4 @@ obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o -obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU8825_SSM4567_MACH) += snd-soc-skl_nau8825_ssm4567.o diff --git a/sound/soc/intel/boards/skl_nau8825_ssm4567.c b/sound/soc/intel/boards/skl_nau8825_ssm4567.c new file mode 100644 index 0000000..7c34927 --- /dev/null +++ b/sound/soc/intel/boards/skl_nau8825_ssm4567.c @@ -0,0 +1,366 @@ +/* + * Intel Skylake I2S Machine Driver for NAU8825+SSM4567 + * + * Copyright (C) 2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Skylake I2S Machine Driver for NAU8825 and SSM4567 + * + * Copyright (C) 2015, 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/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" +#define SKL_SSM_CODEC_DAI "ssm4567-hifi" + +static struct snd_soc_jack skylake_headset; +static struct snd_soc_card skylake_audio_card; + +static inline struct snd_soc_dai *skl_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, SKL_NUVOTON_CODEC_DAI, + strlen(SKL_NUVOTON_CODEC_DAI))) + return rtd->codec_dai; + } + + return NULL; +} + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Speaker"), + SOC_DAPM_PIN_SWITCH("Right Speaker"), +}; + +static int platform_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; + int ret; + + codec_dai = skl_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, + NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return -EIO; + } + } else { + ret = snd_soc_dai_set_sysclk(codec_dai, + NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return -EIO; + } + } + return ret; +} + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Speaker", NULL), + SND_SOC_DAPM_SPK("Right Speaker", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route skylake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + + /* speaker */ + {"Left Speaker", NULL, "Left OUT"}, + {"Right Speaker", NULL, "Right OUT"}, + + /* other jacks */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "Left Playback", NULL, "ssp0 Tx"}, + { "Right Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + + { "AIF1 Playback", NULL, "ssp1 Tx"}, + { "ssp1 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp1 Rx" }, + { "ssp1 Rx", NULL, "AIF1 Capture" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static struct snd_soc_codec_conf ssm4567_codec_conf[] = { + { + .dev_name = "i2c-INT343B:00", + .name_prefix = "Left", + }, + { + .dev_name = "i2c-INT343B:01", + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component ssm4567_codec_components[] = { + { /* Left */ + .name = "i2c-INT343B:00", + .dai_name = SKL_SSM_CODEC_DAI, + }, + { /* Right */ + .name = "i2c-INT343B:01", + .dai_name = SKL_SSM_CODEC_DAI, + }, +}; + +static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + /* Slot 1 for left */ + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48); + if (ret < 0) + return ret; + + /* Slot 2 for right */ + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48); + if (ret < 0) + return ret; + + return ret; +} + +static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_codec *codec = rtd->codec; + + /* + * 4 buttons here map to the google Reference headset + * The use of these buttons can be decided by the user space. + */ + ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, + NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); + return ret; + } + + nau8825_enable_jack_detect(codec, &skylake_headset); + + return ret; +} + +static int skylake_ssp_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); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_nau8825_ops = { + .hw_params = skylake_nau8825_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssm4567_codec_components, + .num_codecs = ARRAY_SIZE(ssm4567_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = skylake_ssm4567_codec_init, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .be_id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-10508825:00", + .codec_dai_name = SKL_NUVOTON_CODEC_DAI, + .init = skylake_nau8825_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp_fixup, + .ops = &skylake_nau8825_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + NAU8825 */ +static struct snd_soc_card skylake_audio_card = { + .name = "sklnau8825adi", + .owner = THIS_MODULE, + .dai_link = skylake_dais, + .num_links = ARRAY_SIZE(skylake_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_map, + .num_dapm_routes = ARRAY_SIZE(skylake_map), + .codec_conf = ssm4567_codec_conf, + .num_configs = ARRAY_SIZE(ssm4567_codec_conf), +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_audio_card.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .driver = { + .name = "skl_nau8825_ssm4567_i2s", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); +MODULE_AUTHOR("Naveen M <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Yong Zhi <yong.zhi(a)intel.com>"); +MODULE_DESCRIPTION("Intel Audio Machine driver for SKL with NAU8825 and SSM4567 in I2S Mode"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_nau8825_ssm4567_i2s"); diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c deleted file mode 100644 index 65c65d4..0000000 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ /dev/null @@ -1,366 +0,0 @@ -/* - * Intel Skylake I2S Machine Driver for NAU88L25+SSM4567 - * - * Copyright (C) 2015, Intel Corporation. All rights reserved. - * - * Modified from: - * Intel Skylake I2S Machine Driver for NAU88L25 and SSM4567 - * - * Copyright (C) 2015, 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/module.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> -#include <sound/pcm_params.h> -#include "../../codecs/nau8825.h" - -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" -#define SKL_SSM_CODEC_DAI "ssm4567-hifi" - -static struct snd_soc_jack skylake_headset; -static struct snd_soc_card skylake_audio_card; - -static inline struct snd_soc_dai *skl_get_codec_dai(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - - list_for_each_entry(rtd, &card->rtd_list, list) { - - if (!strncmp(rtd->codec_dai->name, SKL_NUVOTON_CODEC_DAI, - strlen(SKL_NUVOTON_CODEC_DAI))) - return rtd->codec_dai; - } - - return NULL; -} - -static const struct snd_kcontrol_new skylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Speaker"), - SOC_DAPM_PIN_SWITCH("Right Speaker"), -}; - -static int platform_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; - int ret; - - codec_dai = skl_get_codec_dai(card); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } - return ret; -} - -static const struct snd_soc_dapm_widget skylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Speaker", NULL), - SND_SOC_DAPM_SPK("Right Speaker", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route skylake_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - {"Headphone Jack", NULL, "HPOL"}, - {"Headphone Jack", NULL, "HPOR"}, - - /* speaker */ - {"Left Speaker", NULL, "Left OUT"}, - {"Right Speaker", NULL, "Right OUT"}, - - /* other jacks */ - {"MIC", NULL, "Headset Mic"}, - {"DMIC AIF", NULL, "SoC DMIC"}, - - /* CODEC BE connections */ - { "Left Playback", NULL, "ssp0 Tx"}, - { "Right Playback", NULL, "ssp0 Tx"}, - { "ssp0 Tx", NULL, "codec0_out"}, - - { "AIF1 Playback", NULL, "ssp1 Tx"}, - { "ssp1 Tx", NULL, "codec1_out"}, - - { "codec0_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "AIF1 Capture" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "Capture" }, - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, -}; - -static struct snd_soc_codec_conf ssm4567_codec_conf[] = { - { - .dev_name = "i2c-INT343B:00", - .name_prefix = "Left", - }, - { - .dev_name = "i2c-INT343B:01", - .name_prefix = "Right", - }, -}; - -static struct snd_soc_dai_link_component ssm4567_codec_components[] = { - { /* Left */ - .name = "i2c-INT343B:00", - .dai_name = SKL_SSM_CODEC_DAI, - }, - { /* Right */ - .name = "i2c-INT343B:01", - .dai_name = SKL_SSM_CODEC_DAI, - }, -}; - -static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - /* Slot 1 for left */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48); - if (ret < 0) - return ret; - - /* Slot 2 for right */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48); - if (ret < 0) - return ret; - - return ret; -} - -static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct snd_soc_codec *codec = rtd->codec; - - /* - * 4 buttons here map to the google Reference headset - * The use of these buttons can be decided by the user space. - */ - ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, - NULL, 0); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } - - nau8825_enable_jack_detect(codec, &skylake_headset); - - return ret; -} - -static int skylake_ssp_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); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP0 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; - -/* skylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link skylake_dais[] = { - /* Front End DAI links */ - { - .name = "Skl Audio Port", - .stream_name = "Audio", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - }, - { - .name = "Skl Audio Capture Port", - .stream_name = "Audio Record", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - { - .name = "Skl Audio Reference cap", - .stream_name = "refcap", - .cpu_dai_name = "Reference Pin", - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .platform_name = "0000:00:1f.3", - .init = NULL, - .dpcm_capture = 1, - .ignore_suspend = 1, - .nonatomic = 1, - .dynamic = 1, - }, - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .be_id = 0, - .cpu_dai_name = "SSP0 Pin", - .platform_name = "0000:00:1f.3", - .no_pcm = 1, - .codecs = ssm4567_codec_components, - .num_codecs = ARRAY_SIZE(ssm4567_codec_components), - .dai_fmt = SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .init = skylake_ssm4567_codec_init, - .ignore_suspend = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .dpcm_playback = 1, - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .be_id = 0, - .cpu_dai_name = "SSP1 Pin", - .platform_name = "0000:00:1f.3", - .no_pcm = 1, - .codec_name = "i2c-10508825:00", - .codec_dai_name = SKL_NUVOTON_CODEC_DAI, - .init = skylake_nau8825_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ignore_suspend = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .ops = &skylake_nau8825_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - }, - { - .name = "dmic01", - .be_id = 1, - .cpu_dai_name = "DMIC01 Pin", - .codec_name = "dmic-codec", - .codec_dai_name = "dmic-hifi", - .platform_name = "0000:00:1f.3", - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - }, -}; - -/* skylake audio machine driver for SPT + NAU88L25 */ -static struct snd_soc_card skylake_audio_card = { - .name = "sklnau8825adi", - .owner = THIS_MODULE, - .dai_link = skylake_dais, - .num_links = ARRAY_SIZE(skylake_dais), - .controls = skylake_controls, - .num_controls = ARRAY_SIZE(skylake_controls), - .dapm_widgets = skylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), - .dapm_routes = skylake_map, - .num_dapm_routes = ARRAY_SIZE(skylake_map), - .codec_conf = ssm4567_codec_conf, - .num_configs = ARRAY_SIZE(ssm4567_codec_conf), -}; - -static int skylake_audio_probe(struct platform_device *pdev) -{ - skylake_audio_card.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); -} - -static struct platform_driver skylake_audio = { - .probe = skylake_audio_probe, - .driver = { - .name = "skl_nau88l25_ssm4567_i2s", - .pm = &snd_soc_pm_ops, - }, -}; - -module_platform_driver(skylake_audio) - -/* Module information */ -MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); -MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); -MODULE_AUTHOR("Naveen M <naveen.m(a)intel.com>"); -MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); -MODULE_AUTHOR("Yong Zhi <yong.zhi(a)intel.com>"); -MODULE_DESCRIPTION("Intel Audio Machine driver for SKL with NAU88L25 and SSM4567 in I2S Mode"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:skl_nau88l25_ssm4567_i2s"); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index d3e87b6..0551322 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -574,7 +574,7 @@ static void skl_remove(struct pci_dev *pci) static struct sst_acpi_mach sst_skl_devdata[] = { { "INT343A", "skl_alc286s_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, - { "INT343B", "skl_nau88l25_ssm4567_i2s", "intel/dsp_fw_release.bin", + { "INT343B", "skl_nau8825_ssm4567_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, {} }; -- 2.6.0.rc2.230.g3dd15c0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: care SWSP bit for TDM/non-TDM
by Kuninori Morimoto 02 Dec '15

02 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSICR::SWSP bit controls WS signal low/high, but in case of TDM it is inverted. This patch solves this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 628739f..79c3211 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -265,6 +265,9 @@ static int rsnd_ssi_config_init(struct rsnd_ssi *ssi, u32 cr_own; u32 cr_mode; u32 wsr; + int is_tdm; + + is_tdm = (rsnd_get_slot_runtime(io) >= 6) ? 1 : 0; /* * always use 32bit system word. @@ -274,7 +277,7 @@ static int rsnd_ssi_config_init(struct rsnd_ssi *ssi, if (rdai->bit_clk_inv) cr_own |= SCKP; - if (rdai->frm_clk_inv) + if (rdai->frm_clk_inv ^ is_tdm) cr_own |= SWSP; if (rdai->data_alignment) cr_own |= SDTA; @@ -307,7 +310,7 @@ static int rsnd_ssi_config_init(struct rsnd_ssi *ssi, * rsnd_ssiu_init_gen2() */ wsr = ssi->wsr; - if (rsnd_get_slot_runtime(io) >= 6) { + if (is_tdm) { wsr |= WS_MODE; cr_own |= CHNL_8; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Make enable/disable_pin work with always on widgets
by Charles Keepax 02 Dec '15

02 Dec '15

Always on widgets currently have some odd interactions with DAPM. Enabling/disabling a widget (snd_soc_dapm_enable_pin) then connecting it to a path works as expected, ie. when the widget is disabled the path doesn't power up and it does when the widget is enabled. However once in a path enabling the widget does not cause anything to power up, dapm_widget_set_power will return the current power state of the widget as 1, meaning we never check peer power states. This patch updates dapm_always_on_check_power to return w->connected such that it is effected by snd_soc_dapm_enable_pin and the like. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a8156b5..5a2812f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1300,7 +1300,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) static int dapm_always_on_check_power(struct snd_soc_dapm_widget *w) { - return 1; + return w->connected; } static int dapm_seq_compare(struct snd_soc_dapm_widget *a, -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usx2y: fix inconsistent indenting on if statement
by Colin King 02 Dec '15

02 Dec '15

From: Colin Ian King <colin.king(a)canonical.com> minor change, indenting is one tab out. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 61d5dc2..dd40ca9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -166,7 +166,7 @@ static int usX2Y_urb_play_prepare(struct snd_usX2Y_substream *subs, /* set the buffer pointer */ urb->transfer_buffer = runtime->dma_area + subs->hwptr * usX2Y->stride; if ((subs->hwptr += count) >= runtime->buffer_size) - subs->hwptr -= runtime->buffer_size; + subs->hwptr -= runtime->buffer_size; } else urb->transfer_buffer = subs->tmpbuf; -- 2.6.2

2 1

[alsa-devel] [PATCH v4 0/3]ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 02 Dec '15

02 Dec '15

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.3-rc5 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v4 Changes from v3: - Fix the subject lines to match the style for the subsystem Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 54 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 859 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1508 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 129 ++ 6 files changed, 2564 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

2 6

[alsa-devel] [PATCH] ASoC: dapm: Replace siggen widget type with mic widgets
by Charles Keepax 02 Dec '15

02 Dec '15

The siggen widget type doesn't really require its own special handling. It can be handled identically to a mic widget, ie. it is a source of a signal and causes things to power up when an output is connected. Additionally, several places (cs42l52, cs42l56, wm8962, arizona_haptics) tie siggens in to the input framework and the pattern for this is usually to use snd_soc_dapm_enable_pin/disable_pin on the siggen widget. However, as siggen widgets present as always on (the power check always returns 1), this leads to some odd interactions with DAPM. Enabling/disabling the widget then connecting it to a path works as expected, however once in a path enabling the widget has no effect. dapm_widget_set_power will return immediately as the current power state of the widget will return 1, meaning we never check peer power states. Treating the widget as a mic widget causes the expected behaviour in all cases. This does cause a minor cosmetic side effect that siggen widgets no longer show as always on through debugfs, although in the case described above that is more representive of their behaviour. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-dapm.c | 6 +----- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..67f176f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -47,7 +47,7 @@ struct device; /* platform domain */ #define SND_SOC_DAPM_SIGGEN(wname) \ -{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \ +{ .id = snd_soc_dapm_mic, .name = wname, .kcontrol_news = NULL, \ .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_SINK(wname) \ { .id = snd_soc_dapm_sink, .name = wname, .kcontrol_news = NULL, \ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a8156b5..ea6f477 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1817,12 +1817,9 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Supplies and micbiases only bring the * context up to STANDBY as unless something * else is active and passing audio they - * generally don't require full power. Signal - * generators are virtual pins and have no - * power impact themselves. + * generally don't require full power. */ switch (w->id) { - case snd_soc_dapm_siggen: case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: @@ -3354,7 +3351,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_vmid: - case snd_soc_dapm_siggen: w->is_ep = SND_SOC_DAPM_EP_SOURCE; w->power_check = dapm_always_on_check_power; break; -- 2.1.4

3 3

[alsa-devel] snd-fireworks requires larger period sizes than FFADO
by Dominic Sacré 02 Dec '15

02 Dec '15

Hi, I'm trying to use an Echo AudioFire 4 with the snd-fireworks driver. With FFADO the device works reasonably well at a period size of 64 samples (on a PREEMPT_RT kernel), whereas with snd-fireworks the lowest possible period size seems to be 256 samples. This is what JACK says when I try to start it with a period size of 128 samples: configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: cannot set period size to 128 frames for capture The lowest achievable analog roundtrip latency seems to be around 19ms (at 2x256 frames), as opposed to 8.3ms with FFADO (at 2x64). Are there any known limitations of the snd-fireworks driver in this regard? Thanks, Dominic

4 4

[alsa-devel] [PATCH 0/14] ASoC: rsnd: TDM Extend Mode support
by Kuninori Morimoto 02 Dec '15

02 Dec '15

Hi Mark These patch-set adds TDM Extend Mode to Renesas R-Car sound driver. 1) - 9) tidyup / cleanup patches 10) - 14) TDM Extend Mode support Unfortunately, our board can't use this TDM Extend Mode, because our SoC and codec are using different style TDM. Thus I have tested this patches on special board which doesn't have codec chip, and I checked signal via oscilloscope. oKuninori Morimoto (14): 1) ASoC: rsnd: don't use normal *mod in adg.c 2) ASoC: rsnd: add missing ADINR::CHNUM on DVC/SRC/SSIU 3) ASoC: rsnd: add missing SRC_O_BUSIF_MODE register 4) ASoC: rsnd: src: rename rsnd_src_soft_reset() to rsnd_src_activation() 5) ASoC: rsnd: mix: rename rsnd_mix_soft_reset() to rsnd_mix_activation() 6) ASoC: rsnd: dvc: rename rsnd_dvc_soft_reset() to rsnd_dvc_activation() 7) ASoC: rsnd: src: add rsnd_src_halt() 8) ASoC: rsnd: mix: add rsnd_mix_halt() 9) ASoC: rsnd: dvc: add rsnd_dvc_halt() 10) ASoC: rsnd: add rsnd_ssi_config_init() 11) ASoC: rsnd: set SSIWSR setting on rsnd_ssi_config_init() 12) ASoC: rsnd: ssi enables non-stereo sound 13) ASoC: rsnd: dvc enables non-stereo sound 14) ASoC: rsnd: add TDM Extend Mode support sound/soc/sh/rcar/adg.c | 30 +++++++++++------------------- sound/soc/sh/rcar/core.c | 65 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/dvc.c | 32 +++++++++++++++++++++++++++----- sound/soc/sh/rcar/gen.c | 11 ++++++++++- sound/soc/sh/rcar/mix.c | 12 ++++++++++-- sound/soc/sh/rcar/rsnd.h | 19 +++++++++++++++++-- sound/soc/sh/rcar/src.c | 18 ++++++++++++++---- sound/soc/sh/rcar/ssi.c | 126 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++--------------------------------------------------- sound/soc/sh/rcar/ssiu.c | 12 +++++++++++- 9 files changed, 235 insertions(+), 90 deletions(-) Best regards --- Kuninori Morimoto

2 28

[alsa-devel] [PATCH] ASoC: rsnd: fixup wrong snd_soc_dai_driver pointer access
by Kuninori Morimoto 01 Dec '15

01 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> drv pointer should be "base + offset" instead of "current + offset". This patch fixup this issue, otherwise third and subsequent pointer will be broken Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index b187a89..f1d7af1 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -603,7 +603,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct device_node *playback, *capture; struct rsnd_dai_stream *io_playback; struct rsnd_dai_stream *io_capture; - struct snd_soc_dai_driver *drv; + struct snd_soc_dai_driver *rdrv, *drv; struct rsnd_dai *rdai; struct device *dev = rsnd_priv_to_dev(priv); int nr, dai_i, io_i, np_i; @@ -616,15 +616,15 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) goto rsnd_dai_probe_done; } - drv = devm_kzalloc(dev, sizeof(*drv) * nr, GFP_KERNEL); + rdrv = devm_kzalloc(dev, sizeof(*rdrv) * nr, GFP_KERNEL); rdai = devm_kzalloc(dev, sizeof(*rdai) * nr, GFP_KERNEL); - if (!drv || !rdai) { + if (!rdrv || !rdai) { ret = -ENOMEM; goto rsnd_dai_probe_done; } priv->rdai_nr = nr; - priv->daidrv = drv; + priv->daidrv = rdrv; priv->rdai = rdai; /* @@ -633,7 +633,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i = 0; for_each_child_of_node(dai_node, dai_np) { rdai = rsnd_rdai_get(priv, dai_i); - drv = drv + dai_i; + drv = rdrv + dai_i; io_playback = &rdai->playback; io_capture = &rdai->capture; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: rsrc-card: check return value of snd_soc_of_get_dai_name()
by Kuninori Morimoto 01 Dec '15

01 Dec '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing check of snd_soc_of_get_dai_name(). It might not be able to use sound card, because it might returns -EPROBE_DEFER. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index d61db9c..a3ec13f 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -208,7 +208,9 @@ static int rsrc_card_parse_links(struct device_node *np, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; dai_link->cpu_of_node = args.np; - snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); + ret = snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); + if (ret < 0) + return ret; /* set dai_name */ snprintf(dai_props->dai_name, DAI_NAME_NUM, "fe.%s", @@ -240,7 +242,9 @@ static int rsrc_card_parse_links(struct device_node *np, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup; dai_link->codec_of_node = args.np; - snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); + ret = snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); + if (ret < 0) + return ret; /* additional name prefix */ if (of_data) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Add 32uS delay after putting FLL into freerun
by Charles Keepax 01 Dec '15

01 Dec '15

When switching between two clock sources using the FLL freerun to smooth the transition we should wait 32uS after putting the FLL into freerun before we proceed. In practice we appear to be getting enough delay from the surrounding code, but better to make it explicit. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e76ecc7..a23f7d1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2212,9 +2212,9 @@ static int arizona_enable_fll(struct arizona_fll *fll) /* Facilitate smooth refclk across the transition */ regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); - regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, - ARIZONA_FLL1_FREERUN); + regmap_update_bits(fll->arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + udelay(32); } /* -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: sunxi: Remove useless comments and variable
by Maxime Ripard 01 Dec '15

01 Dec '15

The comment is misleading on how we should support external power amps, and the variable is not used and generates a warning. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun4i-codec.c | 22 ---------------------- 1 file changed, 22 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 30c9e9260491..516c7c2b479a 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -109,11 +109,6 @@ struct sun4i_codec { static void sun4i_codec_start_playback(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO high here on some boards - */ - /* Flush TX FIFO */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), @@ -127,11 +122,6 @@ static void sun4i_codec_start_playback(struct sun4i_codec *scodec) static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO low here on some boards - */ - /* Disable DAC DRQ */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), @@ -140,11 +130,6 @@ static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) static void sun4i_codec_start_capture(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO high here on some boards - */ - /* Enable ADC DRQ */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), @@ -153,11 +138,6 @@ static void sun4i_codec_start_capture(struct sun4i_codec *scodec) static void sun4i_codec_stop_capture(struct sun4i_codec *scodec) { - /* - * FIXME: according to the BSP, we might need to drive a PA - * GPIO low here on some boards - */ - /* Disable ADC DRQ */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); @@ -358,8 +338,6 @@ static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, struct snd_pcm_hw_params *params, unsigned int hwrate) { - u32 val; - /* Set ADC sample rate */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, 7 << SUN4I_CODEC_ADC_FIFOC_ADC_FS, -- 2.6.3

2 3

[alsa-devel] [PATCH] ASoC: tegra_alc5632: check return value
by Sudip Mukherjee 01 Dec '15

01 Dec '15

We have been returning success even if snd_soc_card_jack_new() fails. Lets check the return value and return error if it fails. Fixes: 12cc6d1dca4d ("ASoC: tegra_alc5632: Register jacks at the card level") Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/tegra/tegra_alc5632.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index ba272e2..deb597f 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -101,12 +101,16 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { + int ret; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack, - tegra_alc5632_hs_jack_pins, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); + if (ret) + return ret; if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: hdac_hdmi: check error return
by Sudip Mukherjee 01 Dec '15

01 Dec '15

As hdac->num_nodes is unsigned we can not check if snd_hdac_get_sub_nodes() has returned error or success. Lets have a temporary int to check the error value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v2: made num_nodes as int, but Takashi said that num_nodes should really be an unsigned. v1: removed the error check as that condition is always false. sound/soc/codecs/hdac_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 205f2c2..1a2f33b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -409,17 +409,18 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev, static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev) { hda_nid_t nid; - int i; + int i, num_nodes; struct hdac_device *hdac = &edev->hdac; struct hdac_hdmi_priv *hdmi = edev->private_data; int cvt_nid = 0, pin_nid = 0; - hdac->num_nodes = snd_hdac_get_sub_nodes(hdac, hdac->afg, &nid); - if (!nid || hdac->num_nodes < 0) { + num_nodes = snd_hdac_get_sub_nodes(hdac, hdac->afg, &nid); + if (!nid || num_nodes < 0) { dev_warn(&hdac->dev, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } + hdac->num_nodes = num_nodes; hdac->start_nid = nid; for (i = 0; i < hdac->num_nodes; i++, nid++) { -- 1.9.1

2 1

[alsa-devel] [asoc:topic/da7218 2/2] sound/soc/codecs/da7218.c:3214:9-10: WARNING: return of 0/1 in function 'da7218_volatile_register' with return type bool
by kbuild test robot 01 Dec '15

01 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/da7218 head: 4d50934abd2261fd467320d52c470efff309fd74 commit: 4d50934abd2261fd467320d52c470efff309fd74 [2/2] ASoC: da7218: Add da7218 codec driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/da7218.c:3214:9-10: WARNING: return of 0/1 in function 'da7218_volatile_register' with return type bool Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

Re: [alsa-devel] [PATCH v2] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by Mark Brown 01 Dec '15

01 Dec '15

On Thu, Nov 26, 2015 at 07:39:38AM +0800, kbuild test robot wrote: > Hi Tim, > > [auto build test ERROR on v4.4-rc2] > [cannot apply to asoc/for-next next-20151124] Please look into these issues (especially the build failure) - if kbuild says the code doesn't build that needs fixing up or understanding.

2 2

[alsa-devel] [PATCH 0/3] ASoC: Implement DAI links in a list & some code refactoring
by mengdong.lin＠linux.intel.com 01 Dec '15

01 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This is still a preparation for topology to create PCM at dynamically. First two patches are code refactoring. The 3rd patch implements the DAI links into a list, so that the number of links can add/shrink by topology at runtime. The use scenario may be like this: - When instantiating the soc card, probing a component with topology can bring new DAI links. The ASoC will probe the new links (in later patches). - When unregistering the soc card, removing this component will also remove DAI links created by it. For backward compatiblity, the machine driver can still define a static link array. And the ASoC core will add these links into the list when instantiating the soc card. Mengdong Lin (3): ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add/remove a link include/sound/soc.h | 15 ++- sound/soc/soc-core.c | 251 +++++++++++++++++++++++++++++++++------------------ 2 files changed, 177 insertions(+), 89 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH 0/2] correction of iec958 channel status sampling freq setting
by Moise Gergaud 01 Dec '15

01 Dec '15

patch "ASoC: sti: set iec958 channel status sampling freq" corresponds to the V2 of "[PATCH 4/4] ASoC: sti: reset iec60958 settings on close" Moise Gergaud (2): ASoC: sti: set iec958 channel status sampling freq ASoC: sti: correction for HBRA support sound/soc/sti/uniperif_player.c | 73 +++++++++++++++++++++++++---------------- 1 file changed, 44 insertions(+), 29 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH] ASoC: sun4i: Implement MIC1 capture
by Maxime Ripard 01 Dec '15

01 Dec '15

One of the input path used in the Allwinner codec is the MIC1. Add support for it. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun4i-codec.c | 228 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 197 insertions(+), 31 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da168b6..30c9e9260491 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -69,6 +69,7 @@ /* Codec ADC register offsets and bit fields */ #define SUN4I_CODEC_ADC_FIFOC (0x1c) +#define SUN4I_CODEC_ADC_FIFOC_ADC_FS (29) #define SUN4I_CODEC_ADC_FIFOC_EN_AD (28) #define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24) #define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8) @@ -102,6 +103,7 @@ struct sun4i_codec { struct clk *clk_apb; struct clk *clk_module; + struct snd_dmaengine_dai_dma_data capture_dma_data; struct snd_dmaengine_dai_dma_data playback_dma_data; }; @@ -136,26 +138,54 @@ static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) 0); } +static void sun4i_codec_start_capture(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO high here on some boards + */ + + /* Enable ADC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); +} + +static void sun4i_codec_stop_capture(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO low here on some boards + */ + + /* Disable ADC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); +} + static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return -ENOTSUPP; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - sun4i_codec_start_playback(scodec); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sun4i_codec_start_playback(scodec); + else + sun4i_codec_start_capture(scodec); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - sun4i_codec_stop_playback(scodec); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sun4i_codec_stop_playback(scodec); + else + sun4i_codec_stop_capture(scodec); break; default: @@ -165,15 +195,54 @@ static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_codec_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - u32 val; - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return -ENOTSUPP; + + /* Flush RX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH)); + + + /* Set RX FIFO trigger level */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + 0xf << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL, + 0x7 << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL); + + /* + * FIXME: Undocumented in the datasheet, but + * Allwinner's code mentions that it is related + * related to microphone gain + */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_ACTL, + 0x3 << 25, + 0x1 << 25); + + if (of_device_is_compatible(scodec->dev->of_node, + "allwinner,sun7i-a20-codec")) + /* FIXME: Undocumented bits */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_TUNE, + 0x3 << 8, + 0x1 << 8); + + /* Fill most significant bits with valid data MSB */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); + + return 0; +} + +static int sun4i_codec_prepare_playback(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + u32 val; /* Flush the TX FIFO */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, @@ -202,6 +271,15 @@ static int sun4i_codec_prepare(struct snd_pcm_substream *substream, 0); return 0; +}; + +static int sun4i_codec_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return sun4i_codec_prepare_playback(substream, dai); + + return sun4i_codec_prepare_capture(substream, dai); } static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params) @@ -276,30 +354,34 @@ static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params) } } -static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, + struct snd_pcm_hw_params *params, + unsigned int hwrate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - unsigned long clk_freq; - int ret, hwrate; u32 val; - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return -ENOTSUPP; + /* Set ADC sample rate */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + 7 << SUN4I_CODEC_ADC_FIFOC_ADC_FS, + hwrate << SUN4I_CODEC_ADC_FIFOC_ADC_FS); - clk_freq = sun4i_codec_get_mod_freq(params); - if (!clk_freq) - return -EINVAL; + /* Set the number of channels we want to use */ + if (params_channels(params) == 1) + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN)); + else + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), 0); - ret = clk_set_rate(scodec->clk_module, clk_freq); - if (ret) - return ret; + return 0; +} - hwrate = sun4i_codec_get_hw_rate(params); - if (hwrate < 0) - return hwrate; +static int sun4i_codec_hw_params_playback(struct sun4i_codec *scodec, + struct snd_pcm_hw_params *params, + unsigned int hwrate) +{ + u32 val; /* Set DAC sample rate */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, @@ -344,6 +426,34 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, return 0; } +static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + unsigned long clk_freq; + int hwrate; + + clk_freq = sun4i_codec_get_mod_freq(params); + if (!clk_freq) + return -EINVAL; + + if (clk_set_rate(scodec->clk_module, clk_freq)) + return -EINVAL; + + hwrate = sun4i_codec_get_hw_rate(params); + if (hwrate < 0) + return hwrate; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return sun4i_codec_hw_params_playback(scodec, params, + hwrate); + + return sun4i_codec_hw_params_capture(scodec, params, + hwrate); +} + static int sun4i_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -394,6 +504,20 @@ static struct snd_soc_dai_driver sun4i_codec_dai = { SNDRV_PCM_FMTBIT_S32_LE, .sig_bits = 24, }, + .capture = { + .stream_name = "Codec Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, }; /*** Codec ***/ @@ -429,11 +553,22 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { }; static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { + /* Digital parts of the ADCs */ + SND_SOC_DAPM_SUPPLY("ADC", SUN4I_CODEC_ADC_FIFOC, + SUN4I_CODEC_ADC_FIFOC_EN_AD, 0, + NULL, 0), + /* Digital parts of the DACs */ SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC, SUN4I_CODEC_DAC_DPC_EN_DA, 0, NULL, 0), + /* Analog parts of the ADCs */ + SND_SOC_DAPM_ADC("Left ADC", "Codec Capture", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_ADC_L_EN, 0), + SND_SOC_DAPM_ADC("Right ADC", "Codec Capture", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_ADC_R_EN, 0), + /* Analog parts of the DACs */ SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_DACAENL, 0), @@ -452,6 +587,14 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), + /* VMIC */ + SND_SOC_DAPM_SUPPLY("VMIC", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_VMICEN, 0, NULL, 0), + + /* Mic Pre-Amplifiers */ + SND_SOC_DAPM_PGA("MIC1 Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1EN, 0, NULL, 0), + /* Pre-Amplifier */ SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, SUN4I_CODEC_ADC_ACTL_PA_EN, 0, @@ -460,15 +603,19 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, &sun4i_codec_pa_mute), + SND_SOC_DAPM_INPUT("Mic1"), + SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), }; static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { - /* Left DAC Routes */ + /* Left ADC / DAC Routes */ + { "Left ADC", NULL, "ADC" }, { "Left DAC", NULL, "DAC" }, - /* Right DAC Routes */ + /* Right ADC / DAC Routes */ + { "Right ADC", NULL, "ADC" }, { "Right DAC", NULL, "DAC" }, /* Right Mixer Routes */ @@ -490,6 +637,12 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Mic1 Routes */ + { "Left ADC", NULL, "MIC1 Pre-Amplifier" }, + { "Right ADC", NULL, "MIC1 Pre-Amplifier" }, + { "MIC1 Pre-Amplifier", NULL, "Mic1"}, + { "Mic1", NULL, "VMIC" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { @@ -515,7 +668,7 @@ static int sun4i_codec_dai_probe(struct snd_soc_dai *dai) struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data, - NULL); + &scodec->capture_dma_data); return 0; } @@ -531,6 +684,14 @@ static struct snd_soc_dai_driver dummy_cpu_dai = { .formats = SUN4I_CODEC_FORMATS, .sig_bits = 24, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SUN4I_CODEC_RATES, + .formats = SUN4I_CODEC_FORMATS, + .sig_bits = 24, + }, }; static const struct regmap_config sun4i_codec_regmap_config = { @@ -638,6 +799,11 @@ static int sun4i_codec_probe(struct platform_device *pdev) scodec->playback_dma_data.maxburst = 4; scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + /* DMA configuration for RX FIFO */ + scodec->capture_dma_data.addr = res->start + SUN4I_CODEC_ADC_RXDATA; + scodec->capture_dma_data.maxburst = 4; + scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, &sun4i_codec_dai, 1); if (ret) { -- 2.6.3

2 3

[alsa-devel] [PATCH 1/2] ALSA: hda - Correct codec names for 14f1:50f1 and 14f1:50f3
by Takashi Iwai 01 Dec '15

01 Dec '15

The numbers aren't always linear, just like in the real world. Correct to the right numbers stated in the datasheet (although we can't trust the datasheet as well). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c8b8ef5246a6..89d9e8ccea51 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -972,9 +972,9 @@ static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto), - HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f1, "CX21722", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto), - HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f3, "CX21724", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto), -- 2.6.3

1 1

[alsa-devel] [PATCH 0/7] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 30 Nov '15

30 Nov '15

Hi, this is a patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. The initial draft was posted to alsa-devel as an RFC some weeks ago, and this is a revised version, posted to both i915 and alsa-devel for review. The current patchset is found in sound git tree test/hdmi-jack branch. Takashi Takashi Iwai (7): drm/i915: Remove superfluous NULL check drm/i915: Add get_eld audio component drm/i915: refactoring audio component functions ALSA: hda - Split ELD update code from hdmi_present_sense() ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Do zero-clear in snd_hdmi_parse_eld() itself ALSA: hda - Skip ELD notification during PM process drivers/gpu/drm/i915/intel_audio.c | 75 ++++++++++---- drivers/gpu/drm/i915/intel_drv.h | 1 + include/drm/i915_component.h | 3 + sound/pci/hda/hda_eld.c | 1 + sound/pci/hda/patch_hdmi.c | 195 ++++++++++++++++++++++++++----------- 5 files changed, 200 insertions(+), 75 deletions(-) -- 2.6.3

5 26

[alsa-devel] [PATCH] ASoC: rsnd: fix a possible NULL dereference
by LABBE Corentin 30 Nov '15

30 Nov '15

of_match_device could return NULL, and so cause a NULL pointer dereference later. Even if the probability of this case is very low, fixing it made static analyzers happy. Solving this with of_device_get_match_data made also code simplier. Signed-off-by: LABBE Corentin <clabbe.montjoie(a)gmail.com> --- sound/soc/sh/rcar/core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index deed48e..54cc44c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1204,7 +1204,6 @@ static int rsnd_probe(struct platform_device *pdev) struct rsnd_priv *priv; struct device *dev = &pdev->dev; struct rsnd_dai *rdai; - const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev); const struct rsnd_of_data *of_data; int (*probe_func[])(struct platform_device *pdev, const struct rsnd_of_data *of_data, @@ -1221,11 +1220,13 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; + of_data = of_device_get_match_data(dev); + if (!of_data) + return 1; info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info), GFP_KERNEL); if (!info) return -ENOMEM; - of_data = of_id->data; /* * init priv data -- 2.4.10

3 3

[alsa-devel] [PATCH] ASoC: rt5645: merge DMI tables of google projects
by Bard Liao 30 Nov '15

30 Nov '15

There are more and more google projects need to use DMI to get the platform data configuration. And those projects use the same configuration. To clean those redundant code, we define a general DMI for those projects with the same platform data configuration. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 79 +++++++++++------------------------------------ 1 file changed, 18 insertions(+), 61 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7b140cc..3e8d666 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3514,69 +3514,23 @@ static struct acpi_device_id rt5645_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); #endif -static struct rt5645_platform_data *rt5645_pdata; - -static struct rt5645_platform_data strago_platform_data = { +static struct rt5645_platform_data general_platform_data = { .dmic1_data_pin = RT5645_DMIC1_DISABLE, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, }; -static int strago_quirk_cb(const struct dmi_system_id *id) -{ - rt5645_pdata = &strago_platform_data; - - return 1; -} - static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", - .callback = strago_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Strago"), }, }, { - .ident = "Google Celes", - .callback = strago_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), - }, - }, - { - .ident = "Google Ultima", - .callback = strago_quirk_cb, + .ident = "Google Chrome", .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), - }, - }, - { - .ident = "Google Reks", - .callback = strago_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "Reks"), - }, - }, - { - .ident = "Google Edgar", - .callback = strago_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"), - }, - }, - { - .ident = "Google Wizpig", - .callback = strago_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"), - }, - }, - { - .ident = "Google Terra", - .callback = strago_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "Terra"), + DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, { } @@ -3589,17 +3543,9 @@ static struct rt5645_platform_data buddy_platform_data = { .jd_invert = true, }; -static int buddy_quirk_cb(const struct dmi_system_id *id) -{ - rt5645_pdata = &buddy_platform_data; - - return 1; -} - static struct dmi_system_id dmi_platform_intel_broadwell[] = { { .ident = "Chrome Buddy", - .callback = buddy_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), }, @@ -3607,6 +3553,16 @@ static struct dmi_system_id dmi_platform_intel_broadwell[] = { { } }; +static bool rt5645_check_dp(struct device *dev) +{ + if (device_property_present(dev, "realtek,in2-differential") || + device_property_present(dev, "realtek,dmic1-data-pin") || + device_property_present(dev, "realtek,dmic2-data-pin") || + device_property_present(dev, "realtek,jd-mode")) + return true; + + return false; +} static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { @@ -3641,11 +3597,12 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell) || - dmi_check_system(dmi_platform_intel_broadwell)) - rt5645->pdata = *rt5645_pdata; - else + else if (dmi_check_system(dmi_platform_intel_broadwell)) + rt5645->pdata = buddy_platform_data; + else if (rt5645_check_dp(&i2c->dev)) rt5645_parse_dt(rt5645, &i2c->dev); + else if (dmi_check_system(dmi_platform_intel_braswell)) + rt5645->pdata = general_platform_data; rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect", GPIOD_IN); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: Add support for DA7217 and DA7218 audio codecs
by Adam Thomson 30 Nov '15

30 Nov '15

This patch set adds DT and ASoC codec driver support for DA7217 and DA7218. Changes are based against v4.4-rc2 kernel version. Changes in v4: - DT binding updates for unit suffixes (hz, us) at Rob Herring's request. Changes in v3: - At request of Rob Herring, added units suffix to applicable binding names. - Add changes for codec Kconfig and Makefile which were somehow missed from previous patches. Changes in v2: - Removed Internal LDO voltage selection from DT - BiQuad filter coefficients removed from DT, no programmed using mixer controls. - ALC calibration no longer bypasses regmap cache and instead saves & restores registers around procedure, to avoid issues with concurrent regmap access. - Removed all unnecessary default settings for mixers in probe. - Reporting of mic level detection event to user-space, through input VOICECOMMAND key event, removed as not intended for this purpose. Alternative solution required as follow up. Event masked off until solution in place. - Additional error checking & loggin around mclk usage, in sysclk and bias_level functions. Adam Thomson (2): ASoC: da7218: Add bindings documentation for DA7218 audio codec ASoC: codecs: Add da7218 codec driver Documentation/devicetree/bindings/sound/da7218.txt | 104 + include/sound/da7218.h | 109 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7218.c | 3305 ++++++++++++++++++++ sound/soc/codecs/da7218.h | 1414 +++++++++ 6 files changed, 4938 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7218.txt create mode 100644 include/sound/da7218.h create mode 100644 sound/soc/codecs/da7218.c create mode 100644 sound/soc/codecs/da7218.h -- 1.9.3

3 4

[alsa-devel] Applied "ASoC: da7218: Add da7218 codec driver" to the asoc tree
by Mark Brown 30 Nov '15

30 Nov '15

The patch ASoC: da7218: Add da7218 codec driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4d50934abd2261fd467320d52c470efff309fd74 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Wed, 25 Nov 2015 14:24:38 +0000 Subject: [PATCH] ASoC: da7218: Add da7218 codec driver This adds support for DA7217 and DA7218 audio codecs. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/da7218.h | 109 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7218.c | 3305 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7218.h | 1414 +++++++++++++++++++ 5 files changed, 4834 insertions(+) create mode 100644 include/sound/da7218.h create mode 100644 sound/soc/codecs/da7218.c create mode 100644 sound/soc/codecs/da7218.h diff --git a/include/sound/da7218.h b/include/sound/da7218.h new file mode 100644 index 000000000000..0dbb818ac116 --- /dev/null +++ b/include/sound/da7218.h @@ -0,0 +1,109 @@ +/* + * da7218.h - DA7218 ASoC Codec Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _DA7218_PDATA_H +#define _DA7218_PDATA_H + +/* Mic Bias */ +enum da7218_micbias_voltage { + DA7218_MICBIAS_1_2V = -1, + DA7218_MICBIAS_1_6V, + DA7218_MICBIAS_1_8V, + DA7218_MICBIAS_2_0V, + DA7218_MICBIAS_2_2V, + DA7218_MICBIAS_2_4V, + DA7218_MICBIAS_2_6V, + DA7218_MICBIAS_2_8V, + DA7218_MICBIAS_3_0V, +}; + +enum da7218_mic_amp_in_sel { + DA7218_MIC_AMP_IN_SEL_DIFF = 0, + DA7218_MIC_AMP_IN_SEL_SE_P, + DA7218_MIC_AMP_IN_SEL_SE_N, +}; + +/* DMIC */ +enum da7218_dmic_data_sel { + DA7218_DMIC_DATA_LRISE_RFALL = 0, + DA7218_DMIC_DATA_LFALL_RRISE, +}; + +enum da7218_dmic_samplephase { + DA7218_DMIC_SAMPLE_ON_CLKEDGE = 0, + DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE, +}; + +enum da7218_dmic_clk_rate { + DA7218_DMIC_CLK_3_0MHZ = 0, + DA7218_DMIC_CLK_1_5MHZ, +}; + +/* Headphone Detect */ +enum da7218_hpldet_jack_rate { + DA7218_HPLDET_JACK_RATE_5US = 0, + DA7218_HPLDET_JACK_RATE_10US, + DA7218_HPLDET_JACK_RATE_20US, + DA7218_HPLDET_JACK_RATE_40US, + DA7218_HPLDET_JACK_RATE_80US, + DA7218_HPLDET_JACK_RATE_160US, + DA7218_HPLDET_JACK_RATE_320US, + DA7218_HPLDET_JACK_RATE_640US, +}; + +enum da7218_hpldet_jack_debounce { + DA7218_HPLDET_JACK_DEBOUNCE_OFF = 0, + DA7218_HPLDET_JACK_DEBOUNCE_2, + DA7218_HPLDET_JACK_DEBOUNCE_3, + DA7218_HPLDET_JACK_DEBOUNCE_4, +}; + +enum da7218_hpldet_jack_thr { + DA7218_HPLDET_JACK_THR_84PCT = 0, + DA7218_HPLDET_JACK_THR_88PCT, + DA7218_HPLDET_JACK_THR_92PCT, + DA7218_HPLDET_JACK_THR_96PCT, +}; + +struct da7218_hpldet_pdata { + enum da7218_hpldet_jack_rate jack_rate; + enum da7218_hpldet_jack_debounce jack_debounce; + enum da7218_hpldet_jack_thr jack_thr; + bool comp_inv; + bool hyst; + bool discharge; +}; + +struct da7218_pdata { + /* Mic */ + enum da7218_micbias_voltage micbias1_lvl; + enum da7218_micbias_voltage micbias2_lvl; + enum da7218_mic_amp_in_sel mic1_amp_in_sel; + enum da7218_mic_amp_in_sel mic2_amp_in_sel; + + /* DMIC */ + enum da7218_dmic_data_sel dmic1_data_sel; + enum da7218_dmic_data_sel dmic2_data_sel; + enum da7218_dmic_samplephase dmic1_samplephase; + enum da7218_dmic_samplephase dmic2_samplephase; + enum da7218_dmic_clk_rate dmic1_clk_rate; + enum da7218_dmic_clk_rate dmic2_clk_rate; + + /* HP Diff Supply - DA7217 only */ + bool hp_diff_single_supply; + + /* HP Detect - DA7218 only */ + struct da7218_hpldet_pdata *hpldet_pdata; +}; + +#endif /* _DA7218_PDATA_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cfdafc4c11ea..6dd87ee2c463 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -58,6 +58,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C + select SND_SOC_DA7218 if I2C select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C @@ -439,6 +440,9 @@ config SND_SOC_DA7210 config SND_SOC_DA7213 tristate +config SND_SOC_DA7218 + tristate + config SND_SOC_DA7219 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f632fc42f59f..02057d6582c4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -50,6 +50,7 @@ snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o +snd-soc-da7218-objs := da7218.o snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o @@ -245,6 +246,7 @@ obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o +obj-$(CONFIG_SND_SOC_DA7218) += snd-soc-da7218.o obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c new file mode 100644 index 000000000000..ed0c9a26065b --- /dev/null +++ b/sound/soc/codecs/da7218.c @@ -0,0 +1,3305 @@ +/* + * da7218.c - DA7218 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include <sound/da7218.h> +#include "da7218.h" + + +/* + * TLVs and Enums + */ + +/* Input TLVs */ +static const DECLARE_TLV_DB_SCALE(da7218_mic_gain_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7218_in_dig_gain_tlv, -8325, 75, 0); +static const DECLARE_TLV_DB_SCALE(da7218_ags_trigger_tlv, -9000, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_ags_att_max_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7218_alc_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_alc_ana_gain_tlv, 0, 600, 0); + +/* Input/Output TLVs */ +static const DECLARE_TLV_DB_SCALE(da7218_dmix_gain_tlv, -4200, 150, 0); + +/* Output TLVs */ +static const DECLARE_TLV_DB_SCALE(da7218_dgs_trigger_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7218_dgs_anticlip_tlv, -4200, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_dgs_signal_tlv, -9000, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_out_eq_band_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7218_out_dig_gain_tlv, -8325, 75, 0); +static const DECLARE_TLV_DB_SCALE(da7218_dac_ng_threshold_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7218_mixout_gain_tlv, -100, 50, 0); +static const DECLARE_TLV_DB_SCALE(da7218_hp_gain_tlv, -5700, 150, 0); + +/* Input Enums */ +static const char * const da7218_alc_attack_rate_txt[] = { + "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs", + "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs", + "30024/fs", +}; + +static const struct soc_enum da7218_alc_attack_rate = + SOC_ENUM_SINGLE(DA7218_ALC_CTRL2, DA7218_ALC_ATTACK_SHIFT, + DA7218_ALC_ATTACK_MAX, da7218_alc_attack_rate_txt); + +static const char * const da7218_alc_release_rate_txt[] = { + "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs", + "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs", +}; + +static const struct soc_enum da7218_alc_release_rate = + SOC_ENUM_SINGLE(DA7218_ALC_CTRL2, DA7218_ALC_RELEASE_SHIFT, + DA7218_ALC_RELEASE_MAX, da7218_alc_release_rate_txt); + +static const char * const da7218_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7218_alc_hold_time = + SOC_ENUM_SINGLE(DA7218_ALC_CTRL3, DA7218_ALC_HOLD_SHIFT, + DA7218_ALC_HOLD_MAX, da7218_alc_hold_time_txt); + +static const char * const da7218_alc_anticlip_step_txt[] = { + "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs", +}; + +static const struct soc_enum da7218_alc_anticlip_step = + SOC_ENUM_SINGLE(DA7218_ALC_ANTICLIP_CTRL, + DA7218_ALC_ANTICLIP_STEP_SHIFT, + DA7218_ALC_ANTICLIP_STEP_MAX, + da7218_alc_anticlip_step_txt); + +static const char * const da7218_integ_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7218_integ_attack_rate = + SOC_ENUM_SINGLE(DA7218_ENV_TRACK_CTRL, DA7218_INTEG_ATTACK_SHIFT, + DA7218_INTEG_MAX, da7218_integ_rate_txt); + +static const struct soc_enum da7218_integ_release_rate = + SOC_ENUM_SINGLE(DA7218_ENV_TRACK_CTRL, DA7218_INTEG_RELEASE_SHIFT, + DA7218_INTEG_MAX, da7218_integ_rate_txt); + +/* Input/Output Enums */ +static const char * const da7218_gain_ramp_rate_txt[] = { + "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8", + "Nominal Rate / 16", +}; + +static const struct soc_enum da7218_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7218_GAIN_RAMP_CTRL, DA7218_GAIN_RAMP_RATE_SHIFT, + DA7218_GAIN_RAMP_RATE_MAX, da7218_gain_ramp_rate_txt); + +static const char * const da7218_hpf_mode_txt[] = { + "Disabled", "Audio", "Voice", +}; + +static const unsigned int da7218_hpf_mode_val[] = { + DA7218_HPF_DISABLED, DA7218_HPF_AUDIO_EN, DA7218_HPF_VOICE_EN, +}; + +static const struct soc_enum da7218_in1_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7218_IN_1_HPF_FILTER_CTRL, + DA7218_HPF_MODE_SHIFT, DA7218_HPF_MODE_MASK, + DA7218_HPF_MODE_MAX, da7218_hpf_mode_txt, + da7218_hpf_mode_val); + +static const struct soc_enum da7218_in2_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7218_IN_2_HPF_FILTER_CTRL, + DA7218_HPF_MODE_SHIFT, DA7218_HPF_MODE_MASK, + DA7218_HPF_MODE_MAX, da7218_hpf_mode_txt, + da7218_hpf_mode_val); + +static const struct soc_enum da7218_out1_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7218_OUT_1_HPF_FILTER_CTRL, + DA7218_HPF_MODE_SHIFT, DA7218_HPF_MODE_MASK, + DA7218_HPF_MODE_MAX, da7218_hpf_mode_txt, + da7218_hpf_mode_val); + +static const char * const da7218_audio_hpf_corner_txt[] = { + "2Hz", "4Hz", "8Hz", "16Hz", +}; + +static const struct soc_enum da7218_in1_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7218_IN_1_HPF_FILTER_CTRL, + DA7218_IN_1_AUDIO_HPF_CORNER_SHIFT, + DA7218_AUDIO_HPF_CORNER_MAX, + da7218_audio_hpf_corner_txt); + +static const struct soc_enum da7218_in2_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7218_IN_2_HPF_FILTER_CTRL, + DA7218_IN_2_AUDIO_HPF_CORNER_SHIFT, + DA7218_AUDIO_HPF_CORNER_MAX, + da7218_audio_hpf_corner_txt); + +static const struct soc_enum da7218_out1_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7218_OUT_1_HPF_FILTER_CTRL, + DA7218_OUT_1_AUDIO_HPF_CORNER_SHIFT, + DA7218_AUDIO_HPF_CORNER_MAX, + da7218_audio_hpf_corner_txt); + +static const char * const da7218_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz", +}; + +static const struct soc_enum da7218_in1_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7218_IN_1_HPF_FILTER_CTRL, + DA7218_IN_1_VOICE_HPF_CORNER_SHIFT, + DA7218_VOICE_HPF_CORNER_MAX, + da7218_voice_hpf_corner_txt); + +static const struct soc_enum da7218_in2_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7218_IN_2_HPF_FILTER_CTRL, + DA7218_IN_2_VOICE_HPF_CORNER_SHIFT, + DA7218_VOICE_HPF_CORNER_MAX, + da7218_voice_hpf_corner_txt); + +static const struct soc_enum da7218_out1_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7218_OUT_1_HPF_FILTER_CTRL, + DA7218_OUT_1_VOICE_HPF_CORNER_SHIFT, + DA7218_VOICE_HPF_CORNER_MAX, + da7218_voice_hpf_corner_txt); + +static const char * const da7218_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static const struct soc_enum da7218_tonegen_dtmf_key = + SOC_ENUM_SINGLE(DA7218_TONE_GEN_CFG1, DA7218_DTMF_REG_SHIFT, + DA7218_DTMF_REG_MAX, da7218_tonegen_dtmf_key_txt); + +static const char * const da7218_tonegen_swg_sel_txt[] = { + "Sum", "SWG1", "SWG2", "SWG1_1-Cos" +}; + +static const struct soc_enum da7218_tonegen_swg_sel = + SOC_ENUM_SINGLE(DA7218_TONE_GEN_CFG2, DA7218_SWG_SEL_SHIFT, + DA7218_SWG_SEL_MAX, da7218_tonegen_swg_sel_txt); + +/* Output Enums */ +static const char * const da7218_dgs_rise_coeff_txt[] = { + "1/1", "1/16", "1/64", "1/256", "1/1024", "1/4096", "1/16384", +}; + +static const struct soc_enum da7218_dgs_rise_coeff = + SOC_ENUM_SINGLE(DA7218_DGS_RISE_FALL, DA7218_DGS_RISE_COEFF_SHIFT, + DA7218_DGS_RISE_COEFF_MAX, da7218_dgs_rise_coeff_txt); + +static const char * const da7218_dgs_fall_coeff_txt[] = { + "1/4", "1/16", "1/64", "1/256", "1/1024", "1/4096", "1/16384", "1/65536", +}; + +static const struct soc_enum da7218_dgs_fall_coeff = + SOC_ENUM_SINGLE(DA7218_DGS_RISE_FALL, DA7218_DGS_FALL_COEFF_SHIFT, + DA7218_DGS_FALL_COEFF_MAX, da7218_dgs_fall_coeff_txt); + +static const char * const da7218_dac_ng_setup_time_txt[] = { + "256 Samples", "512 Samples", "1024 Samples", "2048 Samples" +}; + +static const struct soc_enum da7218_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7218_DAC_NG_SETUP_TIME, + DA7218_DAC_NG_SETUP_TIME_SHIFT, + DA7218_DAC_NG_SETUP_TIME_MAX, + da7218_dac_ng_setup_time_txt); + +static const char * const da7218_dac_ng_rampup_txt[] = { + "0.22ms/dB", "0.0138ms/dB" +}; + +static const struct soc_enum da7218_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7218_DAC_NG_SETUP_TIME, + DA7218_DAC_NG_RAMPUP_RATE_SHIFT, + DA7218_DAC_NG_RAMPUP_RATE_MAX, + da7218_dac_ng_rampup_txt); + +static const char * const da7218_dac_ng_rampdown_txt[] = { + "0.88ms/dB", "14.08ms/dB" +}; + +static const struct soc_enum da7218_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7218_DAC_NG_SETUP_TIME, + DA7218_DAC_NG_RAMPDN_RATE_SHIFT, + DA7218_DAC_NG_RAMPDN_RATE_MAX, + da7218_dac_ng_rampdown_txt); + +static const char * const da7218_cp_mchange_txt[] = { + "Largest Volume", "DAC Volume", "Signal Magnitude" +}; + +static const unsigned int da7218_cp_mchange_val[] = { + DA7218_CP_MCHANGE_LARGEST_VOL, DA7218_CP_MCHANGE_DAC_VOL, + DA7218_CP_MCHANGE_SIG_MAG +}; + +static const struct soc_enum da7218_cp_mchange = + SOC_VALUE_ENUM_SINGLE(DA7218_CP_CTRL, DA7218_CP_MCHANGE_SHIFT, + DA7218_CP_MCHANGE_REL_MASK, DA7218_CP_MCHANGE_MAX, + da7218_cp_mchange_txt, da7218_cp_mchange_val); + +static const char * const da7218_cp_fcontrol_txt[] = { + "1MHz", "500KHz", "250KHz", "125KHz", "63KHz", "0KHz" +}; + +static const struct soc_enum da7218_cp_fcontrol = + SOC_ENUM_SINGLE(DA7218_CP_DELAY, DA7218_CP_FCONTROL_SHIFT, + DA7218_CP_FCONTROL_MAX, da7218_cp_fcontrol_txt); + +static const char * const da7218_cp_tau_delay_txt[] = { + "0ms", "2ms", "4ms", "16ms", "64ms", "128ms", "256ms", "512ms" +}; + +static const struct soc_enum da7218_cp_tau_delay = + SOC_ENUM_SINGLE(DA7218_CP_DELAY, DA7218_CP_TAU_DELAY_SHIFT, + DA7218_CP_TAU_DELAY_MAX, da7218_cp_tau_delay_txt); + +/* + * Control Functions + */ + +/* ALC */ +static void da7218_alc_calib(struct snd_soc_codec *codec) +{ + u8 mic_1_ctrl, mic_2_ctrl; + u8 mixin_1_ctrl, mixin_2_ctrl; + u8 in_1l_filt_ctrl, in_1r_filt_ctrl, in_2l_filt_ctrl, in_2r_filt_ctrl; + u8 in_1_hpf_ctrl, in_2_hpf_ctrl; + u8 calib_ctrl; + int i = 0; + bool calibrated = false; + + /* Save current state of MIC control registers */ + mic_1_ctrl = snd_soc_read(codec, DA7218_MIC_1_CTRL); + mic_2_ctrl = snd_soc_read(codec, DA7218_MIC_2_CTRL); + + /* Save current state of input mixer control registers */ + mixin_1_ctrl = snd_soc_read(codec, DA7218_MIXIN_1_CTRL); + mixin_2_ctrl = snd_soc_read(codec, DA7218_MIXIN_2_CTRL); + + /* Save current state of input filter control registers */ + in_1l_filt_ctrl = snd_soc_read(codec, DA7218_IN_1L_FILTER_CTRL); + in_1r_filt_ctrl = snd_soc_read(codec, DA7218_IN_1R_FILTER_CTRL); + in_2l_filt_ctrl = snd_soc_read(codec, DA7218_IN_2L_FILTER_CTRL); + in_2r_filt_ctrl = snd_soc_read(codec, DA7218_IN_2R_FILTER_CTRL); + + /* Save current state of input HPF control registers */ + in_1_hpf_ctrl = snd_soc_read(codec, DA7218_IN_1_HPF_FILTER_CTRL); + in_2_hpf_ctrl = snd_soc_read(codec, DA7218_IN_2_HPF_FILTER_CTRL); + + /* Enable then Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7218_MIC_1_CTRL, DA7218_MIC_1_AMP_EN_MASK, + DA7218_MIC_1_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_MIC_2_CTRL, DA7218_MIC_2_AMP_EN_MASK, + DA7218_MIC_2_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_MIC_1_CTRL, + DA7218_MIC_1_AMP_MUTE_EN_MASK, + DA7218_MIC_1_AMP_MUTE_EN_MASK); + snd_soc_update_bits(codec, DA7218_MIC_2_CTRL, + DA7218_MIC_2_AMP_MUTE_EN_MASK, + DA7218_MIC_2_AMP_MUTE_EN_MASK); + + /* Enable input mixers unmuted */ + snd_soc_update_bits(codec, DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_AMP_EN_MASK | + DA7218_MIXIN_1_AMP_MUTE_EN_MASK, + DA7218_MIXIN_1_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_AMP_EN_MASK | + DA7218_MIXIN_2_AMP_MUTE_EN_MASK, + DA7218_MIXIN_2_AMP_EN_MASK); + + /* Enable input filters unmuted */ + snd_soc_update_bits(codec, DA7218_IN_1L_FILTER_CTRL, + DA7218_IN_1L_FILTER_EN_MASK | + DA7218_IN_1L_MUTE_EN_MASK, + DA7218_IN_1L_FILTER_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_1R_FILTER_CTRL, + DA7218_IN_1R_FILTER_EN_MASK | + DA7218_IN_1R_MUTE_EN_MASK, + DA7218_IN_1R_FILTER_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_2L_FILTER_CTRL, + DA7218_IN_2L_FILTER_EN_MASK | + DA7218_IN_2L_MUTE_EN_MASK, + DA7218_IN_2L_FILTER_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_2R_FILTER_CTRL, + DA7218_IN_2R_FILTER_EN_MASK | + DA7218_IN_2R_MUTE_EN_MASK, + DA7218_IN_2R_FILTER_EN_MASK); + + /* + * Make sure input HPFs voice mode is disabled, otherwise for sampling + * rates above 32KHz the ADC signals will be stopped and will cause + * calibration to lock up. + */ + snd_soc_update_bits(codec, DA7218_IN_1_HPF_FILTER_CTRL, + DA7218_IN_1_VOICE_EN_MASK, 0); + snd_soc_update_bits(codec, DA7218_IN_2_HPF_FILTER_CTRL, + DA7218_IN_2_VOICE_EN_MASK, 0); + + /* Perform auto calibration */ + snd_soc_update_bits(codec, DA7218_CALIB_CTRL, DA7218_CALIB_AUTO_EN_MASK, + DA7218_CALIB_AUTO_EN_MASK); + do { + calib_ctrl = snd_soc_read(codec, DA7218_CALIB_CTRL); + if (calib_ctrl & DA7218_CALIB_AUTO_EN_MASK) { + ++i; + usleep_range(DA7218_ALC_CALIB_DELAY_MIN, + DA7218_ALC_CALIB_DELAY_MAX); + } else { + calibrated = true; + } + + } while ((i < DA7218_ALC_CALIB_MAX_TRIES) && (!calibrated)); + + /* If auto calibration fails, disable DC offset, hybrid ALC */ + if ((!calibrated) || (calib_ctrl & DA7218_CALIB_OVERFLOW_MASK)) { + dev_warn(codec->dev, + "ALC auto calibration failed - %s\n", + (calibrated) ? "overflow" : "timeout"); + snd_soc_update_bits(codec, DA7218_CALIB_CTRL, + DA7218_CALIB_OFFSET_EN_MASK, 0); + snd_soc_update_bits(codec, DA7218_ALC_CTRL1, + DA7218_ALC_SYNC_MODE_MASK, 0); + + } else { + /* Enable DC offset cancellation */ + snd_soc_update_bits(codec, DA7218_CALIB_CTRL, + DA7218_CALIB_OFFSET_EN_MASK, + DA7218_CALIB_OFFSET_EN_MASK); + + /* Enable ALC hybrid mode */ + snd_soc_update_bits(codec, DA7218_ALC_CTRL1, + DA7218_ALC_SYNC_MODE_MASK, + DA7218_ALC_SYNC_MODE_CH1 | + DA7218_ALC_SYNC_MODE_CH2); + } + + /* Restore input HPF control registers to original states */ + snd_soc_write(codec, DA7218_IN_1_HPF_FILTER_CTRL, in_1_hpf_ctrl); + snd_soc_write(codec, DA7218_IN_2_HPF_FILTER_CTRL, in_2_hpf_ctrl); + + /* Restore input filter control registers to original states */ + snd_soc_write(codec, DA7218_IN_1L_FILTER_CTRL, in_1l_filt_ctrl); + snd_soc_write(codec, DA7218_IN_1R_FILTER_CTRL, in_1r_filt_ctrl); + snd_soc_write(codec, DA7218_IN_2L_FILTER_CTRL, in_2l_filt_ctrl); + snd_soc_write(codec, DA7218_IN_2R_FILTER_CTRL, in_2r_filt_ctrl); + + /* Restore input mixer control registers to original state */ + snd_soc_write(codec, DA7218_MIXIN_1_CTRL, mixin_1_ctrl); + snd_soc_write(codec, DA7218_MIXIN_2_CTRL, mixin_2_ctrl); + + /* Restore MIC control registers to original states */ + snd_soc_write(codec, DA7218_MIC_1_CTRL, mic_1_ctrl); + snd_soc_write(codec, DA7218_MIC_2_CTRL, mic_2_ctrl); +} + +static int da7218_mixin_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + /* + * If ALC in operation and value of control has been updated, + * make sure calibrated offsets are updated. + */ + if ((ret == 1) && (da7218->alc_en)) + da7218_alc_calib(codec); + + return ret; +} + +static int da7218_alc_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *) kcontrol->private_value; + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + unsigned int lvalue = ucontrol->value.integer.value[0]; + unsigned int rvalue = ucontrol->value.integer.value[1]; + unsigned int lshift = mc->shift; + unsigned int rshift = mc->rshift; + unsigned int mask = (mc->max << lshift) | (mc->max << rshift); + + /* Force ALC offset calibration if enabling ALC */ + if ((lvalue || rvalue) && (!da7218->alc_en)) + da7218_alc_calib(codec); + + /* Update bits to detail which channels are enabled/disabled */ + da7218->alc_en &= ~mask; + da7218->alc_en |= (lvalue << lshift) | (rvalue << rshift); + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* ToneGen */ +static int da7218_tonegen_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to host endianness here. + */ + ret = regmap_raw_read(da7218->regmap, reg, &val, 2); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = le16_to_cpu(val); + + return 0; +} + +static int da7218_tonegen_freq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to little endian here to align with + * HW registers. + */ + val = cpu_to_le16(ucontrol->value.integer.value[0]); + + return regmap_raw_write(da7218->regmap, reg, &val, 2); +} + +static int da7218_mic_lvl_det_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int lvalue = ucontrol->value.integer.value[0]; + unsigned int rvalue = ucontrol->value.integer.value[1]; + unsigned int lshift = mixer_ctrl->shift; + unsigned int rshift = mixer_ctrl->rshift; + unsigned int mask = (mixer_ctrl->max << lshift) | + (mixer_ctrl->max << rshift); + da7218->mic_lvl_det_en &= ~mask; + da7218->mic_lvl_det_en |= (lvalue << lshift) | (rvalue << rshift); + + /* + * Here we only enable the feature on paths which are already + * powered. If a channel is enabled here for level detect, but that path + * isn't powered, then the channel will actually be enabled when we do + * power the path (IN_FILTER widget events). This handling avoids + * unwanted level detect events. + */ + return snd_soc_write(codec, mixer_ctrl->reg, + (da7218->in_filt_en & da7218->mic_lvl_det_en)); +} + +static int da7218_mic_lvl_det_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int lshift = mixer_ctrl->shift; + unsigned int rshift = mixer_ctrl->rshift; + unsigned int lmask = (mixer_ctrl->max << lshift); + unsigned int rmask = (mixer_ctrl->max << rshift); + + ucontrol->value.integer.value[0] = + (da7218->mic_lvl_det_en & lmask) >> lshift; + ucontrol->value.integer.value[1] = + (da7218->mic_lvl_det_en & rmask) >> rshift; + + return 0; +} + +static int da7218_biquad_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + + /* Determine which BiQuads we're setting based on size of config data */ + switch (bytes_ext->max) { + case DA7218_OUT_1_BIQ_5STAGE_CFG_SIZE: + memcpy(ucontrol->value.bytes.data, da7218->biq_5stage_coeff, + bytes_ext->max); + break; + case DA7218_SIDETONE_BIQ_3STAGE_CFG_SIZE: + memcpy(ucontrol->value.bytes.data, da7218->stbiq_3stage_coeff, + bytes_ext->max); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int da7218_biquad_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + u8 reg, out_filt1l; + u8 cfg[DA7218_BIQ_CFG_SIZE]; + int i; + + /* + * Determine which BiQuads we're setting based on size of config data, + * and stored the data for use by get function. + */ + switch (bytes_ext->max) { + case DA7218_OUT_1_BIQ_5STAGE_CFG_SIZE: + reg = DA7218_OUT_1_BIQ_5STAGE_DATA; + memcpy(da7218->biq_5stage_coeff, ucontrol->value.bytes.data, + bytes_ext->max); + break; + case DA7218_SIDETONE_BIQ_3STAGE_CFG_SIZE: + reg = DA7218_SIDETONE_BIQ_3STAGE_DATA; + memcpy(da7218->stbiq_3stage_coeff, ucontrol->value.bytes.data, + bytes_ext->max); + break; + default: + return -EINVAL; + } + + /* Make sure at least out filter1 enabled to allow programming */ + out_filt1l = snd_soc_read(codec, DA7218_OUT_1L_FILTER_CTRL); + snd_soc_write(codec, DA7218_OUT_1L_FILTER_CTRL, + out_filt1l | DA7218_OUT_1L_FILTER_EN_MASK); + + for (i = 0; i < bytes_ext->max; ++i) { + cfg[DA7218_BIQ_CFG_DATA] = ucontrol->value.bytes.data[i]; + cfg[DA7218_BIQ_CFG_ADDR] = i; + regmap_raw_write(da7218->regmap, reg, cfg, DA7218_BIQ_CFG_SIZE); + } + + /* Restore filter to previous setting */ + snd_soc_write(codec, DA7218_OUT_1L_FILTER_CTRL, out_filt1l); + + return 0; +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7218_snd_controls[] = { + /* Mics */ + SOC_SINGLE_TLV("Mic1 Volume", DA7218_MIC_1_GAIN, + DA7218_MIC_1_AMP_GAIN_SHIFT, DA7218_MIC_AMP_GAIN_MAX, + DA7218_NO_INVERT, da7218_mic_gain_tlv), + SOC_SINGLE("Mic1 Switch", DA7218_MIC_1_CTRL, + DA7218_MIC_1_AMP_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE_TLV("Mic2 Volume", DA7218_MIC_2_GAIN, + DA7218_MIC_2_AMP_GAIN_SHIFT, DA7218_MIC_AMP_GAIN_MAX, + DA7218_NO_INVERT, da7218_mic_gain_tlv), + SOC_SINGLE("Mic2 Switch", DA7218_MIC_2_CTRL, + DA7218_MIC_2_AMP_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + + /* Mixer Input */ + SOC_SINGLE_EXT_TLV("Mixin1 Volume", DA7218_MIXIN_1_GAIN, + DA7218_MIXIN_1_AMP_GAIN_SHIFT, + DA7218_MIXIN_AMP_GAIN_MAX, DA7218_NO_INVERT, + snd_soc_get_volsw, da7218_mixin_gain_put, + da7218_mixin_gain_tlv), + SOC_SINGLE("Mixin1 Switch", DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_AMP_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE("Mixin1 Gain Ramp Switch", DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_AMP_RAMP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("Mixin1 ZC Gain Switch", DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_AMP_ZC_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE_EXT_TLV("Mixin2 Volume", DA7218_MIXIN_2_GAIN, + DA7218_MIXIN_2_AMP_GAIN_SHIFT, + DA7218_MIXIN_AMP_GAIN_MAX, DA7218_NO_INVERT, + snd_soc_get_volsw, da7218_mixin_gain_put, + da7218_mixin_gain_tlv), + SOC_SINGLE("Mixin2 Switch", DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_AMP_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE("Mixin2 Gain Ramp Switch", DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_AMP_RAMP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("Mixin2 ZC Gain Switch", DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_AMP_ZC_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + + /* ADCs */ + SOC_SINGLE("ADC1 AAF Switch", DA7218_ADC_1_CTRL, + DA7218_ADC_1_AAF_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("ADC2 AAF Switch", DA7218_ADC_2_CTRL, + DA7218_ADC_2_AAF_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("ADC LP Mode Switch", DA7218_ADC_MODE, + DA7218_ADC_LP_MODE_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + + /* Input Filters */ + SOC_SINGLE_TLV("In Filter1L Volume", DA7218_IN_1L_GAIN, + DA7218_IN_1L_DIGITAL_GAIN_SHIFT, + DA7218_IN_DIGITAL_GAIN_MAX, DA7218_NO_INVERT, + da7218_in_dig_gain_tlv), + SOC_SINGLE("In Filter1L Switch", DA7218_IN_1L_FILTER_CTRL, + DA7218_IN_1L_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE("In Filter1L Gain Ramp Switch", DA7218_IN_1L_FILTER_CTRL, + DA7218_IN_1L_RAMP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE_TLV("In Filter1R Volume", DA7218_IN_1R_GAIN, + DA7218_IN_1R_DIGITAL_GAIN_SHIFT, + DA7218_IN_DIGITAL_GAIN_MAX, DA7218_NO_INVERT, + da7218_in_dig_gain_tlv), + SOC_SINGLE("In Filter1R Switch", DA7218_IN_1R_FILTER_CTRL, + DA7218_IN_1R_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE("In Filter1R Gain Ramp Switch", + DA7218_IN_1R_FILTER_CTRL, DA7218_IN_1R_RAMP_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), + SOC_SINGLE_TLV("In Filter2L Volume", DA7218_IN_2L_GAIN, + DA7218_IN_2L_DIGITAL_GAIN_SHIFT, + DA7218_IN_DIGITAL_GAIN_MAX, DA7218_NO_INVERT, + da7218_in_dig_gain_tlv), + SOC_SINGLE("In Filter2L Switch", DA7218_IN_2L_FILTER_CTRL, + DA7218_IN_2L_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE("In Filter2L Gain Ramp Switch", DA7218_IN_2L_FILTER_CTRL, + DA7218_IN_2L_RAMP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE_TLV("In Filter2R Volume", DA7218_IN_2R_GAIN, + DA7218_IN_2R_DIGITAL_GAIN_SHIFT, + DA7218_IN_DIGITAL_GAIN_MAX, DA7218_NO_INVERT, + da7218_in_dig_gain_tlv), + SOC_SINGLE("In Filter2R Switch", DA7218_IN_2R_FILTER_CTRL, + DA7218_IN_2R_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_SINGLE("In Filter2R Gain Ramp Switch", + DA7218_IN_2R_FILTER_CTRL, DA7218_IN_2R_RAMP_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), + + /* AGS */ + SOC_SINGLE_TLV("AGS Trigger", DA7218_AGS_TRIGGER, + DA7218_AGS_TRIGGER_SHIFT, DA7218_AGS_TRIGGER_MAX, + DA7218_INVERT, da7218_ags_trigger_tlv), + SOC_SINGLE_TLV("AGS Max Attenuation", DA7218_AGS_ATT_MAX, + DA7218_AGS_ATT_MAX_SHIFT, DA7218_AGS_ATT_MAX_MAX, + DA7218_NO_INVERT, da7218_ags_att_max_tlv), + SOC_SINGLE("AGS Anticlip Switch", DA7218_AGS_ANTICLIP_CTRL, + DA7218_AGS_ANTICLIP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("AGS Channel1 Switch", DA7218_AGS_ENABLE, + DA7218_AGS_ENABLE_CHAN1_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("AGS Channel2 Switch", DA7218_AGS_ENABLE, + DA7218_AGS_ENABLE_CHAN2_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + + /* ALC */ + SOC_ENUM("ALC Attack Rate", da7218_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7218_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7218_alc_hold_time), + SOC_SINGLE_TLV("ALC Noise Threshold", DA7218_ALC_NOISE, + DA7218_ALC_NOISE_SHIFT, DA7218_ALC_THRESHOLD_MAX, + DA7218_INVERT, da7218_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold", DA7218_ALC_TARGET_MIN, + DA7218_ALC_THRESHOLD_MIN_SHIFT, DA7218_ALC_THRESHOLD_MAX, + DA7218_INVERT, da7218_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold", DA7218_ALC_TARGET_MAX, + DA7218_ALC_THRESHOLD_MAX_SHIFT, DA7218_ALC_THRESHOLD_MAX, + DA7218_INVERT, da7218_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation", DA7218_ALC_GAIN_LIMITS, + DA7218_ALC_ATTEN_MAX_SHIFT, DA7218_ALC_ATTEN_GAIN_MAX, + DA7218_NO_INVERT, da7218_alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Gain", DA7218_ALC_GAIN_LIMITS, + DA7218_ALC_GAIN_MAX_SHIFT, DA7218_ALC_ATTEN_GAIN_MAX, + DA7218_NO_INVERT, da7218_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Min Analog Gain", DA7218_ALC_ANA_GAIN_LIMITS, + DA7218_ALC_ANA_GAIN_MIN_SHIFT, + DA7218_ALC_ANA_GAIN_MIN, DA7218_ALC_ANA_GAIN_MAX, + DA7218_NO_INVERT, da7218_alc_ana_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Max Analog Gain", DA7218_ALC_ANA_GAIN_LIMITS, + DA7218_ALC_ANA_GAIN_MAX_SHIFT, + DA7218_ALC_ANA_GAIN_MIN, DA7218_ALC_ANA_GAIN_MAX, + DA7218_NO_INVERT, da7218_alc_ana_gain_tlv), + SOC_ENUM("ALC Anticlip Step", da7218_alc_anticlip_step), + SOC_SINGLE("ALC Anticlip Switch", DA7218_ALC_ANTICLIP_CTRL, + DA7218_ALC_ANTICLIP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_DOUBLE_EXT("ALC Channel1 Switch", DA7218_ALC_CTRL1, + DA7218_ALC_CHAN1_L_EN_SHIFT, DA7218_ALC_CHAN1_R_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT, + snd_soc_get_volsw, da7218_alc_sw_put), + SOC_DOUBLE_EXT("ALC Channel2 Switch", DA7218_ALC_CTRL1, + DA7218_ALC_CHAN2_L_EN_SHIFT, DA7218_ALC_CHAN2_R_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT, + snd_soc_get_volsw, da7218_alc_sw_put), + + /* Envelope Tracking */ + SOC_ENUM("Envelope Tracking Attack Rate", da7218_integ_attack_rate), + SOC_ENUM("Envelope Tracking Release Rate", da7218_integ_release_rate), + + /* Input High-Pass Filters */ + SOC_ENUM("In Filter1 HPF Mode", da7218_in1_hpf_mode), + SOC_ENUM("In Filter1 HPF Corner Audio", da7218_in1_audio_hpf_corner), + SOC_ENUM("In Filter1 HPF Corner Voice", da7218_in1_voice_hpf_corner), + SOC_ENUM("In Filter2 HPF Mode", da7218_in2_hpf_mode), + SOC_ENUM("In Filter2 HPF Corner Audio", da7218_in2_audio_hpf_corner), + SOC_ENUM("In Filter2 HPF Corner Voice", da7218_in2_voice_hpf_corner), + + /* Mic Level Detect */ + SOC_DOUBLE_EXT("Mic Level Detect Channel1 Switch", DA7218_LVL_DET_CTRL, + DA7218_LVL_DET_EN_CHAN1L_SHIFT, + DA7218_LVL_DET_EN_CHAN1R_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT, da7218_mic_lvl_det_sw_get, + da7218_mic_lvl_det_sw_put), + SOC_DOUBLE_EXT("Mic Level Detect Channel2 Switch", DA7218_LVL_DET_CTRL, + DA7218_LVL_DET_EN_CHAN2L_SHIFT, + DA7218_LVL_DET_EN_CHAN2R_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT, da7218_mic_lvl_det_sw_get, + da7218_mic_lvl_det_sw_put), + SOC_SINGLE("Mic Level Detect Level", DA7218_LVL_DET_LEVEL, + DA7218_LVL_DET_LEVEL_SHIFT, DA7218_LVL_DET_LEVEL_MAX, + DA7218_NO_INVERT), + + /* Digital Mixer (Input) */ + SOC_SINGLE_TLV("DMix In Filter1L Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_INFILT_1L_GAIN, + DA7218_OUTDAI_1L_INFILT_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1L Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_INFILT_1L_GAIN, + DA7218_OUTDAI_1R_INFILT_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1L Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_INFILT_1L_GAIN, + DA7218_OUTDAI_2L_INFILT_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1L Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_INFILT_1L_GAIN, + DA7218_OUTDAI_2R_INFILT_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In Filter1R Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_INFILT_1R_GAIN, + DA7218_OUTDAI_1L_INFILT_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1R Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_INFILT_1R_GAIN, + DA7218_OUTDAI_1R_INFILT_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1R Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_INFILT_1R_GAIN, + DA7218_OUTDAI_2L_INFILT_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1R Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_INFILT_1R_GAIN, + DA7218_OUTDAI_2R_INFILT_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In Filter2L Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_INFILT_2L_GAIN, + DA7218_OUTDAI_1L_INFILT_2L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2L Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_INFILT_2L_GAIN, + DA7218_OUTDAI_1R_INFILT_2L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2L Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_INFILT_2L_GAIN, + DA7218_OUTDAI_2L_INFILT_2L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2L Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_INFILT_2L_GAIN, + DA7218_OUTDAI_2R_INFILT_2L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In Filter2R Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_INFILT_2R_GAIN, + DA7218_OUTDAI_1L_INFILT_2R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2R Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_INFILT_2R_GAIN, + DA7218_OUTDAI_1R_INFILT_2R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2R Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_INFILT_2R_GAIN, + DA7218_OUTDAI_2L_INFILT_2R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2R Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_INFILT_2R_GAIN, + DA7218_OUTDAI_2R_INFILT_2R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix ToneGen Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_TONEGEN_GAIN, + DA7218_OUTDAI_1L_TONEGEN_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix ToneGen Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_TONEGEN_GAIN, + DA7218_OUTDAI_1R_TONEGEN_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix ToneGen Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_TONEGEN_GAIN, + DA7218_OUTDAI_2L_TONEGEN_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix ToneGen Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_TONEGEN_GAIN, + DA7218_OUTDAI_2R_TONEGEN_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In DAIL Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_INDAI_1L_GAIN, + DA7218_OUTDAI_1L_INDAI_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIL Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_INDAI_1L_GAIN, + DA7218_OUTDAI_1R_INDAI_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIL Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_INDAI_1L_GAIN, + DA7218_OUTDAI_2L_INDAI_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIL Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_INDAI_1L_GAIN, + DA7218_OUTDAI_2R_INDAI_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In DAIR Out1 DAIL Volume", + DA7218_DMIX_OUTDAI_1L_INDAI_1R_GAIN, + DA7218_OUTDAI_1L_INDAI_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIR Out1 DAIR Volume", + DA7218_DMIX_OUTDAI_1R_INDAI_1R_GAIN, + DA7218_OUTDAI_1R_INDAI_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIR Out2 DAIL Volume", + DA7218_DMIX_OUTDAI_2L_INDAI_1R_GAIN, + DA7218_OUTDAI_2L_INDAI_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIR Out2 DAIR Volume", + DA7218_DMIX_OUTDAI_2R_INDAI_1R_GAIN, + DA7218_OUTDAI_2R_INDAI_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + /* Digital Mixer (Output) */ + SOC_SINGLE_TLV("DMix In Filter1L Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_INFILT_1L_GAIN, + DA7218_OUTFILT_1L_INFILT_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1L Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_INFILT_1L_GAIN, + DA7218_OUTFILT_1R_INFILT_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In Filter1R Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_INFILT_1R_GAIN, + DA7218_OUTFILT_1L_INFILT_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter1R Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_INFILT_1R_GAIN, + DA7218_OUTFILT_1R_INFILT_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In Filter2L Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_INFILT_2L_GAIN, + DA7218_OUTFILT_1L_INFILT_2L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2L Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_INFILT_2L_GAIN, + DA7218_OUTFILT_1R_INFILT_2L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In Filter2R Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_INFILT_2R_GAIN, + DA7218_OUTFILT_1L_INFILT_2R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In Filter2R Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_INFILT_2R_GAIN, + DA7218_OUTFILT_1R_INFILT_2R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix ToneGen Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_TONEGEN_GAIN, + DA7218_OUTFILT_1L_TONEGEN_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix ToneGen Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_TONEGEN_GAIN, + DA7218_OUTFILT_1R_TONEGEN_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In DAIL Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_INDAI_1L_GAIN, + DA7218_OUTFILT_1L_INDAI_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIL Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_INDAI_1L_GAIN, + DA7218_OUTFILT_1R_INDAI_1L_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + SOC_SINGLE_TLV("DMix In DAIR Out FilterL Volume", + DA7218_DMIX_OUTFILT_1L_INDAI_1R_GAIN, + DA7218_OUTFILT_1L_INDAI_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + SOC_SINGLE_TLV("DMix In DAIR Out FilterR Volume", + DA7218_DMIX_OUTFILT_1R_INDAI_1R_GAIN, + DA7218_OUTFILT_1R_INDAI_1R_GAIN_SHIFT, + DA7218_DMIX_GAIN_MAX, DA7218_NO_INVERT, + da7218_dmix_gain_tlv), + + /* Sidetone Filter */ + SND_SOC_BYTES_EXT("Sidetone BiQuad Coefficients", + DA7218_SIDETONE_BIQ_3STAGE_CFG_SIZE, + da7218_biquad_coeff_get, da7218_biquad_coeff_put), + SOC_SINGLE_TLV("Sidetone Volume", DA7218_SIDETONE_GAIN, + DA7218_SIDETONE_GAIN_SHIFT, DA7218_DMIX_GAIN_MAX, + DA7218_NO_INVERT, da7218_dmix_gain_tlv), + SOC_SINGLE("Sidetone Switch", DA7218_SIDETONE_CTRL, + DA7218_SIDETONE_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + + /* Tone Generator */ + SOC_ENUM("ToneGen DTMF Key", da7218_tonegen_dtmf_key), + SOC_SINGLE("ToneGen DTMF Switch", DA7218_TONE_GEN_CFG1, + DA7218_DTMF_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_ENUM("ToneGen Sinewave Gen Type", da7218_tonegen_swg_sel), + SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7218_TONE_GEN_FREQ1_L, + DA7218_FREQ1_L_SHIFT, DA7218_FREQ_MAX, DA7218_NO_INVERT, + da7218_tonegen_freq_get, da7218_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7218_TONE_GEN_FREQ2_L, + DA7218_FREQ2_L_SHIFT, DA7218_FREQ_MAX, DA7218_NO_INVERT, + da7218_tonegen_freq_get, da7218_tonegen_freq_put), + SOC_SINGLE("ToneGen On Time", DA7218_TONE_GEN_ON_PER, + DA7218_BEEP_ON_PER_SHIFT, DA7218_BEEP_ON_OFF_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("ToneGen Off Time", DA7218_TONE_GEN_OFF_PER, + DA7218_BEEP_OFF_PER_SHIFT, DA7218_BEEP_ON_OFF_MAX, + DA7218_NO_INVERT), + + /* Gain ramping */ + SOC_ENUM("Gain Ramp Rate", da7218_gain_ramp_rate), + + /* DGS */ + SOC_SINGLE_TLV("DGS Trigger", DA7218_DGS_TRIGGER, + DA7218_DGS_TRIGGER_LVL_SHIFT, DA7218_DGS_TRIGGER_MAX, + DA7218_INVERT, da7218_dgs_trigger_tlv), + SOC_ENUM("DGS Rise Coefficient", da7218_dgs_rise_coeff), + SOC_ENUM("DGS Fall Coefficient", da7218_dgs_fall_coeff), + SOC_SINGLE("DGS Sync Delay", DA7218_DGS_SYNC_DELAY, + DA7218_DGS_SYNC_DELAY_SHIFT, DA7218_DGS_SYNC_DELAY_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("DGS Fast SR Sync Delay", DA7218_DGS_SYNC_DELAY2, + DA7218_DGS_SYNC_DELAY2_SHIFT, DA7218_DGS_SYNC_DELAY_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("DGS Voice Filter Sync Delay", DA7218_DGS_SYNC_DELAY3, + DA7218_DGS_SYNC_DELAY3_SHIFT, DA7218_DGS_SYNC_DELAY3_MAX, + DA7218_NO_INVERT), + SOC_SINGLE_TLV("DGS Anticlip Level", DA7218_DGS_LEVELS, + DA7218_DGS_ANTICLIP_LVL_SHIFT, + DA7218_DGS_ANTICLIP_LVL_MAX, DA7218_INVERT, + da7218_dgs_anticlip_tlv), + SOC_SINGLE_TLV("DGS Signal Level", DA7218_DGS_LEVELS, + DA7218_DGS_SIGNAL_LVL_SHIFT, DA7218_DGS_SIGNAL_LVL_MAX, + DA7218_INVERT, da7218_dgs_signal_tlv), + SOC_SINGLE("DGS Gain Subrange Switch", DA7218_DGS_GAIN_CTRL, + DA7218_DGS_SUBR_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("DGS Gain Ramp Switch", DA7218_DGS_GAIN_CTRL, + DA7218_DGS_RAMP_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_SINGLE("DGS Gain Steps", DA7218_DGS_GAIN_CTRL, + DA7218_DGS_STEPS_SHIFT, DA7218_DGS_STEPS_MAX, + DA7218_NO_INVERT), + SOC_DOUBLE("DGS Switch", DA7218_DGS_ENABLE, DA7218_DGS_ENABLE_L_SHIFT, + DA7218_DGS_ENABLE_R_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + + /* Output High-Pass Filter */ + SOC_ENUM("Out Filter HPF Mode", da7218_out1_hpf_mode), + SOC_ENUM("Out Filter HPF Corner Audio", da7218_out1_audio_hpf_corner), + SOC_ENUM("Out Filter HPF Corner Voice", da7218_out1_voice_hpf_corner), + + /* 5-Band Equaliser */ + SOC_SINGLE_TLV("Out EQ Band1 Volume", DA7218_OUT_1_EQ_12_FILTER_CTRL, + DA7218_OUT_1_EQ_BAND1_SHIFT, DA7218_OUT_EQ_BAND_MAX, + DA7218_NO_INVERT, da7218_out_eq_band_tlv), + SOC_SINGLE_TLV("Out EQ Band2 Volume", DA7218_OUT_1_EQ_12_FILTER_CTRL, + DA7218_OUT_1_EQ_BAND2_SHIFT, DA7218_OUT_EQ_BAND_MAX, + DA7218_NO_INVERT, da7218_out_eq_band_tlv), + SOC_SINGLE_TLV("Out EQ Band3 Volume", DA7218_OUT_1_EQ_34_FILTER_CTRL, + DA7218_OUT_1_EQ_BAND3_SHIFT, DA7218_OUT_EQ_BAND_MAX, + DA7218_NO_INVERT, da7218_out_eq_band_tlv), + SOC_SINGLE_TLV("Out EQ Band4 Volume", DA7218_OUT_1_EQ_34_FILTER_CTRL, + DA7218_OUT_1_EQ_BAND4_SHIFT, DA7218_OUT_EQ_BAND_MAX, + DA7218_NO_INVERT, da7218_out_eq_band_tlv), + SOC_SINGLE_TLV("Out EQ Band5 Volume", DA7218_OUT_1_EQ_5_FILTER_CTRL, + DA7218_OUT_1_EQ_BAND5_SHIFT, DA7218_OUT_EQ_BAND_MAX, + DA7218_NO_INVERT, da7218_out_eq_band_tlv), + SOC_SINGLE("Out EQ Switch", DA7218_OUT_1_EQ_5_FILTER_CTRL, + DA7218_OUT_1_EQ_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + + /* BiQuad Filters */ + SND_SOC_BYTES_EXT("BiQuad Coefficients", + DA7218_OUT_1_BIQ_5STAGE_CFG_SIZE, + da7218_biquad_coeff_get, da7218_biquad_coeff_put), + SOC_SINGLE("BiQuad Filter Switch", DA7218_OUT_1_BIQ_5STAGE_CTRL, + DA7218_OUT_1_BIQ_5STAGE_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + + /* Output Filters */ + SOC_DOUBLE_R_RANGE_TLV("Out Filter Volume", DA7218_OUT_1L_GAIN, + DA7218_OUT_1R_GAIN, + DA7218_OUT_1L_DIGITAL_GAIN_SHIFT, + DA7218_OUT_DIGITAL_GAIN_MIN, + DA7218_OUT_DIGITAL_GAIN_MAX, DA7218_NO_INVERT, + da7218_out_dig_gain_tlv), + SOC_DOUBLE_R("Out Filter Switch", DA7218_OUT_1L_FILTER_CTRL, + DA7218_OUT_1R_FILTER_CTRL, DA7218_OUT_1L_MUTE_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_INVERT), + SOC_DOUBLE_R("Out Filter Gain Subrange Switch", + DA7218_OUT_1L_FILTER_CTRL, DA7218_OUT_1R_FILTER_CTRL, + DA7218_OUT_1L_SUBRANGE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_NO_INVERT), + SOC_DOUBLE_R("Out Filter Gain Ramp Switch", DA7218_OUT_1L_FILTER_CTRL, + DA7218_OUT_1R_FILTER_CTRL, DA7218_OUT_1L_RAMP_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), + + /* Mixer Output */ + SOC_DOUBLE_R_RANGE_TLV("Mixout Volume", DA7218_MIXOUT_L_GAIN, + DA7218_MIXOUT_R_GAIN, + DA7218_MIXOUT_L_AMP_GAIN_SHIFT, + DA7218_MIXOUT_AMP_GAIN_MIN, + DA7218_MIXOUT_AMP_GAIN_MAX, DA7218_NO_INVERT, + da7218_mixout_gain_tlv), + + /* DAC Noise Gate */ + SOC_ENUM("DAC NG Setup Time", da7218_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7218_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7218_dac_ng_rampdown_rate), + SOC_SINGLE_TLV("DAC NG Off Threshold", DA7218_DAC_NG_OFF_THRESH, + DA7218_DAC_NG_OFF_THRESHOLD_SHIFT, + DA7218_DAC_NG_THRESHOLD_MAX, DA7218_NO_INVERT, + da7218_dac_ng_threshold_tlv), + SOC_SINGLE_TLV("DAC NG On Threshold", DA7218_DAC_NG_ON_THRESH, + DA7218_DAC_NG_ON_THRESHOLD_SHIFT, + DA7218_DAC_NG_THRESHOLD_MAX, DA7218_NO_INVERT, + da7218_dac_ng_threshold_tlv), + SOC_SINGLE("DAC NG Switch", DA7218_DAC_NG_CTRL, DA7218_DAC_NG_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), + + /* CP */ + SOC_ENUM("Charge Pump Track Mode", da7218_cp_mchange), + SOC_ENUM("Charge Pump Frequency", da7218_cp_fcontrol), + SOC_ENUM("Charge Pump Decay Rate", da7218_cp_tau_delay), + SOC_SINGLE("Charge Pump Threshold", DA7218_CP_VOL_THRESHOLD1, + DA7218_CP_THRESH_VDD2_SHIFT, DA7218_CP_THRESH_VDD2_MAX, + DA7218_NO_INVERT), + + /* Headphones */ + SOC_DOUBLE_R_RANGE_TLV("Headphone Volume", DA7218_HP_L_GAIN, + DA7218_HP_R_GAIN, DA7218_HP_L_AMP_GAIN_SHIFT, + DA7218_HP_AMP_GAIN_MIN, DA7218_HP_AMP_GAIN_MAX, + DA7218_NO_INVERT, da7218_hp_gain_tlv), + SOC_DOUBLE_R("Headphone Switch", DA7218_HP_L_CTRL, DA7218_HP_R_CTRL, + DA7218_HP_L_AMP_MUTE_EN_SHIFT, DA7218_SWITCH_EN_MAX, + DA7218_INVERT), + SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7218_HP_L_CTRL, + DA7218_HP_R_CTRL, DA7218_HP_L_AMP_RAMP_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7218_HP_L_CTRL, + DA7218_HP_R_CTRL, DA7218_HP_L_AMP_ZC_EN_SHIFT, + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), +}; + + +/* + * DAPM Mux Controls + */ + +static const char * const da7218_mic_sel_text[] = { "Analog", "Digital" }; + +static const struct soc_enum da7218_mic1_sel = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(da7218_mic_sel_text), + da7218_mic_sel_text); + +static const struct snd_kcontrol_new da7218_mic1_sel_mux = + SOC_DAPM_ENUM("Mic1 Mux", da7218_mic1_sel); + +static const struct soc_enum da7218_mic2_sel = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(da7218_mic_sel_text), + da7218_mic_sel_text); + +static const struct snd_kcontrol_new da7218_mic2_sel_mux = + SOC_DAPM_ENUM("Mic2 Mux", da7218_mic2_sel); + +static const char * const da7218_sidetone_in_sel_txt[] = { + "In Filter1L", "In Filter1R", "In Filter2L", "In Filter2R" +}; + +static const struct soc_enum da7218_sidetone_in_sel = + SOC_ENUM_SINGLE(DA7218_SIDETONE_IN_SELECT, + DA7218_SIDETONE_IN_SELECT_SHIFT, + DA7218_SIDETONE_IN_SELECT_MAX, + da7218_sidetone_in_sel_txt); + +static const struct snd_kcontrol_new da7218_sidetone_in_sel_mux = + SOC_DAPM_ENUM("Sidetone Mux", da7218_sidetone_in_sel); + +static const char * const da7218_out_filt_biq_sel_txt[] = { + "Bypass", "Enabled" +}; + +static const struct soc_enum da7218_out_filtl_biq_sel = + SOC_ENUM_SINGLE(DA7218_OUT_1L_FILTER_CTRL, + DA7218_OUT_1L_BIQ_5STAGE_SEL_SHIFT, + DA7218_OUT_BIQ_5STAGE_SEL_MAX, + da7218_out_filt_biq_sel_txt); + +static const struct snd_kcontrol_new da7218_out_filtl_biq_sel_mux = + SOC_DAPM_ENUM("Out FilterL BiQuad Mux", da7218_out_filtl_biq_sel); + +static const struct soc_enum da7218_out_filtr_biq_sel = + SOC_ENUM_SINGLE(DA7218_OUT_1R_FILTER_CTRL, + DA7218_OUT_1R_BIQ_5STAGE_SEL_SHIFT, + DA7218_OUT_BIQ_5STAGE_SEL_MAX, + da7218_out_filt_biq_sel_txt); + +static const struct snd_kcontrol_new da7218_out_filtr_biq_sel_mux = + SOC_DAPM_ENUM("Out FilterR BiQuad Mux", da7218_out_filtr_biq_sel); + + +/* + * DAPM Mixer Controls + */ + +#define DA7218_DMIX_CTRLS(reg) \ + SOC_DAPM_SINGLE("In Filter1L Switch", reg, \ + DA7218_DMIX_SRC_INFILT1L, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("In Filter1R Switch", reg, \ + DA7218_DMIX_SRC_INFILT1R, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("In Filter2L Switch", reg, \ + DA7218_DMIX_SRC_INFILT2L, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("In Filter2R Switch", reg, \ + DA7218_DMIX_SRC_INFILT2R, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("ToneGen Switch", reg, \ + DA7218_DMIX_SRC_TONEGEN, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("DAIL Switch", reg, DA7218_DMIX_SRC_DAIL, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("DAIR Switch", reg, DA7218_DMIX_SRC_DAIR, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT) + +static const struct snd_kcontrol_new da7218_out_dai1l_mix_controls[] = { + DA7218_DMIX_CTRLS(DA7218_DROUTING_OUTDAI_1L), +}; + +static const struct snd_kcontrol_new da7218_out_dai1r_mix_controls[] = { + DA7218_DMIX_CTRLS(DA7218_DROUTING_OUTDAI_1R), +}; + +static const struct snd_kcontrol_new da7218_out_dai2l_mix_controls[] = { + DA7218_DMIX_CTRLS(DA7218_DROUTING_OUTDAI_2L), +}; + +static const struct snd_kcontrol_new da7218_out_dai2r_mix_controls[] = { + DA7218_DMIX_CTRLS(DA7218_DROUTING_OUTDAI_2R), +}; + +static const struct snd_kcontrol_new da7218_out_filtl_mix_controls[] = { + DA7218_DMIX_CTRLS(DA7218_DROUTING_OUTFILT_1L), +}; + +static const struct snd_kcontrol_new da7218_out_filtr_mix_controls[] = { + DA7218_DMIX_CTRLS(DA7218_DROUTING_OUTFILT_1R), +}; + +#define DA7218_DMIX_ST_CTRLS(reg) \ + SOC_DAPM_SINGLE("Out FilterL Switch", reg, \ + DA7218_DMIX_ST_SRC_OUTFILT1L, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("Out FilterR Switch", reg, \ + DA7218_DMIX_ST_SRC_OUTFILT1R, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT), \ + SOC_DAPM_SINGLE("Sidetone Switch", reg, \ + DA7218_DMIX_ST_SRC_SIDETONE, \ + DA7218_SWITCH_EN_MAX, DA7218_NO_INVERT) \ + +static const struct snd_kcontrol_new da7218_st_out_filtl_mix_controls[] = { + DA7218_DMIX_ST_CTRLS(DA7218_DROUTING_ST_OUTFILT_1L), +}; + +static const struct snd_kcontrol_new da7218_st_out_filtr_mix_controls[] = { + DA7218_DMIX_ST_CTRLS(DA7218_DROUTING_ST_OUTFILT_1R), +}; + + +/* + * DAPM Events + */ + +/* + * We keep track of which input filters are enabled. This is used in the logic + * for controlling the mic level detect feature. + */ +static int da7218_in_filter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + u8 mask; + + switch (w->reg) { + case DA7218_IN_1L_FILTER_CTRL: + mask = (1 << DA7218_LVL_DET_EN_CHAN1L_SHIFT); + break; + case DA7218_IN_1R_FILTER_CTRL: + mask = (1 << DA7218_LVL_DET_EN_CHAN1R_SHIFT); + break; + case DA7218_IN_2L_FILTER_CTRL: + mask = (1 << DA7218_LVL_DET_EN_CHAN2L_SHIFT); + break; + case DA7218_IN_2R_FILTER_CTRL: + mask = (1 << DA7218_LVL_DET_EN_CHAN2R_SHIFT); + break; + default: + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + da7218->in_filt_en |= mask; + /* + * If we're enabling path for mic level detect, wait for path + * to settle before enabling feature to avoid incorrect and + * unwanted detect events. + */ + if (mask & da7218->mic_lvl_det_en) + msleep(DA7218_MIC_LVL_DET_DELAY); + break; + case SND_SOC_DAPM_PRE_PMD: + da7218->in_filt_en &= ~mask; + break; + default: + return -EINVAL; + } + + /* Enable configured level detection paths */ + snd_soc_write(codec, DA7218_LVL_DET_CTRL, + (da7218->in_filt_en & da7218->mic_lvl_det_en)); + + return 0; +} + +static int da7218_dai_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status, refosc_cal; + int i; + bool success; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (da7218->master) + /* Enable DAI clks for master mode */ + snd_soc_update_bits(codec, DA7218_DAI_CLK_MODE, + DA7218_DAI_CLK_EN_MASK, + DA7218_DAI_CLK_EN_MASK); + + /* Tune reference oscillator */ + snd_soc_write(codec, DA7218_PLL_REFOSC_CAL, + DA7218_PLL_REFOSC_CAL_START_MASK); + snd_soc_write(codec, DA7218_PLL_REFOSC_CAL, + DA7218_PLL_REFOSC_CAL_START_MASK | + DA7218_PLL_REFOSC_CAL_EN_MASK); + + /* Check tuning complete */ + i = 0; + success = false; + do { + refosc_cal = snd_soc_read(codec, DA7218_PLL_REFOSC_CAL); + if (!(refosc_cal & DA7218_PLL_REFOSC_CAL_START_MASK)) { + success = true; + } else { + ++i; + usleep_range(DA7218_REF_OSC_CHECK_DELAY_MIN, + DA7218_REF_OSC_CHECK_DELAY_MAX); + } + } while ((i < DA7218_REF_OSC_CHECK_TRIES) && (!success)); + + if (!success) + dev_warn(codec->dev, + "Reference oscillator failed calibration\n"); + + /* PC synchronised to DAI */ + snd_soc_write(codec, DA7218_PC_COUNT, + DA7218_PC_RESYNC_AUTO_MASK); + + /* If SRM not enabled, we don't need to check status */ + pll_ctrl = snd_soc_read(codec, DA7218_PLL_CTRL); + if ((pll_ctrl & DA7218_PLL_MODE_MASK) != DA7218_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + i = 0; + success = false; + do { + pll_status = snd_soc_read(codec, DA7218_PLL_STATUS); + if (pll_status & DA7218_PLL_SRM_STATUS_SRM_LOCK) { + success = true; + } else { + ++i; + msleep(DA7218_SRM_CHECK_DELAY); + } + } while ((i < DA7218_SRM_CHECK_TRIES) & (!success)); + + if (!success) + dev_warn(codec->dev, "SRM failed to lock\n"); + + return 0; + case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_write(codec, DA7218_PC_COUNT, DA7218_PC_FREERUN_MASK); + + if (da7218->master) + /* Disable DAI clks for master mode */ + snd_soc_update_bits(codec, DA7218_DAI_CLK_MODE, + DA7218_DAI_CLK_EN_MASK, 0); + + return 0; + default: + return -EINVAL; + } +} + +static int da7218_cp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + + /* + * If this is DA7217 and we're using single supply for differential + * output, we really don't want to touch the charge pump. + */ + if (da7218->hp_single_supply) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, DA7218_CP_CTRL, DA7218_CP_EN_MASK, + DA7218_CP_EN_MASK); + return 0; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, DA7218_CP_CTRL, DA7218_CP_EN_MASK, + 0); + return 0; + default: + return -EINVAL; + } +} + +static int da7218_hp_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Enable headphone output */ + snd_soc_update_bits(codec, w->reg, DA7218_HP_AMP_OE_MASK, + DA7218_HP_AMP_OE_MASK); + return 0; + case SND_SOC_DAPM_PRE_PMD: + /* Headphone output high impedance */ + snd_soc_update_bits(codec, w->reg, DA7218_HP_AMP_OE_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* + * DAPM Widgets + */ + +static const struct snd_soc_dapm_widget da7218_dapm_widgets[] = { + /* Input Supplies */ + SND_SOC_DAPM_SUPPLY("Mic Bias1", DA7218_MICBIAS_EN, + DA7218_MICBIAS_1_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias2", DA7218_MICBIAS_EN, + DA7218_MICBIAS_2_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMic1 Left", DA7218_DMIC_1_CTRL, + DA7218_DMIC_1L_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMic1 Right", DA7218_DMIC_1_CTRL, + DA7218_DMIC_1R_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMic2 Left", DA7218_DMIC_2_CTRL, + DA7218_DMIC_2L_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMic2 Right", DA7218_DMIC_2_CTRL, + DA7218_DMIC_2R_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMIC1L"), + SND_SOC_DAPM_INPUT("DMIC1R"), + SND_SOC_DAPM_INPUT("DMIC2L"), + SND_SOC_DAPM_INPUT("DMIC2R"), + + /* Input Mixer Supplies */ + SND_SOC_DAPM_SUPPLY("Mixin1 Supply", DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_MIX_SEL_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("Mixin2 Supply", DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_MIX_SEL_SHIFT, DA7218_NO_INVERT, + NULL, 0), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic1 PGA", DA7218_MIC_1_CTRL, + DA7218_MIC_1_AMP_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mic2 PGA", DA7218_MIC_2_CTRL, + DA7218_MIC_2_AMP_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixin1 PGA", DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_AMP_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixin2 PGA", DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_AMP_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + + /* Mic/DMic Muxes */ + SND_SOC_DAPM_MUX("Mic1 Mux", SND_SOC_NOPM, 0, 0, &da7218_mic1_sel_mux), + SND_SOC_DAPM_MUX("Mic2 Mux", SND_SOC_NOPM, 0, 0, &da7218_mic2_sel_mux), + + /* Input Filters */ + SND_SOC_DAPM_ADC_E("In Filter1L", NULL, DA7218_IN_1L_FILTER_CTRL, + DA7218_IN_1L_FILTER_EN_SHIFT, DA7218_NO_INVERT, + da7218_in_filter_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("In Filter1R", NULL, DA7218_IN_1R_FILTER_CTRL, + DA7218_IN_1R_FILTER_EN_SHIFT, DA7218_NO_INVERT, + da7218_in_filter_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("In Filter2L", NULL, DA7218_IN_2L_FILTER_CTRL, + DA7218_IN_2L_FILTER_EN_SHIFT, DA7218_NO_INVERT, + da7218_in_filter_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("In Filter2R", NULL, DA7218_IN_2R_FILTER_CTRL, + DA7218_IN_2R_FILTER_EN_SHIFT, DA7218_NO_INVERT, + da7218_in_filter_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Tone Generator */ + SND_SOC_DAPM_SIGGEN("TONE"), + SND_SOC_DAPM_PGA("Tone Generator", DA7218_TONE_GEN_CFG1, + DA7218_START_STOPN_SHIFT, DA7218_NO_INVERT, NULL, 0), + + /* Sidetone Input */ + SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, + &da7218_sidetone_in_sel_mux), + SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7218_SIDETONE_CTRL, + DA7218_SIDETONE_FILTER_EN_SHIFT, DA7218_NO_INVERT), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("Mixer DAI1L", SND_SOC_NOPM, 0, 0, + da7218_out_dai1l_mix_controls, + ARRAY_SIZE(da7218_out_dai1l_mix_controls)), + SND_SOC_DAPM_MIXER("Mixer DAI1R", SND_SOC_NOPM, 0, 0, + da7218_out_dai1r_mix_controls, + ARRAY_SIZE(da7218_out_dai1r_mix_controls)), + SND_SOC_DAPM_MIXER("Mixer DAI2L", SND_SOC_NOPM, 0, 0, + da7218_out_dai2l_mix_controls, + ARRAY_SIZE(da7218_out_dai2l_mix_controls)), + SND_SOC_DAPM_MIXER("Mixer DAI2R", SND_SOC_NOPM, 0, 0, + da7218_out_dai2r_mix_controls, + ARRAY_SIZE(da7218_out_dai2r_mix_controls)), + + /* DAI Supply */ + SND_SOC_DAPM_SUPPLY("DAI", DA7218_DAI_CTRL, DA7218_DAI_EN_SHIFT, + DA7218_NO_INVERT, da7218_dai_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* DAI */ + SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7218_out_filtl_mix_controls, + ARRAY_SIZE(da7218_out_filtl_mix_controls)), + SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0, + da7218_out_filtr_mix_controls, + ARRAY_SIZE(da7218_out_filtr_mix_controls)), + + /* BiQuad Filters */ + SND_SOC_DAPM_MUX("Out FilterL BiQuad Mux", SND_SOC_NOPM, 0, 0, + &da7218_out_filtl_biq_sel_mux), + SND_SOC_DAPM_MUX("Out FilterR BiQuad Mux", SND_SOC_NOPM, 0, 0, + &da7218_out_filtr_biq_sel_mux), + SND_SOC_DAPM_DAC("BiQuad Filter", NULL, DA7218_OUT_1_BIQ_5STAGE_CTRL, + DA7218_OUT_1_BIQ_5STAGE_FILTER_EN_SHIFT, + DA7218_NO_INVERT), + + /* Sidetone Mixers */ + SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7218_st_out_filtl_mix_controls, + ARRAY_SIZE(da7218_st_out_filtl_mix_controls)), + SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0, 0, + da7218_st_out_filtr_mix_controls, + ARRAY_SIZE(da7218_st_out_filtr_mix_controls)), + + /* Output Filters */ + SND_SOC_DAPM_DAC("Out FilterL", NULL, DA7218_OUT_1L_FILTER_CTRL, + DA7218_OUT_1L_FILTER_EN_SHIFT, DA7218_NO_INVERT), + SND_SOC_DAPM_DAC("Out FilterR", NULL, DA7218_OUT_1R_FILTER_CTRL, + DA7218_IN_1R_FILTER_EN_SHIFT, DA7218_NO_INVERT), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7218_MIXOUT_L_CTRL, + DA7218_MIXOUT_L_AMP_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7218_MIXOUT_R_CTRL, + DA7218_MIXOUT_R_AMP_EN_SHIFT, DA7218_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA_E("Headphone Left PGA", DA7218_HP_L_CTRL, + DA7218_HP_L_AMP_EN_SHIFT, DA7218_NO_INVERT, NULL, 0, + da7218_hp_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Headphone Right PGA", DA7218_HP_R_CTRL, + DA7218_HP_R_AMP_EN_SHIFT, DA7218_NO_INVERT, NULL, 0, + da7218_hp_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Supplies */ + SND_SOC_DAPM_SUPPLY("Charge Pump", SND_SOC_NOPM, 0, 0, da7218_cp_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + + +/* + * DAPM Mixer Routes + */ + +#define DA7218_DMIX_ROUTES(name) \ + {name, "In Filter1L Switch", "In Filter1L"}, \ + {name, "In Filter1R Switch", "In Filter1R"}, \ + {name, "In Filter2L Switch", "In Filter2L"}, \ + {name, "In Filter2R Switch", "In Filter2R"}, \ + {name, "ToneGen Switch", "Tone Generator"}, \ + {name, "DAIL Switch", "DAIIN"}, \ + {name, "DAIR Switch", "DAIIN"} + +#define DA7218_DMIX_ST_ROUTES(name) \ + {name, "Out FilterL Switch", "Out FilterL BiQuad Mux"}, \ + {name, "Out FilterR Switch", "Out FilterR BiQuad Mux"}, \ + {name, "Sidetone Switch", "Sidetone Filter"} + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7218_audio_map[] = { + /* Input paths */ + {"MIC1", NULL, "Mic Bias1"}, + {"MIC2", NULL, "Mic Bias2"}, + {"DMIC1L", NULL, "Mic Bias1"}, + {"DMIC1L", NULL, "DMic1 Left"}, + {"DMIC1R", NULL, "Mic Bias1"}, + {"DMIC1R", NULL, "DMic1 Right"}, + {"DMIC2L", NULL, "Mic Bias2"}, + {"DMIC2L", NULL, "DMic2 Left"}, + {"DMIC2R", NULL, "Mic Bias2"}, + {"DMIC2R", NULL, "DMic2 Right"}, + + {"Mic1 PGA", NULL, "MIC1"}, + {"Mic2 PGA", NULL, "MIC2"}, + + {"Mixin1 PGA", NULL, "Mixin1 Supply"}, + {"Mixin2 PGA", NULL, "Mixin2 Supply"}, + + {"Mixin1 PGA", NULL, "Mic1 PGA"}, + {"Mixin2 PGA", NULL, "Mic2 PGA"}, + + {"Mic1 Mux", "Analog", "Mixin1 PGA"}, + {"Mic1 Mux", "Digital", "DMIC1L"}, + {"Mic1 Mux", "Digital", "DMIC1R"}, + {"Mic2 Mux", "Analog", "Mixin2 PGA"}, + {"Mic2 Mux", "Digital", "DMIC2L"}, + {"Mic2 Mux", "Digital", "DMIC2R"}, + + {"In Filter1L", NULL, "Mic1 Mux"}, + {"In Filter1R", NULL, "Mic1 Mux"}, + {"In Filter2L", NULL, "Mic2 Mux"}, + {"In Filter2R", NULL, "Mic2 Mux"}, + + {"Tone Generator", NULL, "TONE"}, + + {"Sidetone Mux", "In Filter1L", "In Filter1L"}, + {"Sidetone Mux", "In Filter1R", "In Filter1R"}, + {"Sidetone Mux", "In Filter2L", "In Filter2L"}, + {"Sidetone Mux", "In Filter2R", "In Filter2R"}, + {"Sidetone Filter", NULL, "Sidetone Mux"}, + + DA7218_DMIX_ROUTES("Mixer DAI1L"), + DA7218_DMIX_ROUTES("Mixer DAI1R"), + DA7218_DMIX_ROUTES("Mixer DAI2L"), + DA7218_DMIX_ROUTES("Mixer DAI2R"), + + {"DAIOUT", NULL, "Mixer DAI1L"}, + {"DAIOUT", NULL, "Mixer DAI1R"}, + {"DAIOUT", NULL, "Mixer DAI2L"}, + {"DAIOUT", NULL, "Mixer DAI2R"}, + + {"DAIOUT", NULL, "DAI"}, + + /* Output paths */ + {"DAIIN", NULL, "DAI"}, + + DA7218_DMIX_ROUTES("Mixer Out FilterL"), + DA7218_DMIX_ROUTES("Mixer Out FilterR"), + + {"BiQuad Filter", NULL, "Mixer Out FilterL"}, + {"BiQuad Filter", NULL, "Mixer Out FilterR"}, + + {"Out FilterL BiQuad Mux", "Bypass", "Mixer Out FilterL"}, + {"Out FilterL BiQuad Mux", "Enabled", "BiQuad Filter"}, + {"Out FilterR BiQuad Mux", "Bypass", "Mixer Out FilterR"}, + {"Out FilterR BiQuad Mux", "Enabled", "BiQuad Filter"}, + + DA7218_DMIX_ST_ROUTES("ST Mixer Out FilterL"), + DA7218_DMIX_ST_ROUTES("ST Mixer Out FilterR"), + + {"Out FilterL", NULL, "ST Mixer Out FilterL"}, + {"Out FilterR", NULL, "ST Mixer Out FilterR"}, + + {"Mixout Left PGA", NULL, "Out FilterL"}, + {"Mixout Right PGA", NULL, "Out FilterR"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + + {"HPL", NULL, "Headphone Left PGA"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"HPL", NULL, "Charge Pump"}, + {"HPR", NULL, "Charge Pump"}, +}; + + +/* + * DAI operations + */ + +static int da7218_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (da7218->mclk_rate == freq) + return 0; + + if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + + switch (clk_id) { + case DA7218_CLKSRC_MCLK_SQR: + snd_soc_update_bits(codec, DA7218_PLL_CTRL, + DA7218_PLL_MCLK_SQR_EN_MASK, + DA7218_PLL_MCLK_SQR_EN_MASK); + break; + case DA7218_CLKSRC_MCLK: + snd_soc_update_bits(codec, DA7218_PLL_CTRL, + DA7218_PLL_MCLK_SQR_EN_MASK, 0); + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } + + if (da7218->mclk) { + freq = clk_round_rate(da7218->mclk, freq); + ret = clk_set_rate(da7218->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7218->mclk_rate = freq; + + return 0; +} + +static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ + if (da7218->mclk_rate == 32768) { + indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; + indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + } else if (da7218->mclk_rate < 2000000) { + dev_err(codec->dev, "PLL input clock %d below valid range\n", + da7218->mclk_rate); + return -EINVAL; + } else if (da7218->mclk_rate <= 5000000) { + indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; + indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + } else if (da7218->mclk_rate <= 10000000) { + indiv_bits = DA7218_PLL_INDIV_5_10_MHZ; + indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + } else if (da7218->mclk_rate <= 20000000) { + indiv_bits = DA7218_PLL_INDIV_10_20_MHZ; + indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL; + } else if (da7218->mclk_rate <= 40000000) { + indiv_bits = DA7218_PLL_INDIV_20_40_MHZ; + indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7218->mclk_rate <= 54000000) { + indiv_bits = DA7218_PLL_INDIV_40_54_MHZ; + indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL; + } else { + dev_err(codec->dev, "PLL input clock %d above valid range\n", + da7218->mclk_rate); + return -EINVAL; + } + freq_ref = (da7218->mclk_rate / indiv); + pll_ctrl = indiv_bits; + + /* Configure PLL */ + switch (source) { + case DA7218_SYSCLK_MCLK: + pll_ctrl |= DA7218_PLL_MODE_BYPASS; + snd_soc_update_bits(codec, DA7218_PLL_CTRL, + DA7218_PLL_INDIV_MASK | + DA7218_PLL_MODE_MASK, pll_ctrl); + return 0; + case DA7218_SYSCLK_PLL: + pll_ctrl |= DA7218_PLL_MODE_NORMAL; + break; + case DA7218_SYSCLK_PLL_SRM: + pll_ctrl |= DA7218_PLL_MODE_SRM; + break; + case DA7218_SYSCLK_PLL_32KHZ: + pll_ctrl |= DA7218_PLL_MODE_32KHZ; + break; + default: + dev_err(codec->dev, "Invalid PLL config\n"); + return -EINVAL; + } + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7218_BYTE_SHIFT) & DA7218_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7218_BYTE_MASK; + + /* Write PLL config & dividers */ + snd_soc_write(codec, DA7218_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7218_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7218_PLL_INTEGER, pll_integer); + snd_soc_update_bits(codec, DA7218_PLL_CTRL, + DA7218_PLL_MODE_MASK | DA7218_PLL_INDIV_MASK, + pll_ctrl); + + return 0; +} + +static int da7218_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da7218->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + da7218->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | DA7218_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7218_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7218_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7218_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + dai_ctrl |= DA7218_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + /* By default 64 BCLKs per WCLK is supported */ + dai_clk_mode |= DA7218_DAI_BCLKS_PER_WCLK_64; + + snd_soc_write(codec, DA7218_DAI_CLK_MODE, dai_clk_mode); + snd_soc_update_bits(codec, DA7218_DAI_CTRL, DA7218_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7218_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dai_bclks_per_wclk; + u32 frame_size; + + /* No channels enabled so disable TDM, revert to 64-bit frames */ + if (!tx_mask) { + snd_soc_update_bits(codec, DA7218_DAI_TDM_CTRL, + DA7218_DAI_TDM_CH_EN_MASK | + DA7218_DAI_TDM_MODE_EN_MASK, 0); + snd_soc_update_bits(codec, DA7218_DAI_CLK_MODE, + DA7218_DAI_BCLKS_PER_WCLK_MASK, + DA7218_DAI_BCLKS_PER_WCLK_64); + return 0; + } + + /* Check we have valid slots */ + if (fls(tx_mask) > DA7218_DAI_TDM_MAX_SLOTS) { + dev_err(codec->dev, "Invalid number of slots, max = %d\n", + DA7218_DAI_TDM_MAX_SLOTS); + return -EINVAL; + } + + /* Check we have a valid offset given (first 2 bytes of rx_mask) */ + if (rx_mask >> DA7218_2BYTE_SHIFT) { + dev_err(codec->dev, "Invalid slot offset, max = %d\n", + DA7218_2BYTE_MASK); + return -EINVAL; + } + + /* Calculate & validate frame size based on slot info provided. */ + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7218_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7218_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7218_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7218_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(codec->dev, "Invalid frame size\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7218_DAI_CLK_MODE, + DA7218_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + snd_soc_write(codec, DA7218_DAI_OFFSET_LOWER, + (rx_mask & DA7218_BYTE_MASK)); + snd_soc_write(codec, DA7218_DAI_OFFSET_UPPER, + ((rx_mask >> DA7218_BYTE_SHIFT) & DA7218_BYTE_MASK)); + snd_soc_update_bits(codec, DA7218_DAI_TDM_CTRL, + DA7218_DAI_TDM_CH_EN_MASK | + DA7218_DAI_TDM_MODE_EN_MASK, + (tx_mask << DA7218_DAI_TDM_CH_EN_SHIFT) | + DA7218_DAI_TDM_MODE_EN_MASK); + + return 0; +} + +static int da7218_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dai_ctrl = 0, fs; + unsigned int channels; + + switch (params_width(params)) { + case 16: + dai_ctrl |= DA7218_DAI_WORD_LENGTH_S16_LE; + break; + case 20: + dai_ctrl |= DA7218_DAI_WORD_LENGTH_S20_LE; + break; + case 24: + dai_ctrl |= DA7218_DAI_WORD_LENGTH_S24_LE; + break; + case 32: + dai_ctrl |= DA7218_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + channels = params_channels(params); + if ((channels < 1) || (channels > DA7218_DAI_CH_NUM_MAX)) { + dev_err(codec->dev, + "Invalid number of channels, only 1 to %d supported\n", + DA7218_DAI_CH_NUM_MAX); + return -EINVAL; + } + dai_ctrl |= channels << DA7218_DAI_CH_NUM_SHIFT; + + switch (params_rate(params)) { + case 8000: + fs = DA7218_SR_8000; + break; + case 11025: + fs = DA7218_SR_11025; + break; + case 12000: + fs = DA7218_SR_12000; + break; + case 16000: + fs = DA7218_SR_16000; + break; + case 22050: + fs = DA7218_SR_22050; + break; + case 24000: + fs = DA7218_SR_24000; + break; + case 32000: + fs = DA7218_SR_32000; + break; + case 44100: + fs = DA7218_SR_44100; + break; + case 48000: + fs = DA7218_SR_48000; + break; + case 88200: + fs = DA7218_SR_88200; + break; + case 96000: + fs = DA7218_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7218_DAI_CTRL, + DA7218_DAI_WORD_LENGTH_MASK | DA7218_DAI_CH_NUM_MASK, + dai_ctrl); + /* SRs tied for ADCs and DACs. */ + snd_soc_write(codec, DA7218_SR, + (fs << DA7218_SR_DAC_SHIFT) | (fs << DA7218_SR_ADC_SHIFT)); + + return 0; +} + +static const struct snd_soc_dai_ops da7218_dai_ops = { + .hw_params = da7218_hw_params, + .set_sysclk = da7218_set_dai_sysclk, + .set_pll = da7218_set_dai_pll, + .set_fmt = da7218_set_dai_fmt, + .set_tdm_slot = da7218_set_dai_tdm_slot, +}; + +#define DA7218_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver da7218_dai = { + .name = "da7218-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, /* Only 2 channels of data */ + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7218_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7218_FORMATS, + }, + .ops = &da7218_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, +}; + + +/* + * HP Detect + */ + +int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + + if (da7218->dev_id == DA7217_DEV_ID) + return -EINVAL; + + da7218->jack = jack; + snd_soc_update_bits(codec, DA7218_HPLDET_JACK, + DA7218_HPLDET_JACK_EN_MASK, + jack ? DA7218_HPLDET_JACK_EN_MASK : 0); + + return 0; +} +EXPORT_SYMBOL_GPL(da7218_hpldet); + +static void da7218_hpldet_irq(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + u8 jack_status; + int report; + + jack_status = snd_soc_read(codec, DA7218_EVENT_STATUS); + + if (jack_status & DA7218_HPLDET_JACK_STS_MASK) + report = SND_JACK_HEADPHONE; + else + report = 0; + + snd_soc_jack_report(da7218->jack, report, SND_JACK_HEADPHONE); +} + +/* + * IRQ + */ + +static irqreturn_t da7218_irq_thread(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + u8 status; + + /* Read IRQ status reg */ + status = snd_soc_read(codec, DA7218_EVENT); + if (!status) + return IRQ_NONE; + + /* HP detect */ + if (status & DA7218_HPLDET_JACK_EVENT_MASK) + da7218_hpldet_irq(codec); + + /* Clear interrupts */ + snd_soc_write(codec, DA7218_EVENT, status); + + return IRQ_HANDLED; +} + +/* + * DT + */ + +static const struct of_device_id da7218_of_match[] = { + { .compatible = "dlg,da7217", .data = (void *) DA7217_DEV_ID }, + { .compatible = "dlg,da7218", .data = (void *) DA7218_DEV_ID }, + { } +}; +MODULE_DEVICE_TABLE(of, da7218_of_match); + +static inline int da7218_of_get_id(struct device *dev) +{ + const struct of_device_id *id = of_match_device(da7218_of_match, dev); + + if (id) + return (int) id->data; + else + return -EINVAL; +} + +static enum da7218_micbias_voltage + da7218_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1200: + return DA7218_MICBIAS_1_2V; + case 1600: + return DA7218_MICBIAS_1_6V; + case 1800: + return DA7218_MICBIAS_1_8V; + case 2000: + return DA7218_MICBIAS_2_0V; + case 2200: + return DA7218_MICBIAS_2_2V; + case 2400: + return DA7218_MICBIAS_2_4V; + case 2600: + return DA7218_MICBIAS_2_6V; + case 2800: + return DA7218_MICBIAS_2_8V; + case 3000: + return DA7218_MICBIAS_3_0V; + default: + dev_warn(codec->dev, "Invalid micbias level"); + return DA7218_MICBIAS_1_6V; + } +} + +static enum da7218_mic_amp_in_sel + da7218_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "diff")) { + return DA7218_MIC_AMP_IN_SEL_DIFF; + } else if (!strcmp(str, "se_p")) { + return DA7218_MIC_AMP_IN_SEL_SE_P; + } else if (!strcmp(str, "se_n")) { + return DA7218_MIC_AMP_IN_SEL_SE_N; + } else { + dev_warn(codec->dev, "Invalid mic input type selection"); + return DA7218_MIC_AMP_IN_SEL_DIFF; + } +} + +static enum da7218_dmic_data_sel + da7218_of_dmic_data_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "lrise_rfall")) { + return DA7218_DMIC_DATA_LRISE_RFALL; + } else if (!strcmp(str, "lfall_rrise")) { + return DA7218_DMIC_DATA_LFALL_RRISE; + } else { + dev_warn(codec->dev, "Invalid DMIC data type selection"); + return DA7218_DMIC_DATA_LRISE_RFALL; + } +} + +static enum da7218_dmic_samplephase + da7218_of_dmic_samplephase(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "on_clkedge")) { + return DA7218_DMIC_SAMPLE_ON_CLKEDGE; + } else if (!strcmp(str, "between_clkedge")) { + return DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE; + } else { + dev_warn(codec->dev, "Invalid DMIC sample phase"); + return DA7218_DMIC_SAMPLE_ON_CLKEDGE; + } +} + +static enum da7218_dmic_clk_rate + da7218_of_dmic_clkrate(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1500000: + return DA7218_DMIC_CLK_1_5MHZ; + case 3000000: + return DA7218_DMIC_CLK_3_0MHZ; + default: + dev_warn(codec->dev, "Invalid DMIC clock rate"); + return DA7218_DMIC_CLK_3_0MHZ; + } +} + +static enum da7218_hpldet_jack_rate + da7218_of_jack_rate(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 5: + return DA7218_HPLDET_JACK_RATE_5US; + case 10: + return DA7218_HPLDET_JACK_RATE_10US; + case 20: + return DA7218_HPLDET_JACK_RATE_20US; + case 40: + return DA7218_HPLDET_JACK_RATE_40US; + case 80: + return DA7218_HPLDET_JACK_RATE_80US; + case 160: + return DA7218_HPLDET_JACK_RATE_160US; + case 320: + return DA7218_HPLDET_JACK_RATE_320US; + case 640: + return DA7218_HPLDET_JACK_RATE_640US; + default: + dev_warn(codec->dev, "Invalid jack detect rate"); + return DA7218_HPLDET_JACK_RATE_40US; + } +} + +static enum da7218_hpldet_jack_debounce + da7218_of_jack_debounce(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 0: + return DA7218_HPLDET_JACK_DEBOUNCE_OFF; + case 2: + return DA7218_HPLDET_JACK_DEBOUNCE_2; + case 3: + return DA7218_HPLDET_JACK_DEBOUNCE_3; + case 4: + return DA7218_HPLDET_JACK_DEBOUNCE_4; + default: + dev_warn(codec->dev, "Invalid jack debounce"); + return DA7218_HPLDET_JACK_DEBOUNCE_2; + } +} + +static enum da7218_hpldet_jack_thr + da7218_of_jack_thr(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 84: + return DA7218_HPLDET_JACK_THR_84PCT; + case 88: + return DA7218_HPLDET_JACK_THR_88PCT; + case 92: + return DA7218_HPLDET_JACK_THR_92PCT; + case 96: + return DA7218_HPLDET_JACK_THR_96PCT; + default: + dev_warn(codec->dev, "Invalid jack threshold level"); + return DA7218_HPLDET_JACK_THR_84PCT; + } +} + +static struct da7218_pdata *da7218_of_to_pdata(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct device_node *np = codec->dev->of_node; + struct device_node *hpldet_np; + struct da7218_pdata *pdata; + struct da7218_hpldet_pdata *hpldet_pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) { + dev_warn(codec->dev, "Failed to allocate memory for pdata\n"); + return NULL; + } + + if (of_property_read_u32(np, "dlg,micbias1-lvl-millivolt", &of_val32) >= 0) + pdata->micbias1_lvl = da7218_of_micbias_lvl(codec, of_val32); + else + pdata->micbias1_lvl = DA7218_MICBIAS_1_6V; + + if (of_property_read_u32(np, "dlg,micbias2-lvl-millivolt", &of_val32) >= 0) + pdata->micbias2_lvl = da7218_of_micbias_lvl(codec, of_val32); + else + pdata->micbias2_lvl = DA7218_MICBIAS_1_6V; + + if (!of_property_read_string(np, "dlg,mic1-amp-in-sel", &of_str)) + pdata->mic1_amp_in_sel = + da7218_of_mic_amp_in_sel(codec, of_str); + else + pdata->mic1_amp_in_sel = DA7218_MIC_AMP_IN_SEL_DIFF; + + if (!of_property_read_string(np, "dlg,mic2-amp-in-sel", &of_str)) + pdata->mic2_amp_in_sel = + da7218_of_mic_amp_in_sel(codec, of_str); + else + pdata->mic2_amp_in_sel = DA7218_MIC_AMP_IN_SEL_DIFF; + + if (!of_property_read_string(np, "dlg,dmic1-data-sel", &of_str)) + pdata->dmic1_data_sel = da7218_of_dmic_data_sel(codec, of_str); + else + pdata->dmic1_data_sel = DA7218_DMIC_DATA_LRISE_RFALL; + + if (!of_property_read_string(np, "dlg,dmic1-samplephase", &of_str)) + pdata->dmic1_samplephase = + da7218_of_dmic_samplephase(codec, of_str); + else + pdata->dmic1_samplephase = DA7218_DMIC_SAMPLE_ON_CLKEDGE; + + if (of_property_read_u32(np, "dlg,dmic1-clkrate-hz", &of_val32) >= 0) + pdata->dmic1_clk_rate = da7218_of_dmic_clkrate(codec, of_val32); + else + pdata->dmic1_clk_rate = DA7218_DMIC_CLK_3_0MHZ; + + if (!of_property_read_string(np, "dlg,dmic2-data-sel", &of_str)) + pdata->dmic2_data_sel = da7218_of_dmic_data_sel(codec, of_str); + else + pdata->dmic2_data_sel = DA7218_DMIC_DATA_LRISE_RFALL; + + if (!of_property_read_string(np, "dlg,dmic2-samplephase", &of_str)) + pdata->dmic2_samplephase = + da7218_of_dmic_samplephase(codec, of_str); + else + pdata->dmic2_samplephase = DA7218_DMIC_SAMPLE_ON_CLKEDGE; + + if (of_property_read_u32(np, "dlg,dmic2-clkrate-hz", &of_val32) >= 0) + pdata->dmic2_clk_rate = da7218_of_dmic_clkrate(codec, of_val32); + else + pdata->dmic2_clk_rate = DA7218_DMIC_CLK_3_0MHZ; + + if (da7218->dev_id == DA7217_DEV_ID) { + if (of_property_read_bool(np, "dlg,hp-diff-single-supply")) + pdata->hp_diff_single_supply = true; + } + + if (da7218->dev_id == DA7218_DEV_ID) { + hpldet_np = of_find_node_by_name(np, "da7218_hpldet"); + if (!hpldet_np) + return pdata; + + hpldet_pdata = devm_kzalloc(codec->dev, sizeof(*hpldet_pdata), + GFP_KERNEL); + if (!hpldet_pdata) { + dev_warn(codec->dev, + "Failed to allocate memory for hpldet pdata\n"); + of_node_put(hpldet_np); + return pdata; + } + pdata->hpldet_pdata = hpldet_pdata; + + if (of_property_read_u32(hpldet_np, "dlg,jack-rate-us", + &of_val32) >= 0) + hpldet_pdata->jack_rate = + da7218_of_jack_rate(codec, of_val32); + else + hpldet_pdata->jack_rate = DA7218_HPLDET_JACK_RATE_40US; + + if (of_property_read_u32(hpldet_np, "dlg,jack-debounce", + &of_val32) >= 0) + hpldet_pdata->jack_debounce = + da7218_of_jack_debounce(codec, of_val32); + else + hpldet_pdata->jack_debounce = + DA7218_HPLDET_JACK_DEBOUNCE_2; + + if (of_property_read_u32(hpldet_np, "dlg,jack-threshold-pct", + &of_val32) >= 0) + hpldet_pdata->jack_thr = + da7218_of_jack_thr(codec, of_val32); + else + hpldet_pdata->jack_thr = DA7218_HPLDET_JACK_THR_84PCT; + + if (of_property_read_bool(hpldet_np, "dlg,comp-inv")) + hpldet_pdata->comp_inv = true; + + if (of_property_read_bool(hpldet_np, "dlg,hyst")) + hpldet_pdata->hyst = true; + + if (of_property_read_bool(hpldet_np, "dlg,discharge")) + hpldet_pdata->discharge = true; + + of_node_put(hpldet_np); + } + + return pdata; +} + + +/* + * Codec driver functions + */ + +static int da7218_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + if (da7218->mclk) { + ret = clk_prepare_enable(da7218->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable mclk\n"); + return ret; + } + } + + /* Master bias */ + snd_soc_update_bits(codec, DA7218_REFERENCES, + DA7218_BIAS_EN_MASK, + DA7218_BIAS_EN_MASK); + + /* Internal LDO */ + snd_soc_update_bits(codec, DA7218_LDO_CTRL, + DA7218_LDO_EN_MASK, + DA7218_LDO_EN_MASK); + } + break; + case SND_SOC_BIAS_OFF: + /* Only disable if jack detection disabled */ + if (!da7218->jack) { + /* Internal LDO */ + snd_soc_update_bits(codec, DA7218_LDO_CTRL, + DA7218_LDO_EN_MASK, 0); + + /* Master bias */ + snd_soc_update_bits(codec, DA7218_REFERENCES, + DA7218_BIAS_EN_MASK, 0); + } + + /* MCLK */ + if (da7218->mclk) + clk_disable_unprepare(da7218->mclk); + break; + } + + return 0; +} + +static const char *da7218_supply_names[DA7218_NUM_SUPPLIES] = { + [DA7218_SUPPLY_VDD] = "VDD", + [DA7218_SUPPLY_VDDMIC] = "VDDMIC", + [DA7218_SUPPLY_VDDIO] = "VDDIO", +}; + +static int da7218_handle_supplies(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddio; + u8 io_voltage_lvl = DA7218_IO_VOLTAGE_LEVEL_2_5V_3_6V; + int i, ret; + + /* Get required supplies */ + for (i = 0; i < DA7218_NUM_SUPPLIES; ++i) + da7218->supplies[i].supply = da7218_supply_names[i]; + + ret = devm_regulator_bulk_get(codec->dev, DA7218_NUM_SUPPLIES, + da7218->supplies); + if (ret) { + dev_err(codec->dev, "Failed to get supplies\n"); + return ret; + } + + /* Determine VDDIO voltage provided */ + vddio = da7218->supplies[DA7218_SUPPLY_VDDIO].consumer; + ret = regulator_get_voltage(vddio); + if (ret < 1500000) + dev_warn(codec->dev, "Invalid VDDIO voltage\n"); + else if (ret < 2500000) + io_voltage_lvl = DA7218_IO_VOLTAGE_LEVEL_1_5V_2_5V; + + /* Enable main supplies */ + ret = regulator_bulk_enable(DA7218_NUM_SUPPLIES, da7218->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies\n"); + return ret; + } + + /* Ensure device in active mode */ + snd_soc_write(codec, DA7218_SYSTEM_ACTIVE, DA7218_SYSTEM_ACTIVE_MASK); + + /* Update IO voltage level range */ + snd_soc_write(codec, DA7218_IO_CTRL, io_voltage_lvl); + + return 0; +} + +static void da7218_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + struct da7218_pdata *pdata = da7218->pdata; + + if (pdata) { + u8 micbias_lvl = 0, dmic_cfg = 0; + + /* Mic Bias voltages */ + switch (pdata->micbias1_lvl) { + case DA7218_MICBIAS_1_2V: + micbias_lvl |= DA7218_MICBIAS_1_LP_MODE_MASK; + break; + case DA7218_MICBIAS_1_6V: + case DA7218_MICBIAS_1_8V: + case DA7218_MICBIAS_2_0V: + case DA7218_MICBIAS_2_2V: + case DA7218_MICBIAS_2_4V: + case DA7218_MICBIAS_2_6V: + case DA7218_MICBIAS_2_8V: + case DA7218_MICBIAS_3_0V: + micbias_lvl |= (pdata->micbias1_lvl << + DA7218_MICBIAS_1_LEVEL_SHIFT); + break; + } + + switch (pdata->micbias2_lvl) { + case DA7218_MICBIAS_1_2V: + micbias_lvl |= DA7218_MICBIAS_2_LP_MODE_MASK; + break; + case DA7218_MICBIAS_1_6V: + case DA7218_MICBIAS_1_8V: + case DA7218_MICBIAS_2_0V: + case DA7218_MICBIAS_2_2V: + case DA7218_MICBIAS_2_4V: + case DA7218_MICBIAS_2_6V: + case DA7218_MICBIAS_2_8V: + case DA7218_MICBIAS_3_0V: + micbias_lvl |= (pdata->micbias2_lvl << + DA7218_MICBIAS_2_LEVEL_SHIFT); + break; + } + + snd_soc_write(codec, DA7218_MICBIAS_CTRL, micbias_lvl); + + /* Mic */ + switch (pdata->mic1_amp_in_sel) { + case DA7218_MIC_AMP_IN_SEL_DIFF: + case DA7218_MIC_AMP_IN_SEL_SE_P: + case DA7218_MIC_AMP_IN_SEL_SE_N: + snd_soc_write(codec, DA7218_MIC_1_SELECT, + pdata->mic1_amp_in_sel); + break; + } + + switch (pdata->mic2_amp_in_sel) { + case DA7218_MIC_AMP_IN_SEL_DIFF: + case DA7218_MIC_AMP_IN_SEL_SE_P: + case DA7218_MIC_AMP_IN_SEL_SE_N: + snd_soc_write(codec, DA7218_MIC_2_SELECT, + pdata->mic2_amp_in_sel); + break; + } + + /* DMic */ + switch (pdata->dmic1_data_sel) { + case DA7218_DMIC_DATA_LFALL_RRISE: + case DA7218_DMIC_DATA_LRISE_RFALL: + dmic_cfg |= (pdata->dmic1_data_sel << + DA7218_DMIC_1_DATA_SEL_SHIFT); + break; + } + + switch (pdata->dmic1_samplephase) { + case DA7218_DMIC_SAMPLE_ON_CLKEDGE: + case DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE: + dmic_cfg |= (pdata->dmic1_samplephase << + DA7218_DMIC_1_SAMPLEPHASE_SHIFT); + break; + } + + switch (pdata->dmic1_clk_rate) { + case DA7218_DMIC_CLK_3_0MHZ: + case DA7218_DMIC_CLK_1_5MHZ: + dmic_cfg |= (pdata->dmic1_clk_rate << + DA7218_DMIC_1_CLK_RATE_SHIFT); + break; + } + + snd_soc_update_bits(codec, DA7218_DMIC_1_CTRL, + DA7218_DMIC_1_DATA_SEL_MASK | + DA7218_DMIC_1_SAMPLEPHASE_MASK | + DA7218_DMIC_1_CLK_RATE_MASK, dmic_cfg); + + dmic_cfg = 0; + switch (pdata->dmic2_data_sel) { + case DA7218_DMIC_DATA_LFALL_RRISE: + case DA7218_DMIC_DATA_LRISE_RFALL: + dmic_cfg |= (pdata->dmic2_data_sel << + DA7218_DMIC_2_DATA_SEL_SHIFT); + break; + } + + switch (pdata->dmic2_samplephase) { + case DA7218_DMIC_SAMPLE_ON_CLKEDGE: + case DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE: + dmic_cfg |= (pdata->dmic2_samplephase << + DA7218_DMIC_2_SAMPLEPHASE_SHIFT); + break; + } + + switch (pdata->dmic2_clk_rate) { + case DA7218_DMIC_CLK_3_0MHZ: + case DA7218_DMIC_CLK_1_5MHZ: + dmic_cfg |= (pdata->dmic2_clk_rate << + DA7218_DMIC_2_CLK_RATE_SHIFT); + break; + } + + snd_soc_update_bits(codec, DA7218_DMIC_2_CTRL, + DA7218_DMIC_2_DATA_SEL_MASK | + DA7218_DMIC_2_SAMPLEPHASE_MASK | + DA7218_DMIC_2_CLK_RATE_MASK, dmic_cfg); + + /* DA7217 Specific */ + if (da7218->dev_id == DA7217_DEV_ID) { + da7218->hp_single_supply = + pdata->hp_diff_single_supply; + + if (da7218->hp_single_supply) { + snd_soc_write(codec, DA7218_HP_DIFF_UNLOCK, + DA7218_HP_DIFF_UNLOCK_VAL); + snd_soc_update_bits(codec, DA7218_HP_DIFF_CTRL, + DA7218_HP_AMP_SINGLE_SUPPLY_EN_MASK, + DA7218_HP_AMP_SINGLE_SUPPLY_EN_MASK); + } + } + + /* DA7218 Specific */ + if ((da7218->dev_id == DA7218_DEV_ID) && + (pdata->hpldet_pdata)) { + struct da7218_hpldet_pdata *hpldet_pdata = + pdata->hpldet_pdata; + u8 hpldet_cfg = 0; + + switch (hpldet_pdata->jack_rate) { + case DA7218_HPLDET_JACK_RATE_5US: + case DA7218_HPLDET_JACK_RATE_10US: + case DA7218_HPLDET_JACK_RATE_20US: + case DA7218_HPLDET_JACK_RATE_40US: + case DA7218_HPLDET_JACK_RATE_80US: + case DA7218_HPLDET_JACK_RATE_160US: + case DA7218_HPLDET_JACK_RATE_320US: + case DA7218_HPLDET_JACK_RATE_640US: + hpldet_cfg |= + (hpldet_pdata->jack_rate << + DA7218_HPLDET_JACK_RATE_SHIFT); + break; + } + + switch (hpldet_pdata->jack_debounce) { + case DA7218_HPLDET_JACK_DEBOUNCE_OFF: + case DA7218_HPLDET_JACK_DEBOUNCE_2: + case DA7218_HPLDET_JACK_DEBOUNCE_3: + case DA7218_HPLDET_JACK_DEBOUNCE_4: + hpldet_cfg |= + (hpldet_pdata->jack_debounce << + DA7218_HPLDET_JACK_DEBOUNCE_SHIFT); + break; + } + + switch (hpldet_pdata->jack_thr) { + case DA7218_HPLDET_JACK_THR_84PCT: + case DA7218_HPLDET_JACK_THR_88PCT: + case DA7218_HPLDET_JACK_THR_92PCT: + case DA7218_HPLDET_JACK_THR_96PCT: + hpldet_cfg |= + (hpldet_pdata->jack_thr << + DA7218_HPLDET_JACK_THR_SHIFT); + break; + } + snd_soc_update_bits(codec, DA7218_HPLDET_JACK, + DA7218_HPLDET_JACK_RATE_MASK | + DA7218_HPLDET_JACK_DEBOUNCE_MASK | + DA7218_HPLDET_JACK_THR_MASK, + hpldet_cfg); + + hpldet_cfg = 0; + if (hpldet_pdata->comp_inv) + hpldet_cfg |= DA7218_HPLDET_COMP_INV_MASK; + + if (hpldet_pdata->hyst) + hpldet_cfg |= DA7218_HPLDET_HYST_EN_MASK; + + if (hpldet_pdata->discharge) + hpldet_cfg |= DA7218_HPLDET_DISCHARGE_EN_MASK; + + snd_soc_write(codec, DA7218_HPLDET_CTRL, hpldet_cfg); + } + } +} + +static int da7218_probe(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + int ret; + + /* Regulator configuration */ + ret = da7218_handle_supplies(codec); + if (ret) + return ret; + + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7218->pdata = da7218_of_to_pdata(codec); + else + da7218->pdata = dev_get_platdata(codec->dev); + + da7218_handle_pdata(codec); + + /* Check if MCLK provided, if not the clock is NULL */ + da7218->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7218->mclk)) { + if (PTR_ERR(da7218->mclk) != -ENOENT) { + ret = PTR_ERR(da7218->mclk); + goto err_disable_reg; + } else { + da7218->mclk = NULL; + } + } + + /* Default PC to free-running */ + snd_soc_write(codec, DA7218_PC_COUNT, DA7218_PC_FREERUN_MASK); + + /* + * Default Output Filter mixers to off otherwise DAPM will power + * Mic to HP passthrough paths by default at startup. + */ + snd_soc_write(codec, DA7218_DROUTING_OUTFILT_1L, 0); + snd_soc_write(codec, DA7218_DROUTING_OUTFILT_1R, 0); + + /* Default CP to normal load, power mode */ + snd_soc_update_bits(codec, DA7218_CP_CTRL, + DA7218_CP_SMALL_SWITCH_FREQ_EN_MASK, 0); + + /* Default gain ramping */ + snd_soc_update_bits(codec, DA7218_MIXIN_1_CTRL, + DA7218_MIXIN_1_AMP_RAMP_EN_MASK, + DA7218_MIXIN_1_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_MIXIN_2_CTRL, + DA7218_MIXIN_2_AMP_RAMP_EN_MASK, + DA7218_MIXIN_2_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_1L_FILTER_CTRL, + DA7218_IN_1L_RAMP_EN_MASK, + DA7218_IN_1L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_1R_FILTER_CTRL, + DA7218_IN_1R_RAMP_EN_MASK, + DA7218_IN_1R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_2L_FILTER_CTRL, + DA7218_IN_2L_RAMP_EN_MASK, + DA7218_IN_2L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_IN_2R_FILTER_CTRL, + DA7218_IN_2R_RAMP_EN_MASK, + DA7218_IN_2R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_DGS_GAIN_CTRL, + DA7218_DGS_RAMP_EN_MASK, DA7218_DGS_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_OUT_1L_FILTER_CTRL, + DA7218_OUT_1L_RAMP_EN_MASK, + DA7218_OUT_1L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_OUT_1R_FILTER_CTRL, + DA7218_OUT_1R_RAMP_EN_MASK, + DA7218_OUT_1R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_HP_L_CTRL, + DA7218_HP_L_AMP_RAMP_EN_MASK, + DA7218_HP_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7218_HP_R_CTRL, + DA7218_HP_R_AMP_RAMP_EN_MASK, + DA7218_HP_R_AMP_RAMP_EN_MASK); + + /* Default infinite tone gen, start/stop by Kcontrol */ + snd_soc_write(codec, DA7218_TONE_GEN_CYCLES, DA7218_BEEP_CYCLES_MASK); + + /* DA7217 specific config */ + if (da7218->dev_id == DA7217_DEV_ID) { + snd_soc_update_bits(codec, DA7218_HP_DIFF_CTRL, + DA7218_HP_AMP_DIFF_MODE_EN_MASK, + DA7218_HP_AMP_DIFF_MODE_EN_MASK); + + /* Only DA7218 supports HP detect, mask off for DA7217 */ + snd_soc_write(codec, DA7218_EVENT_MASK, + DA7218_HPLDET_JACK_EVENT_IRQ_MSK_MASK); + } + + if (da7218->irq) { + /* Mask off mic level events, currently not handled */ + snd_soc_update_bits(codec, DA7218_EVENT_MASK, + DA7218_LVL_DET_EVENT_MSK_MASK, + DA7218_LVL_DET_EVENT_MSK_MASK); + + ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, + da7218_irq_thread, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "da7218", codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to request IRQ %d: %d\n", + da7218->irq, ret); + goto err_disable_reg; + } + + } + + return 0; + +err_disable_reg: + regulator_bulk_disable(DA7218_NUM_SUPPLIES, da7218->supplies); + + return ret; +} + +static int da7218_remove(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(DA7218_NUM_SUPPLIES, da7218->supplies); + + return 0; +} + +#ifdef CONFIG_PM +static int da7218_suspend(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + + da7218_set_bias_level(codec, SND_SOC_BIAS_OFF); + + /* Put device into standby mode if jack detection disabled */ + if (!da7218->jack) + snd_soc_write(codec, DA7218_SYSTEM_ACTIVE, 0); + + return 0; +} + +static int da7218_resume(struct snd_soc_codec *codec) +{ + struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); + + /* Put device into active mode if previously moved to standby */ + if (!da7218->jack) + snd_soc_write(codec, DA7218_SYSTEM_ACTIVE, + DA7218_SYSTEM_ACTIVE_MASK); + + da7218_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define da7218_suspend NULL +#define da7218_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_da7218 = { + .probe = da7218_probe, + .remove = da7218_remove, + .suspend = da7218_suspend, + .resume = da7218_resume, + .set_bias_level = da7218_set_bias_level, + + .controls = da7218_snd_controls, + .num_controls = ARRAY_SIZE(da7218_snd_controls), + + .dapm_widgets = da7218_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7218_dapm_widgets), + .dapm_routes = da7218_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7218_audio_map), +}; + + +/* + * Regmap configs + */ + +static struct reg_default da7218_reg_defaults[] = { + { DA7218_SYSTEM_ACTIVE, 0x00 }, + { DA7218_CIF_CTRL, 0x00 }, + { DA7218_SPARE1, 0x00 }, + { DA7218_SR, 0xAA }, + { DA7218_PC_COUNT, 0x02 }, + { DA7218_GAIN_RAMP_CTRL, 0x00 }, + { DA7218_CIF_TIMEOUT_CTRL, 0x01 }, + { DA7218_SYSTEM_MODES_INPUT, 0x00 }, + { DA7218_SYSTEM_MODES_OUTPUT, 0x00 }, + { DA7218_IN_1L_FILTER_CTRL, 0x00 }, + { DA7218_IN_1R_FILTER_CTRL, 0x00 }, + { DA7218_IN_2L_FILTER_CTRL, 0x00 }, + { DA7218_IN_2R_FILTER_CTRL, 0x00 }, + { DA7218_OUT_1L_FILTER_CTRL, 0x40 }, + { DA7218_OUT_1R_FILTER_CTRL, 0x40 }, + { DA7218_OUT_1_HPF_FILTER_CTRL, 0x80 }, + { DA7218_OUT_1_EQ_12_FILTER_CTRL, 0x77 }, + { DA7218_OUT_1_EQ_34_FILTER_CTRL, 0x77 }, + { DA7218_OUT_1_EQ_5_FILTER_CTRL, 0x07 }, + { DA7218_OUT_1_BIQ_5STAGE_CTRL, 0x40 }, + { DA7218_OUT_1_BIQ_5STAGE_DATA, 0x00 }, + { DA7218_OUT_1_BIQ_5STAGE_ADDR, 0x00 }, + { DA7218_MIXIN_1_CTRL, 0x48 }, + { DA7218_MIXIN_1_GAIN, 0x03 }, + { DA7218_MIXIN_2_CTRL, 0x48 }, + { DA7218_MIXIN_2_GAIN, 0x03 }, + { DA7218_ALC_CTRL1, 0x00 }, + { DA7218_ALC_CTRL2, 0x00 }, + { DA7218_ALC_CTRL3, 0x00 }, + { DA7218_ALC_NOISE, 0x3F }, + { DA7218_ALC_TARGET_MIN, 0x3F }, + { DA7218_ALC_TARGET_MAX, 0x00 }, + { DA7218_ALC_GAIN_LIMITS, 0xFF }, + { DA7218_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7218_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7218_AGS_ENABLE, 0x00 }, + { DA7218_AGS_TRIGGER, 0x09 }, + { DA7218_AGS_ATT_MAX, 0x00 }, + { DA7218_AGS_TIMEOUT, 0x00 }, + { DA7218_AGS_ANTICLIP_CTRL, 0x00 }, + { DA7218_ENV_TRACK_CTRL, 0x00 }, + { DA7218_LVL_DET_CTRL, 0x00 }, + { DA7218_LVL_DET_LEVEL, 0x7F }, + { DA7218_DGS_TRIGGER, 0x24 }, + { DA7218_DGS_ENABLE, 0x00 }, + { DA7218_DGS_RISE_FALL, 0x50 }, + { DA7218_DGS_SYNC_DELAY, 0xA3 }, + { DA7218_DGS_SYNC_DELAY2, 0x31 }, + { DA7218_DGS_SYNC_DELAY3, 0x11 }, + { DA7218_DGS_LEVELS, 0x01 }, + { DA7218_DGS_GAIN_CTRL, 0x74 }, + { DA7218_DROUTING_OUTDAI_1L, 0x01 }, + { DA7218_DMIX_OUTDAI_1L_INFILT_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1L_INFILT_1R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1L_INFILT_2L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1L_INFILT_2R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1L_TONEGEN_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1L_INDAI_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1L_INDAI_1R_GAIN, 0x1C }, + { DA7218_DROUTING_OUTDAI_1R, 0x04 }, + { DA7218_DMIX_OUTDAI_1R_INFILT_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1R_INFILT_1R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1R_INFILT_2L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1R_INFILT_2R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1R_TONEGEN_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1R_INDAI_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_1R_INDAI_1R_GAIN, 0x1C }, + { DA7218_DROUTING_OUTFILT_1L, 0x01 }, + { DA7218_DMIX_OUTFILT_1L_INFILT_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1L_INFILT_1R_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1L_INFILT_2L_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1L_INFILT_2R_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1L_TONEGEN_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1L_INDAI_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1L_INDAI_1R_GAIN, 0x1C }, + { DA7218_DROUTING_OUTFILT_1R, 0x04 }, + { DA7218_DMIX_OUTFILT_1R_INFILT_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1R_INFILT_1R_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1R_INFILT_2L_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1R_INFILT_2R_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1R_TONEGEN_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1R_INDAI_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTFILT_1R_INDAI_1R_GAIN, 0x1C }, + { DA7218_DROUTING_OUTDAI_2L, 0x04 }, + { DA7218_DMIX_OUTDAI_2L_INFILT_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2L_INFILT_1R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2L_INFILT_2L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2L_INFILT_2R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2L_TONEGEN_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2L_INDAI_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2L_INDAI_1R_GAIN, 0x1C }, + { DA7218_DROUTING_OUTDAI_2R, 0x08 }, + { DA7218_DMIX_OUTDAI_2R_INFILT_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2R_INFILT_1R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2R_INFILT_2L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2R_INFILT_2R_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2R_TONEGEN_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2R_INDAI_1L_GAIN, 0x1C }, + { DA7218_DMIX_OUTDAI_2R_INDAI_1R_GAIN, 0x1C }, + { DA7218_DAI_CTRL, 0x28 }, + { DA7218_DAI_TDM_CTRL, 0x40 }, + { DA7218_DAI_OFFSET_LOWER, 0x00 }, + { DA7218_DAI_OFFSET_UPPER, 0x00 }, + { DA7218_DAI_CLK_MODE, 0x01 }, + { DA7218_PLL_CTRL, 0x04 }, + { DA7218_PLL_FRAC_TOP, 0x00 }, + { DA7218_PLL_FRAC_BOT, 0x00 }, + { DA7218_PLL_INTEGER, 0x20 }, + { DA7218_DAC_NG_CTRL, 0x00 }, + { DA7218_DAC_NG_SETUP_TIME, 0x00 }, + { DA7218_DAC_NG_OFF_THRESH, 0x00 }, + { DA7218_DAC_NG_ON_THRESH, 0x00 }, + { DA7218_TONE_GEN_CFG2, 0x00 }, + { DA7218_TONE_GEN_FREQ1_L, 0x55 }, + { DA7218_TONE_GEN_FREQ1_U, 0x15 }, + { DA7218_TONE_GEN_FREQ2_L, 0x00 }, + { DA7218_TONE_GEN_FREQ2_U, 0x40 }, + { DA7218_TONE_GEN_CYCLES, 0x00 }, + { DA7218_TONE_GEN_ON_PER, 0x02 }, + { DA7218_TONE_GEN_OFF_PER, 0x01 }, + { DA7218_CP_CTRL, 0x60 }, + { DA7218_CP_DELAY, 0x11 }, + { DA7218_CP_VOL_THRESHOLD1, 0x0E }, + { DA7218_MIC_1_CTRL, 0x40 }, + { DA7218_MIC_1_GAIN, 0x01 }, + { DA7218_MIC_1_SELECT, 0x00 }, + { DA7218_MIC_2_CTRL, 0x40 }, + { DA7218_MIC_2_GAIN, 0x01 }, + { DA7218_MIC_2_SELECT, 0x00 }, + { DA7218_IN_1_HPF_FILTER_CTRL, 0x80 }, + { DA7218_IN_2_HPF_FILTER_CTRL, 0x80 }, + { DA7218_ADC_1_CTRL, 0x07 }, + { DA7218_ADC_2_CTRL, 0x07 }, + { DA7218_MIXOUT_L_CTRL, 0x00 }, + { DA7218_MIXOUT_L_GAIN, 0x03 }, + { DA7218_MIXOUT_R_CTRL, 0x00 }, + { DA7218_MIXOUT_R_GAIN, 0x03 }, + { DA7218_HP_L_CTRL, 0x40 }, + { DA7218_HP_L_GAIN, 0x3B }, + { DA7218_HP_R_CTRL, 0x40 }, + { DA7218_HP_R_GAIN, 0x3B }, + { DA7218_HP_DIFF_CTRL, 0x00 }, + { DA7218_HP_DIFF_UNLOCK, 0xC3 }, + { DA7218_HPLDET_JACK, 0x0B }, + { DA7218_HPLDET_CTRL, 0x00 }, + { DA7218_REFERENCES, 0x08 }, + { DA7218_IO_CTRL, 0x00 }, + { DA7218_LDO_CTRL, 0x00 }, + { DA7218_SIDETONE_CTRL, 0x40 }, + { DA7218_SIDETONE_IN_SELECT, 0x00 }, + { DA7218_SIDETONE_GAIN, 0x1C }, + { DA7218_DROUTING_ST_OUTFILT_1L, 0x01 }, + { DA7218_DROUTING_ST_OUTFILT_1R, 0x02 }, + { DA7218_SIDETONE_BIQ_3STAGE_DATA, 0x00 }, + { DA7218_SIDETONE_BIQ_3STAGE_ADDR, 0x00 }, + { DA7218_EVENT_MASK, 0x00 }, + { DA7218_DMIC_1_CTRL, 0x00 }, + { DA7218_DMIC_2_CTRL, 0x00 }, + { DA7218_IN_1L_GAIN, 0x6F }, + { DA7218_IN_1R_GAIN, 0x6F }, + { DA7218_IN_2L_GAIN, 0x6F }, + { DA7218_IN_2R_GAIN, 0x6F }, + { DA7218_OUT_1L_GAIN, 0x6F }, + { DA7218_OUT_1R_GAIN, 0x6F }, + { DA7218_MICBIAS_CTRL, 0x00 }, + { DA7218_MICBIAS_EN, 0x00 }, +}; + +static bool da7218_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7218_STATUS1: + case DA7218_SOFT_RESET: + case DA7218_SYSTEM_STATUS: + case DA7218_CALIB_CTRL: + case DA7218_CALIB_OFFSET_AUTO_M_1: + case DA7218_CALIB_OFFSET_AUTO_U_1: + case DA7218_CALIB_OFFSET_AUTO_M_2: + case DA7218_CALIB_OFFSET_AUTO_U_2: + case DA7218_PLL_STATUS: + case DA7218_PLL_REFOSC_CAL: + case DA7218_TONE_GEN_CFG1: + case DA7218_ADC_MODE: + case DA7218_HP_SNGL_CTRL: + case DA7218_HPLDET_TEST: + case DA7218_EVENT_STATUS: + case DA7218_EVENT: + return 1; + default: + return 0; + } +} + +static const struct regmap_config da7218_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA7218_MICBIAS_EN, + .reg_defaults = da7218_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7218_reg_defaults), + .volatile_reg = da7218_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + + +/* + * I2C layer + */ + +static int da7218_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7218_priv *da7218; + int ret; + + da7218 = devm_kzalloc(&i2c->dev, sizeof(struct da7218_priv), + GFP_KERNEL); + if (!da7218) + return -ENOMEM; + + i2c_set_clientdata(i2c, da7218); + + if (i2c->dev.of_node) + da7218->dev_id = da7218_of_get_id(&i2c->dev); + else + da7218->dev_id = id->driver_data; + + if ((da7218->dev_id != DA7217_DEV_ID) && + (da7218->dev_id != DA7218_DEV_ID)) { + dev_err(&i2c->dev, "Invalid device Id\n"); + return -EINVAL; + } + + da7218->irq = i2c->irq; + + da7218->regmap = devm_regmap_init_i2c(i2c, &da7218_regmap_config); + if (IS_ERR(da7218->regmap)) { + ret = PTR_ERR(da7218->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da7218, &da7218_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7218 codec: %d\n", + ret); + } + return ret; +} + +static int da7218_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7218_i2c_id[] = { + { "da7217", DA7217_DEV_ID }, + { "da7218", DA7218_DEV_ID }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7218_i2c_id); + +static struct i2c_driver da7218_i2c_driver = { + .driver = { + .name = "da7218", + .of_match_table = of_match_ptr(da7218_of_match), + }, + .probe = da7218_i2c_probe, + .remove = da7218_i2c_remove, + .id_table = da7218_i2c_id, +}; + +module_i2c_driver(da7218_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7218 Codec driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h new file mode 100644 index 000000000000..c2c59049a2ad --- /dev/null +++ b/sound/soc/codecs/da7218.h @@ -0,0 +1,1414 @@ +/* + * da7218.h - DA7218 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _DA7218_H +#define _DA7218_H + +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/da7218.h> + + +/* + * Registers + */ +#define DA7218_SYSTEM_ACTIVE 0x0 +#define DA7218_CIF_CTRL 0x1 +#define DA7218_CHIP_ID1 0x4 +#define DA7218_CHIP_ID2 0x5 +#define DA7218_CHIP_REVISION 0x6 +#define DA7218_SPARE1 0x7 +#define DA7218_STATUS1 0x8 +#define DA7218_SOFT_RESET 0x9 +#define DA7218_SR 0xB +#define DA7218_PC_COUNT 0xC +#define DA7218_GAIN_RAMP_CTRL 0xD +#define DA7218_CIF_TIMEOUT_CTRL 0x10 +#define DA7218_SYSTEM_MODES_INPUT 0x14 +#define DA7218_SYSTEM_MODES_OUTPUT 0x15 +#define DA7218_SYSTEM_STATUS 0x16 +#define DA7218_IN_1L_FILTER_CTRL 0x18 +#define DA7218_IN_1R_FILTER_CTRL 0x19 +#define DA7218_IN_2L_FILTER_CTRL 0x1A +#define DA7218_IN_2R_FILTER_CTRL 0x1B +#define DA7218_OUT_1L_FILTER_CTRL 0x20 +#define DA7218_OUT_1R_FILTER_CTRL 0x21 +#define DA7218_OUT_1_HPF_FILTER_CTRL 0x24 +#define DA7218_OUT_1_EQ_12_FILTER_CTRL 0x25 +#define DA7218_OUT_1_EQ_34_FILTER_CTRL 0x26 +#define DA7218_OUT_1_EQ_5_FILTER_CTRL 0x27 +#define DA7218_OUT_1_BIQ_5STAGE_CTRL 0x28 +#define DA7218_OUT_1_BIQ_5STAGE_DATA 0x29 +#define DA7218_OUT_1_BIQ_5STAGE_ADDR 0x2A +#define DA7218_MIXIN_1_CTRL 0x2C +#define DA7218_MIXIN_1_GAIN 0x2D +#define DA7218_MIXIN_2_CTRL 0x2E +#define DA7218_MIXIN_2_GAIN 0x2F +#define DA7218_ALC_CTRL1 0x30 +#define DA7218_ALC_CTRL2 0x31 +#define DA7218_ALC_CTRL3 0x32 +#define DA7218_ALC_NOISE 0x33 +#define DA7218_ALC_TARGET_MIN 0x34 +#define DA7218_ALC_TARGET_MAX 0x35 +#define DA7218_ALC_GAIN_LIMITS 0x36 +#define DA7218_ALC_ANA_GAIN_LIMITS 0x37 +#define DA7218_ALC_ANTICLIP_CTRL 0x38 +#define DA7218_AGS_ENABLE 0x3C +#define DA7218_AGS_TRIGGER 0x3D +#define DA7218_AGS_ATT_MAX 0x3E +#define DA7218_AGS_TIMEOUT 0x3F +#define DA7218_AGS_ANTICLIP_CTRL 0x40 +#define DA7218_CALIB_CTRL 0x44 +#define DA7218_CALIB_OFFSET_AUTO_M_1 0x45 +#define DA7218_CALIB_OFFSET_AUTO_U_1 0x46 +#define DA7218_CALIB_OFFSET_AUTO_M_2 0x47 +#define DA7218_CALIB_OFFSET_AUTO_U_2 0x48 +#define DA7218_ENV_TRACK_CTRL 0x4C +#define DA7218_LVL_DET_CTRL 0x50 +#define DA7218_LVL_DET_LEVEL 0x51 +#define DA7218_DGS_TRIGGER 0x54 +#define DA7218_DGS_ENABLE 0x55 +#define DA7218_DGS_RISE_FALL 0x56 +#define DA7218_DGS_SYNC_DELAY 0x57 +#define DA7218_DGS_SYNC_DELAY2 0x58 +#define DA7218_DGS_SYNC_DELAY3 0x59 +#define DA7218_DGS_LEVELS 0x5A +#define DA7218_DGS_GAIN_CTRL 0x5B +#define DA7218_DROUTING_OUTDAI_1L 0x5C +#define DA7218_DMIX_OUTDAI_1L_INFILT_1L_GAIN 0x5D +#define DA7218_DMIX_OUTDAI_1L_INFILT_1R_GAIN 0x5E +#define DA7218_DMIX_OUTDAI_1L_INFILT_2L_GAIN 0x5F +#define DA7218_DMIX_OUTDAI_1L_INFILT_2R_GAIN 0x60 +#define DA7218_DMIX_OUTDAI_1L_TONEGEN_GAIN 0x61 +#define DA7218_DMIX_OUTDAI_1L_INDAI_1L_GAIN 0x62 +#define DA7218_DMIX_OUTDAI_1L_INDAI_1R_GAIN 0x63 +#define DA7218_DROUTING_OUTDAI_1R 0x64 +#define DA7218_DMIX_OUTDAI_1R_INFILT_1L_GAIN 0x65 +#define DA7218_DMIX_OUTDAI_1R_INFILT_1R_GAIN 0x66 +#define DA7218_DMIX_OUTDAI_1R_INFILT_2L_GAIN 0x67 +#define DA7218_DMIX_OUTDAI_1R_INFILT_2R_GAIN 0x68 +#define DA7218_DMIX_OUTDAI_1R_TONEGEN_GAIN 0x69 +#define DA7218_DMIX_OUTDAI_1R_INDAI_1L_GAIN 0x6A +#define DA7218_DMIX_OUTDAI_1R_INDAI_1R_GAIN 0x6B +#define DA7218_DROUTING_OUTFILT_1L 0x6C +#define DA7218_DMIX_OUTFILT_1L_INFILT_1L_GAIN 0x6D +#define DA7218_DMIX_OUTFILT_1L_INFILT_1R_GAIN 0x6E +#define DA7218_DMIX_OUTFILT_1L_INFILT_2L_GAIN 0x6F +#define DA7218_DMIX_OUTFILT_1L_INFILT_2R_GAIN 0x70 +#define DA7218_DMIX_OUTFILT_1L_TONEGEN_GAIN 0x71 +#define DA7218_DMIX_OUTFILT_1L_INDAI_1L_GAIN 0x72 +#define DA7218_DMIX_OUTFILT_1L_INDAI_1R_GAIN 0x73 +#define DA7218_DROUTING_OUTFILT_1R 0x74 +#define DA7218_DMIX_OUTFILT_1R_INFILT_1L_GAIN 0x75 +#define DA7218_DMIX_OUTFILT_1R_INFILT_1R_GAIN 0x76 +#define DA7218_DMIX_OUTFILT_1R_INFILT_2L_GAIN 0x77 +#define DA7218_DMIX_OUTFILT_1R_INFILT_2R_GAIN 0x78 +#define DA7218_DMIX_OUTFILT_1R_TONEGEN_GAIN 0x79 +#define DA7218_DMIX_OUTFILT_1R_INDAI_1L_GAIN 0x7A +#define DA7218_DMIX_OUTFILT_1R_INDAI_1R_GAIN 0x7B +#define DA7218_DROUTING_OUTDAI_2L 0x7C +#define DA7218_DMIX_OUTDAI_2L_INFILT_1L_GAIN 0x7D +#define DA7218_DMIX_OUTDAI_2L_INFILT_1R_GAIN 0x7E +#define DA7218_DMIX_OUTDAI_2L_INFILT_2L_GAIN 0x7F +#define DA7218_DMIX_OUTDAI_2L_INFILT_2R_GAIN 0x80 +#define DA7218_DMIX_OUTDAI_2L_TONEGEN_GAIN 0x81 +#define DA7218_DMIX_OUTDAI_2L_INDAI_1L_GAIN 0x82 +#define DA7218_DMIX_OUTDAI_2L_INDAI_1R_GAIN 0x83 +#define DA7218_DROUTING_OUTDAI_2R 0x84 +#define DA7218_DMIX_OUTDAI_2R_INFILT_1L_GAIN 0x85 +#define DA7218_DMIX_OUTDAI_2R_INFILT_1R_GAIN 0x86 +#define DA7218_DMIX_OUTDAI_2R_INFILT_2L_GAIN 0x87 +#define DA7218_DMIX_OUTDAI_2R_INFILT_2R_GAIN 0x88 +#define DA7218_DMIX_OUTDAI_2R_TONEGEN_GAIN 0x89 +#define DA7218_DMIX_OUTDAI_2R_INDAI_1L_GAIN 0x8A +#define DA7218_DMIX_OUTDAI_2R_INDAI_1R_GAIN 0x8B +#define DA7218_DAI_CTRL 0x8C +#define DA7218_DAI_TDM_CTRL 0x8D +#define DA7218_DAI_OFFSET_LOWER 0x8E +#define DA7218_DAI_OFFSET_UPPER 0x8F +#define DA7218_DAI_CLK_MODE 0x90 +#define DA7218_PLL_CTRL 0x91 +#define DA7218_PLL_FRAC_TOP 0x92 +#define DA7218_PLL_FRAC_BOT 0x93 +#define DA7218_PLL_INTEGER 0x94 +#define DA7218_PLL_STATUS 0x95 +#define DA7218_PLL_REFOSC_CAL 0x98 +#define DA7218_DAC_NG_CTRL 0x9C +#define DA7218_DAC_NG_SETUP_TIME 0x9D +#define DA7218_DAC_NG_OFF_THRESH 0x9E +#define DA7218_DAC_NG_ON_THRESH 0x9F +#define DA7218_TONE_GEN_CFG1 0xA0 +#define DA7218_TONE_GEN_CFG2 0xA1 +#define DA7218_TONE_GEN_FREQ1_L 0xA2 +#define DA7218_TONE_GEN_FREQ1_U 0xA3 +#define DA7218_TONE_GEN_FREQ2_L 0xA4 +#define DA7218_TONE_GEN_FREQ2_U 0xA5 +#define DA7218_TONE_GEN_CYCLES 0xA6 +#define DA7218_TONE_GEN_ON_PER 0xA7 +#define DA7218_TONE_GEN_OFF_PER 0xA8 +#define DA7218_CP_CTRL 0xAC +#define DA7218_CP_DELAY 0xAD +#define DA7218_CP_VOL_THRESHOLD1 0xAE +#define DA7218_MIC_1_CTRL 0xB4 +#define DA7218_MIC_1_GAIN 0xB5 +#define DA7218_MIC_1_SELECT 0xB7 +#define DA7218_MIC_2_CTRL 0xB8 +#define DA7218_MIC_2_GAIN 0xB9 +#define DA7218_MIC_2_SELECT 0xBB +#define DA7218_IN_1_HPF_FILTER_CTRL 0xBC +#define DA7218_IN_2_HPF_FILTER_CTRL 0xBD +#define DA7218_ADC_1_CTRL 0xC0 +#define DA7218_ADC_2_CTRL 0xC1 +#define DA7218_ADC_MODE 0xC2 +#define DA7218_MIXOUT_L_CTRL 0xCC +#define DA7218_MIXOUT_L_GAIN 0xCD +#define DA7218_MIXOUT_R_CTRL 0xCE +#define DA7218_MIXOUT_R_GAIN 0xCF +#define DA7218_HP_L_CTRL 0xD0 +#define DA7218_HP_L_GAIN 0xD1 +#define DA7218_HP_R_CTRL 0xD2 +#define DA7218_HP_R_GAIN 0xD3 +#define DA7218_HP_SNGL_CTRL 0xD4 +#define DA7218_HP_DIFF_CTRL 0xD5 +#define DA7218_HP_DIFF_UNLOCK 0xD7 +#define DA7218_HPLDET_JACK 0xD8 +#define DA7218_HPLDET_CTRL 0xD9 +#define DA7218_HPLDET_TEST 0xDA +#define DA7218_REFERENCES 0xDC +#define DA7218_IO_CTRL 0xE0 +#define DA7218_LDO_CTRL 0xE1 +#define DA7218_SIDETONE_CTRL 0xE4 +#define DA7218_SIDETONE_IN_SELECT 0xE5 +#define DA7218_SIDETONE_GAIN 0xE6 +#define DA7218_DROUTING_ST_OUTFILT_1L 0xE8 +#define DA7218_DROUTING_ST_OUTFILT_1R 0xE9 +#define DA7218_SIDETONE_BIQ_3STAGE_DATA 0xEA +#define DA7218_SIDETONE_BIQ_3STAGE_ADDR 0xEB +#define DA7218_EVENT_STATUS 0xEC +#define DA7218_EVENT 0xED +#define DA7218_EVENT_MASK 0xEE +#define DA7218_DMIC_1_CTRL 0xF0 +#define DA7218_DMIC_2_CTRL 0xF1 +#define DA7218_IN_1L_GAIN 0xF4 +#define DA7218_IN_1R_GAIN 0xF5 +#define DA7218_IN_2L_GAIN 0xF6 +#define DA7218_IN_2R_GAIN 0xF7 +#define DA7218_OUT_1L_GAIN 0xF8 +#define DA7218_OUT_1R_GAIN 0xF9 +#define DA7218_MICBIAS_CTRL 0xFC +#define DA7218_MICBIAS_EN 0xFD + + +/* + * Bit Fields + */ + +#define DA7218_SWITCH_EN_MAX 0x1 + +/* DA7218_SYSTEM_ACTIVE = 0x0 */ +#define DA7218_SYSTEM_ACTIVE_SHIFT 0 +#define DA7218_SYSTEM_ACTIVE_MASK (0x1 << 0) + +/* DA7218_CIF_CTRL = 0x1 */ +#define DA7218_CIF_I2C_WRITE_MODE_SHIFT 0 +#define DA7218_CIF_I2C_WRITE_MODE_MASK (0x1 << 0) + +/* DA7218_CHIP_ID1 = 0x4 */ +#define DA7218_CHIP_ID1_SHIFT 0 +#define DA7218_CHIP_ID1_MASK (0xFF << 0) + +/* DA7218_CHIP_ID2 = 0x5 */ +#define DA7218_CHIP_ID2_SHIFT 0 +#define DA7218_CHIP_ID2_MASK (0xFF << 0) + +/* DA7218_CHIP_REVISION = 0x6 */ +#define DA7218_CHIP_MINOR_SHIFT 0 +#define DA7218_CHIP_MINOR_MASK (0xF << 0) +#define DA7218_CHIP_MAJOR_SHIFT 4 +#define DA7218_CHIP_MAJOR_MASK (0xF << 4) + +/* DA7218_SPARE1 = 0x7 */ +#define DA7218_SPARE1_SHIFT 0 +#define DA7218_SPARE1_MASK (0xFF << 0) + +/* DA7218_STATUS1 = 0x8 */ +#define DA7218_STATUS_SPARE1_SHIFT 0 +#define DA7218_STATUS_SPARE1_MASK (0xFF << 0) + +/* DA7218_SOFT_RESET = 0x9 */ +#define DA7218_CIF_REG_SOFT_RESET_SHIFT 7 +#define DA7218_CIF_REG_SOFT_RESET_MASK (0x1 << 7) + +/* DA7218_SR = 0xB */ +#define DA7218_SR_ADC_SHIFT 0 +#define DA7218_SR_ADC_MASK (0xF << 0) +#define DA7218_SR_DAC_SHIFT 4 +#define DA7218_SR_DAC_MASK (0xF << 4) +#define DA7218_SR_8000 0x01 +#define DA7218_SR_11025 0x02 +#define DA7218_SR_12000 0x03 +#define DA7218_SR_16000 0x05 +#define DA7218_SR_22050 0x06 +#define DA7218_SR_24000 0x07 +#define DA7218_SR_32000 0x09 +#define DA7218_SR_44100 0x0A +#define DA7218_SR_48000 0x0B +#define DA7218_SR_88200 0x0E +#define DA7218_SR_96000 0x0F + +/* DA7218_PC_COUNT = 0xC */ +#define DA7218_PC_FREERUN_SHIFT 0 +#define DA7218_PC_FREERUN_MASK (0x1 << 0) +#define DA7218_PC_RESYNC_AUTO_SHIFT 1 +#define DA7218_PC_RESYNC_AUTO_MASK (0x1 << 1) + +/* DA7218_GAIN_RAMP_CTRL = 0xD */ +#define DA7218_GAIN_RAMP_RATE_SHIFT 0 +#define DA7218_GAIN_RAMP_RATE_MASK (0x3 << 0) +#define DA7218_GAIN_RAMP_RATE_MAX 4 + +/* DA7218_CIF_TIMEOUT_CTRL = 0x10 */ +#define DA7218_I2C_TIMEOUT_EN_SHIFT 0 +#define DA7218_I2C_TIMEOUT_EN_MASK (0x1 << 0) + +/* DA7218_SYSTEM_MODES_INPUT = 0x14 */ +#define DA7218_MODE_SUBMIT_SHIFT 0 +#define DA7218_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7218_ADC_MODE_SHIFT 1 +#define DA7218_ADC_MODE_MASK (0x7F << 1) + +/* DA7218_SYSTEM_MODES_OUTPUT = 0x15 */ +#define DA7218_MODE_SUBMIT_SHIFT 0 +#define DA7218_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7218_DAC_MODE_SHIFT 1 +#define DA7218_DAC_MODE_MASK (0x7F << 1) + +/* DA7218_SYSTEM_STATUS = 0x16 */ +#define DA7218_SC1_BUSY_SHIFT 0 +#define DA7218_SC1_BUSY_MASK (0x1 << 0) +#define DA7218_SC2_BUSY_SHIFT 1 +#define DA7218_SC2_BUSY_MASK (0x1 << 1) + +/* DA7218_IN_1L_FILTER_CTRL = 0x18 */ +#define DA7218_IN_1L_RAMP_EN_SHIFT 5 +#define DA7218_IN_1L_RAMP_EN_MASK (0x1 << 5) +#define DA7218_IN_1L_MUTE_EN_SHIFT 6 +#define DA7218_IN_1L_MUTE_EN_MASK (0x1 << 6) +#define DA7218_IN_1L_FILTER_EN_SHIFT 7 +#define DA7218_IN_1L_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_IN_1R_FILTER_CTRL = 0x19 */ +#define DA7218_IN_1R_RAMP_EN_SHIFT 5 +#define DA7218_IN_1R_RAMP_EN_MASK (0x1 << 5) +#define DA7218_IN_1R_MUTE_EN_SHIFT 6 +#define DA7218_IN_1R_MUTE_EN_MASK (0x1 << 6) +#define DA7218_IN_1R_FILTER_EN_SHIFT 7 +#define DA7218_IN_1R_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_IN_2L_FILTER_CTRL = 0x1A */ +#define DA7218_IN_2L_RAMP_EN_SHIFT 5 +#define DA7218_IN_2L_RAMP_EN_MASK (0x1 << 5) +#define DA7218_IN_2L_MUTE_EN_SHIFT 6 +#define DA7218_IN_2L_MUTE_EN_MASK (0x1 << 6) +#define DA7218_IN_2L_FILTER_EN_SHIFT 7 +#define DA7218_IN_2L_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_IN_2R_FILTER_CTRL = 0x1B */ +#define DA7218_IN_2R_RAMP_EN_SHIFT 5 +#define DA7218_IN_2R_RAMP_EN_MASK (0x1 << 5) +#define DA7218_IN_2R_MUTE_EN_SHIFT 6 +#define DA7218_IN_2R_MUTE_EN_MASK (0x1 << 6) +#define DA7218_IN_2R_FILTER_EN_SHIFT 7 +#define DA7218_IN_2R_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_OUT_1L_FILTER_CTRL = 0x20 */ +#define DA7218_OUT_1L_BIQ_5STAGE_SEL_SHIFT 3 +#define DA7218_OUT_1L_BIQ_5STAGE_SEL_MASK (0x1 << 3) +#define DA7218_OUT_BIQ_5STAGE_SEL_MAX 2 +#define DA7218_OUT_1L_SUBRANGE_EN_SHIFT 4 +#define DA7218_OUT_1L_SUBRANGE_EN_MASK (0x1 << 4) +#define DA7218_OUT_1L_RAMP_EN_SHIFT 5 +#define DA7218_OUT_1L_RAMP_EN_MASK (0x1 << 5) +#define DA7218_OUT_1L_MUTE_EN_SHIFT 6 +#define DA7218_OUT_1L_MUTE_EN_MASK (0x1 << 6) +#define DA7218_OUT_1L_FILTER_EN_SHIFT 7 +#define DA7218_OUT_1L_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_OUT_1R_FILTER_CTRL = 0x21 */ +#define DA7218_OUT_1R_BIQ_5STAGE_SEL_SHIFT 3 +#define DA7218_OUT_1R_BIQ_5STAGE_SEL_MASK (0x1 << 3) +#define DA7218_OUT_1R_SUBRANGE_EN_SHIFT 4 +#define DA7218_OUT_1R_SUBRANGE_EN_MASK (0x1 << 4) +#define DA7218_OUT_1R_RAMP_EN_SHIFT 5 +#define DA7218_OUT_1R_RAMP_EN_MASK (0x1 << 5) +#define DA7218_OUT_1R_MUTE_EN_SHIFT 6 +#define DA7218_OUT_1R_MUTE_EN_MASK (0x1 << 6) +#define DA7218_OUT_1R_FILTER_EN_SHIFT 7 +#define DA7218_OUT_1R_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_OUT_1_HPF_FILTER_CTRL = 0x24 */ +#define DA7218_OUT_1_VOICE_HPF_CORNER_SHIFT 0 +#define DA7218_OUT_1_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7218_VOICE_HPF_CORNER_MAX 8 +#define DA7218_OUT_1_VOICE_EN_SHIFT 3 +#define DA7218_OUT_1_VOICE_EN_MASK (0x1 << 3) +#define DA7218_OUT_1_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7218_OUT_1_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7218_AUDIO_HPF_CORNER_MAX 4 +#define DA7218_OUT_1_HPF_EN_SHIFT 7 +#define DA7218_OUT_1_HPF_EN_MASK (0x1 << 7) +#define DA7218_HPF_MODE_SHIFT 0 +#define DA7218_HPF_DISABLED ((0x0 << 3) | (0x0 << 7)) +#define DA7218_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7)) +#define DA7218_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7)) +#define DA7218_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7)) +#define DA7218_HPF_MODE_MAX 3 + +/* DA7218_OUT_1_EQ_12_FILTER_CTRL = 0x25 */ +#define DA7218_OUT_1_EQ_BAND1_SHIFT 0 +#define DA7218_OUT_1_EQ_BAND1_MASK (0xF << 0) +#define DA7218_OUT_EQ_BAND_MAX 0xF +#define DA7218_OUT_1_EQ_BAND2_SHIFT 4 +#define DA7218_OUT_1_EQ_BAND2_MASK (0xF << 4) + +/* DA7218_OUT_1_EQ_34_FILTER_CTRL = 0x26 */ +#define DA7218_OUT_1_EQ_BAND3_SHIFT 0 +#define DA7218_OUT_1_EQ_BAND3_MASK (0xF << 0) +#define DA7218_OUT_1_EQ_BAND4_SHIFT 4 +#define DA7218_OUT_1_EQ_BAND4_MASK (0xF << 4) + +/* DA7218_OUT_1_EQ_5_FILTER_CTRL = 0x27 */ +#define DA7218_OUT_1_EQ_BAND5_SHIFT 0 +#define DA7218_OUT_1_EQ_BAND5_MASK (0xF << 0) +#define DA7218_OUT_1_EQ_EN_SHIFT 7 +#define DA7218_OUT_1_EQ_EN_MASK (0x1 << 7) + +/* DA7218_OUT_1_BIQ_5STAGE_CTRL = 0x28 */ +#define DA7218_OUT_1_BIQ_5STAGE_MUTE_EN_SHIFT 6 +#define DA7218_OUT_1_BIQ_5STAGE_MUTE_EN_MASK (0x1 << 6) +#define DA7218_OUT_1_BIQ_5STAGE_FILTER_EN_SHIFT 7 +#define DA7218_OUT_1_BIQ_5STAGE_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_OUT_1_BIQ_5STAGE_DATA = 0x29 */ +#define DA7218_OUT_1_BIQ_5STAGE_DATA_SHIFT 0 +#define DA7218_OUT_1_BIQ_5STAGE_DATA_MASK (0xFF << 0) + +/* DA7218_OUT_1_BIQ_5STAGE_ADDR = 0x2A */ +#define DA7218_OUT_1_BIQ_5STAGE_ADDR_SHIFT 0 +#define DA7218_OUT_1_BIQ_5STAGE_ADDR_MASK (0x3F << 0) +#define DA7218_OUT_1_BIQ_5STAGE_CFG_SIZE 50 + +/* DA7218_MIXIN_1_CTRL = 0x2C */ +#define DA7218_MIXIN_1_MIX_SEL_SHIFT 3 +#define DA7218_MIXIN_1_MIX_SEL_MASK (0x1 << 3) +#define DA7218_MIXIN_1_AMP_ZC_EN_SHIFT 4 +#define DA7218_MIXIN_1_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7218_MIXIN_1_AMP_RAMP_EN_SHIFT 5 +#define DA7218_MIXIN_1_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7218_MIXIN_1_AMP_MUTE_EN_SHIFT 6 +#define DA7218_MIXIN_1_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7218_MIXIN_1_AMP_EN_SHIFT 7 +#define DA7218_MIXIN_1_AMP_EN_MASK (0x1 << 7) + +/* DA7218_MIXIN_1_GAIN = 0x2D */ +#define DA7218_MIXIN_1_AMP_GAIN_SHIFT 0 +#define DA7218_MIXIN_1_AMP_GAIN_MASK (0xF << 0) +#define DA7218_MIXIN_AMP_GAIN_MAX 0xF + +/* DA7218_MIXIN_2_CTRL = 0x2E */ +#define DA7218_MIXIN_2_MIX_SEL_SHIFT 3 +#define DA7218_MIXIN_2_MIX_SEL_MASK (0x1 << 3) +#define DA7218_MIXIN_2_AMP_ZC_EN_SHIFT 4 +#define DA7218_MIXIN_2_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7218_MIXIN_2_AMP_RAMP_EN_SHIFT 5 +#define DA7218_MIXIN_2_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7218_MIXIN_2_AMP_MUTE_EN_SHIFT 6 +#define DA7218_MIXIN_2_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7218_MIXIN_2_AMP_EN_SHIFT 7 +#define DA7218_MIXIN_2_AMP_EN_MASK (0x1 << 7) + +/* DA7218_MIXIN_2_GAIN = 0x2F */ +#define DA7218_MIXIN_2_AMP_GAIN_SHIFT 0 +#define DA7218_MIXIN_2_AMP_GAIN_MASK (0xF << 0) + +/* DA7218_ALC_CTRL1 = 0x30 */ +#define DA7218_ALC_EN_SHIFT 0 +#define DA7218_ALC_EN_MASK (0xF << 0) +#define DA7218_ALC_CHAN1_L_EN_SHIFT 0 +#define DA7218_ALC_CHAN1_R_EN_SHIFT 1 +#define DA7218_ALC_CHAN2_L_EN_SHIFT 2 +#define DA7218_ALC_CHAN2_R_EN_SHIFT 3 +#define DA7218_ALC_SYNC_MODE_SHIFT 4 +#define DA7218_ALC_SYNC_MODE_MASK (0xF << 4) +#define DA7218_ALC_SYNC_MODE_CH1 (0x1 << 4) +#define DA7218_ALC_SYNC_MODE_CH2 (0x4 << 4) + +/* DA7218_ALC_CTRL2 = 0x31 */ +#define DA7218_ALC_ATTACK_SHIFT 0 +#define DA7218_ALC_ATTACK_MASK (0xF << 0) +#define DA7218_ALC_ATTACK_MAX 13 +#define DA7218_ALC_RELEASE_SHIFT 4 +#define DA7218_ALC_RELEASE_MASK (0xF << 4) +#define DA7218_ALC_RELEASE_MAX 11 + +/* DA7218_ALC_CTRL3 = 0x32 */ +#define DA7218_ALC_HOLD_SHIFT 0 +#define DA7218_ALC_HOLD_MASK (0xF << 0) +#define DA7218_ALC_HOLD_MAX 16 + +/* DA7218_ALC_NOISE = 0x33 */ +#define DA7218_ALC_NOISE_SHIFT 0 +#define DA7218_ALC_NOISE_MASK (0x3F << 0) +#define DA7218_ALC_THRESHOLD_MAX 0x3F + +/* DA7218_ALC_TARGET_MIN = 0x34 */ +#define DA7218_ALC_THRESHOLD_MIN_SHIFT 0 +#define DA7218_ALC_THRESHOLD_MIN_MASK (0x3F << 0) + +/* DA7218_ALC_TARGET_MAX = 0x35 */ +#define DA7218_ALC_THRESHOLD_MAX_SHIFT 0 +#define DA7218_ALC_THRESHOLD_MAX_MASK (0x3F << 0) + +/* DA7218_ALC_GAIN_LIMITS = 0x36 */ +#define DA7218_ALC_ATTEN_MAX_SHIFT 0 +#define DA7218_ALC_ATTEN_MAX_MASK (0xF << 0) +#define DA7218_ALC_ATTEN_GAIN_MAX 0xF +#define DA7218_ALC_GAIN_MAX_SHIFT 4 +#define DA7218_ALC_GAIN_MAX_MASK (0xF << 4) + +/* DA7218_ALC_ANA_GAIN_LIMITS = 0x37 */ +#define DA7218_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7218_ALC_ANA_GAIN_MIN_MASK (0x7 << 0) +#define DA7218_ALC_ANA_GAIN_MIN 0x1 +#define DA7218_ALC_ANA_GAIN_MAX 0x7 +#define DA7218_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7218_ALC_ANA_GAIN_MAX_MASK (0x7 << 4) + +/* DA7218_ALC_ANTICLIP_CTRL = 0x38 */ +#define DA7218_ALC_ANTICLIP_STEP_SHIFT 0 +#define DA7218_ALC_ANTICLIP_STEP_MASK (0x3 << 0) +#define DA7218_ALC_ANTICLIP_STEP_MAX 4 +#define DA7218_ALC_ANTICLIP_EN_SHIFT 7 +#define DA7218_ALC_ANTICLIP_EN_MASK (0x1 << 7) + +/* DA7218_AGS_ENABLE = 0x3C */ +#define DA7218_AGS_ENABLE_SHIFT 0 +#define DA7218_AGS_ENABLE_MASK (0x3 << 0) +#define DA7218_AGS_ENABLE_CHAN1_SHIFT 0 +#define DA7218_AGS_ENABLE_CHAN2_SHIFT 1 + +/* DA7218_AGS_TRIGGER = 0x3D */ +#define DA7218_AGS_TRIGGER_SHIFT 0 +#define DA7218_AGS_TRIGGER_MASK (0xF << 0) +#define DA7218_AGS_TRIGGER_MAX 0xF + +/* DA7218_AGS_ATT_MAX = 0x3E */ +#define DA7218_AGS_ATT_MAX_SHIFT 0 +#define DA7218_AGS_ATT_MAX_MASK (0x7 << 0) +#define DA7218_AGS_ATT_MAX_MAX 0x7 + +/* DA7218_AGS_TIMEOUT = 0x3F */ +#define DA7218_AGS_TIMEOUT_EN_SHIFT 0 +#define DA7218_AGS_TIMEOUT_EN_MASK (0x1 << 0) + +/* DA7218_AGS_ANTICLIP_CTRL = 0x40 */ +#define DA7218_AGS_ANTICLIP_EN_SHIFT 7 +#define DA7218_AGS_ANTICLIP_EN_MASK (0x1 << 7) + +/* DA7218_CALIB_CTRL = 0x44 */ +#define DA7218_CALIB_OFFSET_EN_SHIFT 0 +#define DA7218_CALIB_OFFSET_EN_MASK (0x1 << 0) +#define DA7218_CALIB_AUTO_EN_SHIFT 2 +#define DA7218_CALIB_AUTO_EN_MASK (0x1 << 2) +#define DA7218_CALIB_OVERFLOW_SHIFT 3 +#define DA7218_CALIB_OVERFLOW_MASK (0x1 << 3) + +/* DA7218_CALIB_OFFSET_AUTO_M_1 = 0x45 */ +#define DA7218_CALIB_OFFSET_AUTO_M_1_SHIFT 0 +#define DA7218_CALIB_OFFSET_AUTO_M_1_MASK (0xFF << 0) + +/* DA7218_CALIB_OFFSET_AUTO_U_1 = 0x46 */ +#define DA7218_CALIB_OFFSET_AUTO_U_1_SHIFT 0 +#define DA7218_CALIB_OFFSET_AUTO_U_1_MASK (0xF << 0) + +/* DA7218_CALIB_OFFSET_AUTO_M_2 = 0x47 */ +#define DA7218_CALIB_OFFSET_AUTO_M_2_SHIFT 0 +#define DA7218_CALIB_OFFSET_AUTO_M_2_MASK (0xFF << 0) + +/* DA7218_CALIB_OFFSET_AUTO_U_2 = 0x48 */ +#define DA7218_CALIB_OFFSET_AUTO_U_2_SHIFT 0 +#define DA7218_CALIB_OFFSET_AUTO_U_2_MASK (0xF << 0) + +/* DA7218_ENV_TRACK_CTRL = 0x4C */ +#define DA7218_INTEG_ATTACK_SHIFT 0 +#define DA7218_INTEG_ATTACK_MASK (0x3 << 0) +#define DA7218_INTEG_RELEASE_SHIFT 4 +#define DA7218_INTEG_RELEASE_MASK (0x3 << 4) +#define DA7218_INTEG_MAX 4 + +/* DA7218_LVL_DET_CTRL = 0x50 */ +#define DA7218_LVL_DET_EN_SHIFT 0 +#define DA7218_LVL_DET_EN_MASK (0xF << 0) +#define DA7218_LVL_DET_EN_CHAN1L_SHIFT 0 +#define DA7218_LVL_DET_EN_CHAN1R_SHIFT 1 +#define DA7218_LVL_DET_EN_CHAN2L_SHIFT 2 +#define DA7218_LVL_DET_EN_CHAN2R_SHIFT 3 + +/* DA7218_LVL_DET_LEVEL = 0x51 */ +#define DA7218_LVL_DET_LEVEL_SHIFT 0 +#define DA7218_LVL_DET_LEVEL_MASK (0x7F << 0) +#define DA7218_LVL_DET_LEVEL_MAX 0x7F + +/* DA7218_DGS_TRIGGER = 0x54 */ +#define DA7218_DGS_TRIGGER_LVL_SHIFT 0 +#define DA7218_DGS_TRIGGER_LVL_MASK (0x3F << 0) +#define DA7218_DGS_TRIGGER_MAX 0x3F + +/* DA7218_DGS_ENABLE = 0x55 */ +#define DA7218_DGS_ENABLE_SHIFT 0 +#define DA7218_DGS_ENABLE_MASK (0x3 << 0) +#define DA7218_DGS_ENABLE_L_SHIFT 0 +#define DA7218_DGS_ENABLE_R_SHIFT 1 + +/* DA7218_DGS_RISE_FALL = 0x56 */ +#define DA7218_DGS_RISE_COEFF_SHIFT 0 +#define DA7218_DGS_RISE_COEFF_MASK (0x7 << 0) +#define DA7218_DGS_RISE_COEFF_MAX 7 +#define DA7218_DGS_FALL_COEFF_SHIFT 4 +#define DA7218_DGS_FALL_COEFF_MASK (0x7 << 4) +#define DA7218_DGS_FALL_COEFF_MAX 8 + +/* DA7218_DGS_SYNC_DELAY = 0x57 */ +#define DA7218_DGS_SYNC_DELAY_SHIFT 0 +#define DA7218_DGS_SYNC_DELAY_MASK (0xFF << 0) +#define DA7218_DGS_SYNC_DELAY_MAX 0xFF + +/* DA7218_DGS_SYNC_DELAY2 = 0x58 */ +#define DA7218_DGS_SYNC_DELAY2_SHIFT 0 +#define DA7218_DGS_SYNC_DELAY2_MASK (0xFF << 0) + +/* DA7218_DGS_SYNC_DELAY3 = 0x59 */ +#define DA7218_DGS_SYNC_DELAY3_SHIFT 0 +#define DA7218_DGS_SYNC_DELAY3_MASK (0x7F << 0) +#define DA7218_DGS_SYNC_DELAY3_MAX 0x7F + +/* DA7218_DGS_LEVELS = 0x5A */ +#define DA7218_DGS_ANTICLIP_LVL_SHIFT 0 +#define DA7218_DGS_ANTICLIP_LVL_MASK (0x7 << 0) +#define DA7218_DGS_ANTICLIP_LVL_MAX 0x7 +#define DA7218_DGS_SIGNAL_LVL_SHIFT 4 +#define DA7218_DGS_SIGNAL_LVL_MASK (0xF << 4) +#define DA7218_DGS_SIGNAL_LVL_MAX 0xF + +/* DA7218_DGS_GAIN_CTRL = 0x5B */ +#define DA7218_DGS_STEPS_SHIFT 0 +#define DA7218_DGS_STEPS_MASK (0x1F << 0) +#define DA7218_DGS_STEPS_MAX 0x1F +#define DA7218_DGS_RAMP_EN_SHIFT 5 +#define DA7218_DGS_RAMP_EN_MASK (0x1 << 5) +#define DA7218_DGS_SUBR_EN_SHIFT 6 +#define DA7218_DGS_SUBR_EN_MASK (0x1 << 6) + +/* DA7218_DROUTING_OUTDAI_1L = 0x5C */ +#define DA7218_OUTDAI_1L_SRC_SHIFT 0 +#define DA7218_OUTDAI_1L_SRC_MASK (0x7F << 0) +#define DA7218_DMIX_SRC_INFILT1L 0 +#define DA7218_DMIX_SRC_INFILT1R 1 +#define DA7218_DMIX_SRC_INFILT2L 2 +#define DA7218_DMIX_SRC_INFILT2R 3 +#define DA7218_DMIX_SRC_TONEGEN 4 +#define DA7218_DMIX_SRC_DAIL 5 +#define DA7218_DMIX_SRC_DAIR 6 + +/* DA7218_DMIX_OUTDAI_1L_INFILT_1L_GAIN = 0x5D */ +#define DA7218_OUTDAI_1L_INFILT_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_INFILT_1L_GAIN_MASK (0x1F << 0) +#define DA7218_DMIX_GAIN_MAX 0x1F + +/* DA7218_DMIX_OUTDAI_1L_INFILT_1R_GAIN = 0x5E */ +#define DA7218_OUTDAI_1L_INFILT_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_INFILT_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1L_INFILT_2L_GAIN = 0x5F */ +#define DA7218_OUTDAI_1L_INFILT_2L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_INFILT_2L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1L_INFILT_2R_GAIN = 0x60 */ +#define DA7218_OUTDAI_1L_INFILT_2R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_INFILT_2R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1L_TONEGEN_GAIN = 0x61 */ +#define DA7218_OUTDAI_1L_TONEGEN_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_TONEGEN_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1L_INDAI_1L_GAIN = 0x62 */ +#define DA7218_OUTDAI_1L_INDAI_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_INDAI_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1L_INDAI_1R_GAIN = 0x63 */ +#define DA7218_OUTDAI_1L_INDAI_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1L_INDAI_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DROUTING_OUTDAI_1R = 0x64 */ +#define DA7218_OUTDAI_1R_SRC_SHIFT 0 +#define DA7218_OUTDAI_1R_SRC_MASK (0x7F << 0) + +/* DA7218_DMIX_OUTDAI_1R_INFILT_1L_GAIN = 0x65 */ +#define DA7218_OUTDAI_1R_INFILT_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_INFILT_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1R_INFILT_1R_GAIN = 0x66 */ +#define DA7218_OUTDAI_1R_INFILT_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_INFILT_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1R_INFILT_2L_GAIN = 0x67 */ +#define DA7218_OUTDAI_1R_INFILT_2L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_INFILT_2L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1R_INFILT_2R_GAIN = 0x68 */ +#define DA7218_OUTDAI_1R_INFILT_2R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_INFILT_2R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1R_TONEGEN_GAIN = 0x69 */ +#define DA7218_OUTDAI_1R_TONEGEN_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_TONEGEN_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1R_INDAI_1L_GAIN = 0x6A */ +#define DA7218_OUTDAI_1R_INDAI_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_INDAI_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_1R_INDAI_1R_GAIN = 0x6B */ +#define DA7218_OUTDAI_1R_INDAI_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_1R_INDAI_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DROUTING_OUTFILT_1L = 0x6C */ +#define DA7218_OUTFILT_1L_SRC_SHIFT 0 +#define DA7218_OUTFILT_1L_SRC_MASK (0x7F << 0) + +/* DA7218_DMIX_OUTFILT_1L_INFILT_1L_GAIN = 0x6D */ +#define DA7218_OUTFILT_1L_INFILT_1L_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_INFILT_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1L_INFILT_1R_GAIN = 0x6E */ +#define DA7218_OUTFILT_1L_INFILT_1R_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_INFILT_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1L_INFILT_2L_GAIN = 0x6F */ +#define DA7218_OUTFILT_1L_INFILT_2L_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_INFILT_2L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1L_INFILT_2R_GAIN = 0x70 */ +#define DA7218_OUTFILT_1L_INFILT_2R_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_INFILT_2R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1L_TONEGEN_GAIN = 0x71 */ +#define DA7218_OUTFILT_1L_TONEGEN_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_TONEGEN_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1L_INDAI_1L_GAIN = 0x72 */ +#define DA7218_OUTFILT_1L_INDAI_1L_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_INDAI_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1L_INDAI_1R_GAIN = 0x73 */ +#define DA7218_OUTFILT_1L_INDAI_1R_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1L_INDAI_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DROUTING_OUTFILT_1R = 0x74 */ +#define DA7218_OUTFILT_1R_SRC_SHIFT 0 +#define DA7218_OUTFILT_1R_SRC_MASK (0x7F << 0) + +/* DA7218_DMIX_OUTFILT_1R_INFILT_1L_GAIN = 0x75 */ +#define DA7218_OUTFILT_1R_INFILT_1L_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_INFILT_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1R_INFILT_1R_GAIN = 0x76 */ +#define DA7218_OUTFILT_1R_INFILT_1R_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_INFILT_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1R_INFILT_2L_GAIN = 0x77 */ +#define DA7218_OUTFILT_1R_INFILT_2L_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_INFILT_2L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1R_INFILT_2R_GAIN = 0x78 */ +#define DA7218_OUTFILT_1R_INFILT_2R_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_INFILT_2R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1R_TONEGEN_GAIN = 0x79 */ +#define DA7218_OUTFILT_1R_TONEGEN_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_TONEGEN_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1R_INDAI_1L_GAIN = 0x7A */ +#define DA7218_OUTFILT_1R_INDAI_1L_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_INDAI_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTFILT_1R_INDAI_1R_GAIN = 0x7B */ +#define DA7218_OUTFILT_1R_INDAI_1R_GAIN_SHIFT 0 +#define DA7218_OUTFILT_1R_INDAI_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DROUTING_OUTDAI_2L = 0x7C */ +#define DA7218_OUTDAI_2L_SRC_SHIFT 0 +#define DA7218_OUTDAI_2L_SRC_MASK (0x7F << 0) + +/* DA7218_DMIX_OUTDAI_2L_INFILT_1L_GAIN = 0x7D */ +#define DA7218_OUTDAI_2L_INFILT_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_INFILT_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2L_INFILT_1R_GAIN = 0x7E */ +#define DA7218_OUTDAI_2L_INFILT_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_INFILT_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2L_INFILT_2L_GAIN = 0x7F */ +#define DA7218_OUTDAI_2L_INFILT_2L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_INFILT_2L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2L_INFILT_2R_GAIN = 0x80 */ +#define DA7218_OUTDAI_2L_INFILT_2R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_INFILT_2R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2L_TONEGEN_GAIN = 0x81 */ +#define DA7218_OUTDAI_2L_TONEGEN_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_TONEGEN_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2L_INDAI_1L_GAIN = 0x82 */ +#define DA7218_OUTDAI_2L_INDAI_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_INDAI_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2L_INDAI_1R_GAIN = 0x83 */ +#define DA7218_OUTDAI_2L_INDAI_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2L_INDAI_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DROUTING_OUTDAI_2R = 0x84 */ +#define DA7218_OUTDAI_2R_SRC_SHIFT 0 +#define DA7218_OUTDAI_2R_SRC_MASK (0x7F << 0) + +/* DA7218_DMIX_OUTDAI_2R_INFILT_1L_GAIN = 0x85 */ +#define DA7218_OUTDAI_2R_INFILT_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_INFILT_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2R_INFILT_1R_GAIN = 0x86 */ +#define DA7218_OUTDAI_2R_INFILT_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_INFILT_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2R_INFILT_2L_GAIN = 0x87 */ +#define DA7218_OUTDAI_2R_INFILT_2L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_INFILT_2L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2R_INFILT_2R_GAIN = 0x88 */ +#define DA7218_OUTDAI_2R_INFILT_2R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_INFILT_2R_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2R_TONEGEN_GAIN = 0x89 */ +#define DA7218_OUTDAI_2R_TONEGEN_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_TONEGEN_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2R_INDAI_1L_GAIN = 0x8A */ +#define DA7218_OUTDAI_2R_INDAI_1L_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_INDAI_1L_GAIN_MASK (0x1F << 0) + +/* DA7218_DMIX_OUTDAI_2R_INDAI_1R_GAIN = 0x8B */ +#define DA7218_OUTDAI_2R_INDAI_1R_GAIN_SHIFT 0 +#define DA7218_OUTDAI_2R_INDAI_1R_GAIN_MASK (0x1F << 0) + +/* DA7218_DAI_CTRL = 0x8C */ +#define DA7218_DAI_FORMAT_SHIFT 0 +#define DA7218_DAI_FORMAT_MASK (0x3 << 0) +#define DA7218_DAI_FORMAT_I2S (0x0 << 0) +#define DA7218_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7218_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7218_DAI_FORMAT_DSP (0x3 << 0) +#define DA7218_DAI_WORD_LENGTH_SHIFT 2 +#define DA7218_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7218_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7218_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7218_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7218_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7218_DAI_CH_NUM_SHIFT 4 +#define DA7218_DAI_CH_NUM_MASK (0x7 << 4) +#define DA7218_DAI_CH_NUM_MAX 4 +#define DA7218_DAI_EN_SHIFT 7 +#define DA7218_DAI_EN_MASK (0x1 << 7) + +/* DA7218_DAI_TDM_CTRL = 0x8D */ +#define DA7218_DAI_TDM_CH_EN_SHIFT 0 +#define DA7218_DAI_TDM_CH_EN_MASK (0xF << 0) +#define DA7218_DAI_TDM_MAX_SLOTS 4 +#define DA7218_DAI_OE_SHIFT 6 +#define DA7218_DAI_OE_MASK (0x1 << 6) +#define DA7218_DAI_TDM_MODE_EN_SHIFT 7 +#define DA7218_DAI_TDM_MODE_EN_MASK (0x1 << 7) + +/* DA7218_DAI_OFFSET_LOWER = 0x8E */ +#define DA7218_DAI_OFFSET_LOWER_SHIFT 0 +#define DA7218_DAI_OFFSET_LOWER_MASK (0xFF << 0) + +/* DA7218_DAI_OFFSET_UPPER = 0x8F */ +#define DA7218_DAI_OFFSET_UPPER_SHIFT 0 +#define DA7218_DAI_OFFSET_UPPER_MASK (0x7 << 0) + +/* DA7218_DAI_CLK_MODE = 0x90 */ +#define DA7218_DAI_BCLKS_PER_WCLK_SHIFT 0 +#define DA7218_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7218_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7218_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7218_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7218_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7218_DAI_CLK_POL_SHIFT 2 +#define DA7218_DAI_CLK_POL_MASK (0x1 << 2) +#define DA7218_DAI_CLK_POL_INV (0x1 << 2) +#define DA7218_DAI_WCLK_POL_SHIFT 3 +#define DA7218_DAI_WCLK_POL_MASK (0x1 << 3) +#define DA7218_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7218_DAI_WCLK_TRI_STATE_SHIFT 4 +#define DA7218_DAI_WCLK_TRI_STATE_MASK (0x1 << 4) +#define DA7218_DAI_CLK_EN_SHIFT 7 +#define DA7218_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7218_PLL_CTRL = 0x91 */ +#define DA7218_PLL_INDIV_SHIFT 0 +#define DA7218_PLL_INDIV_MASK (0x7 << 0) +#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0) +#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0) +#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0) +#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0) +#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0) +#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2 +#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4 +#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8 +#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7218_PLL_MCLK_SQR_EN_SHIFT 4 +#define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4) +#define DA7218_PLL_MODE_SHIFT 6 +#define DA7218_PLL_MODE_MASK (0x3 << 6) +#define DA7218_PLL_MODE_BYPASS (0x0 << 6) +#define DA7218_PLL_MODE_NORMAL (0x1 << 6) +#define DA7218_PLL_MODE_SRM (0x2 << 6) +#define DA7218_PLL_MODE_32KHZ (0x3 << 6) + +/* DA7218_PLL_FRAC_TOP = 0x92 */ +#define DA7218_PLL_FBDIV_FRAC_TOP_SHIFT 0 +#define DA7218_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0) + +/* DA7218_PLL_FRAC_BOT = 0x93 */ +#define DA7218_PLL_FBDIV_FRAC_BOT_SHIFT 0 +#define DA7218_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0) + +/* DA7218_PLL_INTEGER = 0x94 */ +#define DA7218_PLL_FBDIV_INTEGER_SHIFT 0 +#define DA7218_PLL_FBDIV_INTEGER_MASK (0x7F << 0) + +/* DA7218_PLL_STATUS = 0x95 */ +#define DA7218_PLL_SRM_STATUS_SHIFT 0 +#define DA7218_PLL_SRM_STATUS_MASK (0xFF << 0) +#define DA7218_PLL_SRM_STATUS_SRM_LOCK (0x1 << 7) + +/* DA7218_PLL_REFOSC_CAL = 0x98 */ +#define DA7218_PLL_REFOSC_CAL_CTRL_SHIFT 0 +#define DA7218_PLL_REFOSC_CAL_CTRL_MASK (0x1F << 0) +#define DA7218_PLL_REFOSC_CAL_START_SHIFT 6 +#define DA7218_PLL_REFOSC_CAL_START_MASK (0x1 << 6) +#define DA7218_PLL_REFOSC_CAL_EN_SHIFT 7 +#define DA7218_PLL_REFOSC_CAL_EN_MASK (0x1 << 7) + +/* DA7218_DAC_NG_CTRL = 0x9C */ +#define DA7218_DAC_NG_EN_SHIFT 7 +#define DA7218_DAC_NG_EN_MASK (0x1 << 7) + +/* DA7218_DAC_NG_SETUP_TIME = 0x9D */ +#define DA7218_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7218_DAC_NG_SETUP_TIME_MASK (0x3 << 0) +#define DA7218_DAC_NG_SETUP_TIME_MAX 4 +#define DA7218_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7218_DAC_NG_RAMPUP_RATE_MASK (0x1 << 2) +#define DA7218_DAC_NG_RAMPUP_RATE_MAX 2 +#define DA7218_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7218_DAC_NG_RAMPDN_RATE_MASK (0x1 << 3) +#define DA7218_DAC_NG_RAMPDN_RATE_MAX 2 + +/* DA7218_DAC_NG_OFF_THRESH = 0x9E */ +#define DA7218_DAC_NG_OFF_THRESHOLD_SHIFT 0 +#define DA7218_DAC_NG_OFF_THRESHOLD_MASK (0x7 << 0) +#define DA7218_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7218_DAC_NG_ON_THRESH = 0x9F */ +#define DA7218_DAC_NG_ON_THRESHOLD_SHIFT 0 +#define DA7218_DAC_NG_ON_THRESHOLD_MASK (0x7 << 0) + +/* DA7218_TONE_GEN_CFG1 = 0xA0 */ +#define DA7218_DTMF_REG_SHIFT 0 +#define DA7218_DTMF_REG_MASK (0xF << 0) +#define DA7218_DTMF_REG_MAX 16 +#define DA7218_DTMF_EN_SHIFT 4 +#define DA7218_DTMF_EN_MASK (0x1 << 4) +#define DA7218_START_STOPN_SHIFT 7 +#define DA7218_START_STOPN_MASK (0x1 << 7) + +/* DA7218_TONE_GEN_CFG2 = 0xA1 */ +#define DA7218_SWG_SEL_SHIFT 0 +#define DA7218_SWG_SEL_MASK (0x3 << 0) +#define DA7218_SWG_SEL_MAX 4 + +/* DA7218_TONE_GEN_FREQ1_L = 0xA2 */ +#define DA7218_FREQ1_L_SHIFT 0 +#define DA7218_FREQ1_L_MASK (0xFF << 0) +#define DA7218_FREQ_MAX 0xFFFF + +/* DA7218_TONE_GEN_FREQ1_U = 0xA3 */ +#define DA7218_FREQ1_U_SHIFT 0 +#define DA7218_FREQ1_U_MASK (0xFF << 0) + +/* DA7218_TONE_GEN_FREQ2_L = 0xA4 */ +#define DA7218_FREQ2_L_SHIFT 0 +#define DA7218_FREQ2_L_MASK (0xFF << 0) + +/* DA7218_TONE_GEN_FREQ2_U = 0xA5 */ +#define DA7218_FREQ2_U_SHIFT 0 +#define DA7218_FREQ2_U_MASK (0xFF << 0) + +/* DA7218_TONE_GEN_CYCLES = 0xA6 */ +#define DA7218_BEEP_CYCLES_SHIFT 0 +#define DA7218_BEEP_CYCLES_MASK (0x7 << 0) + +/* DA7218_TONE_GEN_ON_PER = 0xA7 */ +#define DA7218_BEEP_ON_PER_SHIFT 0 +#define DA7218_BEEP_ON_PER_MASK (0x3F << 0) + +/* DA7218_TONE_GEN_OFF_PER = 0xA8 */ +#define DA7218_BEEP_OFF_PER_SHIFT 0 +#define DA7218_BEEP_OFF_PER_MASK (0x3F << 0) +#define DA7218_BEEP_ON_OFF_MAX 0x3F + +/* DA7218_CP_CTRL = 0xAC */ +#define DA7218_CP_MOD_SHIFT 2 +#define DA7218_CP_MOD_MASK (0x3 << 2) +#define DA7218_CP_MCHANGE_SHIFT 4 +#define DA7218_CP_MCHANGE_MASK (0x3 << 4) +#define DA7218_CP_MCHANGE_REL_MASK 0x3 +#define DA7218_CP_MCHANGE_MAX 3 +#define DA7218_CP_MCHANGE_LARGEST_VOL 0x1 +#define DA7218_CP_MCHANGE_DAC_VOL 0x2 +#define DA7218_CP_MCHANGE_SIG_MAG 0x3 +#define DA7218_CP_SMALL_SWITCH_FREQ_EN_SHIFT 6 +#define DA7218_CP_SMALL_SWITCH_FREQ_EN_MASK (0x1 << 6) +#define DA7218_CP_EN_SHIFT 7 +#define DA7218_CP_EN_MASK (0x1 << 7) + +/* DA7218_CP_DELAY = 0xAD */ +#define DA7218_CP_FCONTROL_SHIFT 0 +#define DA7218_CP_FCONTROL_MASK (0x7 << 0) +#define DA7218_CP_FCONTROL_MAX 6 +#define DA7218_CP_TAU_DELAY_SHIFT 3 +#define DA7218_CP_TAU_DELAY_MASK (0x7 << 3) +#define DA7218_CP_TAU_DELAY_MAX 8 + +/* DA7218_CP_VOL_THRESHOLD1 = 0xAE */ +#define DA7218_CP_THRESH_VDD2_SHIFT 0 +#define DA7218_CP_THRESH_VDD2_MASK (0x3F << 0) +#define DA7218_CP_THRESH_VDD2_MAX 0x3F + +/* DA7218_MIC_1_CTRL = 0xB4 */ +#define DA7218_MIC_1_AMP_MUTE_EN_SHIFT 6 +#define DA7218_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7218_MIC_1_AMP_EN_SHIFT 7 +#define DA7218_MIC_1_AMP_EN_MASK (0x1 << 7) + +/* DA7218_MIC_1_GAIN = 0xB5 */ +#define DA7218_MIC_1_AMP_GAIN_SHIFT 0 +#define DA7218_MIC_1_AMP_GAIN_MASK (0x7 << 0) +#define DA7218_MIC_AMP_GAIN_MAX 0x7 + +/* DA7218_MIC_1_SELECT = 0xB7 */ +#define DA7218_MIC_1_AMP_IN_SEL_SHIFT 0 +#define DA7218_MIC_1_AMP_IN_SEL_MASK (0x3 << 0) + +/* DA7218_MIC_2_CTRL = 0xB8 */ +#define DA7218_MIC_2_AMP_MUTE_EN_SHIFT 6 +#define DA7218_MIC_2_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7218_MIC_2_AMP_EN_SHIFT 7 +#define DA7218_MIC_2_AMP_EN_MASK (0x1 << 7) + +/* DA7218_MIC_2_GAIN = 0xB9 */ +#define DA7218_MIC_2_AMP_GAIN_SHIFT 0 +#define DA7218_MIC_2_AMP_GAIN_MASK (0x7 << 0) + +/* DA7218_MIC_2_SELECT = 0xBB */ +#define DA7218_MIC_2_AMP_IN_SEL_SHIFT 0 +#define DA7218_MIC_2_AMP_IN_SEL_MASK (0x3 << 0) + +/* DA7218_IN_1_HPF_FILTER_CTRL = 0xBC */ +#define DA7218_IN_1_VOICE_HPF_CORNER_SHIFT 0 +#define DA7218_IN_1_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7218_IN_VOICE_HPF_CORNER_MAX 8 +#define DA7218_IN_1_VOICE_EN_SHIFT 3 +#define DA7218_IN_1_VOICE_EN_MASK (0x1 << 3) +#define DA7218_IN_1_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7218_IN_1_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7218_IN_1_HPF_EN_SHIFT 7 +#define DA7218_IN_1_HPF_EN_MASK (0x1 << 7) + +/* DA7218_IN_2_HPF_FILTER_CTRL = 0xBD */ +#define DA7218_IN_2_VOICE_HPF_CORNER_SHIFT 0 +#define DA7218_IN_2_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7218_IN_2_VOICE_EN_SHIFT 3 +#define DA7218_IN_2_VOICE_EN_MASK (0x1 << 3) +#define DA7218_IN_2_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7218_IN_2_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7218_IN_2_HPF_EN_SHIFT 7 +#define DA7218_IN_2_HPF_EN_MASK (0x1 << 7) + +/* DA7218_ADC_1_CTRL = 0xC0 */ +#define DA7218_ADC_1_AAF_EN_SHIFT 2 +#define DA7218_ADC_1_AAF_EN_MASK (0x1 << 2) + +/* DA7218_ADC_2_CTRL = 0xC1 */ +#define DA7218_ADC_2_AAF_EN_SHIFT 2 +#define DA7218_ADC_2_AAF_EN_MASK (0x1 << 2) + +/* DA7218_ADC_MODE = 0xC2 */ +#define DA7218_ADC_LP_MODE_SHIFT 0 +#define DA7218_ADC_LP_MODE_MASK (0x1 << 0) +#define DA7218_ADC_LVLDET_MODE_SHIFT 1 +#define DA7218_ADC_LVLDET_MODE_MASK (0x1 << 1) +#define DA7218_ADC_LVLDET_AUTO_EXIT_SHIFT 2 +#define DA7218_ADC_LVLDET_AUTO_EXIT_MASK (0x1 << 2) + +/* DA7218_MIXOUT_L_CTRL = 0xCC */ +#define DA7218_MIXOUT_L_AMP_EN_SHIFT 7 +#define DA7218_MIXOUT_L_AMP_EN_MASK (0x1 << 7) + +/* DA7218_MIXOUT_L_GAIN = 0xCD */ +#define DA7218_MIXOUT_L_AMP_GAIN_SHIFT 0 +#define DA7218_MIXOUT_L_AMP_GAIN_MASK (0x3 << 0) +#define DA7218_MIXOUT_AMP_GAIN_MIN 0x1 +#define DA7218_MIXOUT_AMP_GAIN_MAX 0x3 + +/* DA7218_MIXOUT_R_CTRL = 0xCE */ +#define DA7218_MIXOUT_R_AMP_EN_SHIFT 7 +#define DA7218_MIXOUT_R_AMP_EN_MASK (0x1 << 7) + +/* DA7218_MIXOUT_R_GAIN = 0xCF */ +#define DA7218_MIXOUT_R_AMP_GAIN_SHIFT 0 +#define DA7218_MIXOUT_R_AMP_GAIN_MASK (0x3 << 0) + +/* DA7218_HP_L_CTRL = 0xD0 */ +#define DA7218_HP_L_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7218_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7218_HP_L_AMP_OE_SHIFT 3 +#define DA7218_HP_L_AMP_OE_MASK (0x1 << 3) +#define DA7218_HP_L_AMP_ZC_EN_SHIFT 4 +#define DA7218_HP_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7218_HP_L_AMP_RAMP_EN_SHIFT 5 +#define DA7218_HP_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7218_HP_L_AMP_MUTE_EN_SHIFT 6 +#define DA7218_HP_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7218_HP_L_AMP_EN_SHIFT 7 +#define DA7218_HP_L_AMP_EN_MASK (0x1 << 7) +#define DA7218_HP_AMP_OE_MASK (0x1 << 3) + +/* DA7218_HP_L_GAIN = 0xD1 */ +#define DA7218_HP_L_AMP_GAIN_SHIFT 0 +#define DA7218_HP_L_AMP_GAIN_MASK (0x3F << 0) +#define DA7218_HP_AMP_GAIN_MIN 0x15 +#define DA7218_HP_AMP_GAIN_MAX 0x3F + +/* DA7218_HP_R_CTRL = 0xD2 */ +#define DA7218_HP_R_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7218_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7218_HP_R_AMP_OE_SHIFT 3 +#define DA7218_HP_R_AMP_OE_MASK (0x1 << 3) +#define DA7218_HP_R_AMP_ZC_EN_SHIFT 4 +#define DA7218_HP_R_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7218_HP_R_AMP_RAMP_EN_SHIFT 5 +#define DA7218_HP_R_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7218_HP_R_AMP_MUTE_EN_SHIFT 6 +#define DA7218_HP_R_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7218_HP_R_AMP_EN_SHIFT 7 +#define DA7218_HP_R_AMP_EN_MASK (0x1 << 7) + +/* DA7218_HP_R_GAIN = 0xD3 */ +#define DA7218_HP_R_AMP_GAIN_SHIFT 0 +#define DA7218_HP_R_AMP_GAIN_MASK (0x3F << 0) + +/* DA7218_HP_SNGL_CTRL = 0xD4 */ +#define DA7218_HP_AMP_STEREO_DETECT_STATUS_SHIFT 0 +#define DA7218_HP_AMP_STEREO_DETECT_STATUS_MASK (0x1 << 0) +#define DA7218_HPL_AMP_LOAD_DETECT_STATUS_SHIFT 1 +#define DA7218_HPL_AMP_LOAD_DETECT_STATUS_MASK (0x1 << 1) +#define DA7218_HPR_AMP_LOAD_DETECT_STATUS_SHIFT 2 +#define DA7218_HPR_AMP_LOAD_DETECT_STATUS_MASK (0x1 << 2) +#define DA7218_HP_AMP_LOAD_DETECT_EN_SHIFT 6 +#define DA7218_HP_AMP_LOAD_DETECT_EN_MASK (0x1 << 6) +#define DA7218_HP_AMP_STEREO_DETECT_EN_SHIFT 7 +#define DA7218_HP_AMP_STEREO_DETECT_EN_MASK (0x1 << 7) + +/* DA7218_HP_DIFF_CTRL = 0xD5 */ +#define DA7218_HP_AMP_DIFF_MODE_EN_SHIFT 0 +#define DA7218_HP_AMP_DIFF_MODE_EN_MASK (0x1 << 0) +#define DA7218_HP_AMP_SINGLE_SUPPLY_EN_SHIFT 4 +#define DA7218_HP_AMP_SINGLE_SUPPLY_EN_MASK (0x1 << 4) + +/* DA7218_HP_DIFF_UNLOCK = 0xD7 */ +#define DA7218_HP_DIFF_UNLOCK_SHIFT 0 +#define DA7218_HP_DIFF_UNLOCK_MASK (0x1 << 0) +#define DA7218_HP_DIFF_UNLOCK_VAL 0xC3 + +/* DA7218_HPLDET_JACK = 0xD8 */ +#define DA7218_HPLDET_JACK_RATE_SHIFT 0 +#define DA7218_HPLDET_JACK_RATE_MASK (0x7 << 0) +#define DA7218_HPLDET_JACK_DEBOUNCE_SHIFT 3 +#define DA7218_HPLDET_JACK_DEBOUNCE_MASK (0x3 << 3) +#define DA7218_HPLDET_JACK_THR_SHIFT 5 +#define DA7218_HPLDET_JACK_THR_MASK (0x3 << 5) +#define DA7218_HPLDET_JACK_EN_SHIFT 7 +#define DA7218_HPLDET_JACK_EN_MASK (0x1 << 7) + +/* DA7218_HPLDET_CTRL = 0xD9 */ +#define DA7218_HPLDET_COMP_INV_SHIFT 0 +#define DA7218_HPLDET_COMP_INV_MASK (0x1 << 0) +#define DA7218_HPLDET_HYST_EN_SHIFT 1 +#define DA7218_HPLDET_HYST_EN_MASK (0x1 << 1) +#define DA7218_HPLDET_DISCHARGE_EN_SHIFT 7 +#define DA7218_HPLDET_DISCHARGE_EN_MASK (0x1 << 7) + +/* DA7218_HPLDET_TEST = 0xDA */ +#define DA7218_HPLDET_COMP_STS_SHIFT 4 +#define DA7218_HPLDET_COMP_STS_MASK (0x1 << 4) + +/* DA7218_REFERENCES = 0xDC */ +#define DA7218_BIAS_EN_SHIFT 3 +#define DA7218_BIAS_EN_MASK (0x1 << 3) + +/* DA7218_IO_CTRL = 0xE0 */ +#define DA7218_IO_VOLTAGE_LEVEL_SHIFT 0 +#define DA7218_IO_VOLTAGE_LEVEL_MASK (0x1 << 0) +#define DA7218_IO_VOLTAGE_LEVEL_2_5V_3_6V 0 +#define DA7218_IO_VOLTAGE_LEVEL_1_5V_2_5V 1 + +/* DA7218_LDO_CTRL = 0xE1 */ +#define DA7218_LDO_LEVEL_SELECT_SHIFT 4 +#define DA7218_LDO_LEVEL_SELECT_MASK (0x3 << 4) +#define DA7218_LDO_EN_SHIFT 7 +#define DA7218_LDO_EN_MASK (0x1 << 7) + +/* DA7218_SIDETONE_CTRL = 0xE4 */ +#define DA7218_SIDETONE_MUTE_EN_SHIFT 6 +#define DA7218_SIDETONE_MUTE_EN_MASK (0x1 << 6) +#define DA7218_SIDETONE_FILTER_EN_SHIFT 7 +#define DA7218_SIDETONE_FILTER_EN_MASK (0x1 << 7) + +/* DA7218_SIDETONE_IN_SELECT = 0xE5 */ +#define DA7218_SIDETONE_IN_SELECT_SHIFT 0 +#define DA7218_SIDETONE_IN_SELECT_MASK (0x3 << 0) +#define DA7218_SIDETONE_IN_SELECT_MAX 4 + +/* DA7218_SIDETONE_GAIN = 0xE6 */ +#define DA7218_SIDETONE_GAIN_SHIFT 0 +#define DA7218_SIDETONE_GAIN_MASK (0x1F << 0) + +/* DA7218_DROUTING_ST_OUTFILT_1L = 0xE8 */ +#define DA7218_OUTFILT_ST_1L_SRC_SHIFT 0 +#define DA7218_OUTFILT_ST_1L_SRC_MASK (0x7 << 0) +#define DA7218_DMIX_ST_SRC_OUTFILT1L 0 +#define DA7218_DMIX_ST_SRC_OUTFILT1R 1 +#define DA7218_DMIX_ST_SRC_SIDETONE 2 + +/* DA7218_DROUTING_ST_OUTFILT_1R = 0xE9 */ +#define DA7218_OUTFILT_ST_1R_SRC_SHIFT 0 +#define DA7218_OUTFILT_ST_1R_SRC_MASK (0x7 << 0) + +/* DA7218_SIDETONE_BIQ_3STAGE_DATA = 0xEA */ +#define DA7218_SIDETONE_BIQ_3STAGE_DATA_SHIFT 0 +#define DA7218_SIDETONE_BIQ_3STAGE_DATA_MASK (0xFF << 0) + +/* DA7218_SIDETONE_BIQ_3STAGE_ADDR = 0xEB */ +#define DA7218_SIDETONE_BIQ_3STAGE_ADDR_SHIFT 0 +#define DA7218_SIDETONE_BIQ_3STAGE_ADDR_MASK (0x1F << 0) +#define DA7218_SIDETONE_BIQ_3STAGE_CFG_SIZE 30 + +/* DA7218_EVENT_STATUS = 0xEC */ +#define DA7218_HPLDET_JACK_STS_SHIFT 7 +#define DA7218_HPLDET_JACK_STS_MASK (0x1 << 7) + +/* DA7218_EVENT = 0xED */ +#define DA7218_LVL_DET_EVENT_SHIFT 0 +#define DA7218_LVL_DET_EVENT_MASK (0x1 << 0) +#define DA7218_HPLDET_JACK_EVENT_SHIFT 7 +#define DA7218_HPLDET_JACK_EVENT_MASK (0x1 << 7) + +/* DA7218_EVENT_MASK = 0xEE */ +#define DA7218_LVL_DET_EVENT_MSK_SHIFT 0 +#define DA7218_LVL_DET_EVENT_MSK_MASK (0x1 << 0) +#define DA7218_HPLDET_JACK_EVENT_IRQ_MSK_SHIFT 7 +#define DA7218_HPLDET_JACK_EVENT_IRQ_MSK_MASK (0x1 << 7) + +/* DA7218_DMIC_1_CTRL = 0xF0 */ +#define DA7218_DMIC_1_DATA_SEL_SHIFT 0 +#define DA7218_DMIC_1_DATA_SEL_MASK (0x1 << 0) +#define DA7218_DMIC_1_SAMPLEPHASE_SHIFT 1 +#define DA7218_DMIC_1_SAMPLEPHASE_MASK (0x1 << 1) +#define DA7218_DMIC_1_CLK_RATE_SHIFT 2 +#define DA7218_DMIC_1_CLK_RATE_MASK (0x1 << 2) +#define DA7218_DMIC_1L_EN_SHIFT 6 +#define DA7218_DMIC_1L_EN_MASK (0x1 << 6) +#define DA7218_DMIC_1R_EN_SHIFT 7 +#define DA7218_DMIC_1R_EN_MASK (0x1 << 7) + +/* DA7218_DMIC_2_CTRL = 0xF1 */ +#define DA7218_DMIC_2_DATA_SEL_SHIFT 0 +#define DA7218_DMIC_2_DATA_SEL_MASK (0x1 << 0) +#define DA7218_DMIC_2_SAMPLEPHASE_SHIFT 1 +#define DA7218_DMIC_2_SAMPLEPHASE_MASK (0x1 << 1) +#define DA7218_DMIC_2_CLK_RATE_SHIFT 2 +#define DA7218_DMIC_2_CLK_RATE_MASK (0x1 << 2) +#define DA7218_DMIC_2L_EN_SHIFT 6 +#define DA7218_DMIC_2L_EN_MASK (0x1 << 6) +#define DA7218_DMIC_2R_EN_SHIFT 7 +#define DA7218_DMIC_2R_EN_MASK (0x1 << 7) + +/* DA7218_IN_1L_GAIN = 0xF4 */ +#define DA7218_IN_1L_DIGITAL_GAIN_SHIFT 0 +#define DA7218_IN_1L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7218_IN_DIGITAL_GAIN_MAX 0x7F + +/* DA7218_IN_1R_GAIN = 0xF5 */ +#define DA7218_IN_1R_DIGITAL_GAIN_SHIFT 0 +#define DA7218_IN_1R_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7218_IN_2L_GAIN = 0xF6 */ +#define DA7218_IN_2L_DIGITAL_GAIN_SHIFT 0 +#define DA7218_IN_2L_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7218_IN_2R_GAIN = 0xF7 */ +#define DA7218_IN_2R_DIGITAL_GAIN_SHIFT 0 +#define DA7218_IN_2R_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7218_OUT_1L_GAIN = 0xF8 */ +#define DA7218_OUT_1L_DIGITAL_GAIN_SHIFT 0 +#define DA7218_OUT_1L_DIGITAL_GAIN_MASK (0xFF << 0) +#define DA7218_OUT_DIGITAL_GAIN_MIN 0x0 +#define DA7218_OUT_DIGITAL_GAIN_MAX 0x97 + +/* DA7218_OUT_1R_GAIN = 0xF9 */ +#define DA7218_OUT_1R_DIGITAL_GAIN_SHIFT 0 +#define DA7218_OUT_1R_DIGITAL_GAIN_MASK (0xFF << 0) + +/* DA7218_MICBIAS_CTRL = 0xFC */ +#define DA7218_MICBIAS_1_LEVEL_SHIFT 0 +#define DA7218_MICBIAS_1_LEVEL_MASK (0x7 << 0) +#define DA7218_MICBIAS_1_LP_MODE_SHIFT 3 +#define DA7218_MICBIAS_1_LP_MODE_MASK (0x1 << 3) +#define DA7218_MICBIAS_2_LEVEL_SHIFT 4 +#define DA7218_MICBIAS_2_LEVEL_MASK (0x7 << 4) +#define DA7218_MICBIAS_2_LP_MODE_SHIFT 7 +#define DA7218_MICBIAS_2_LP_MODE_MASK (0x1 << 7) + +/* DA7218_MICBIAS_EN = 0xFD */ +#define DA7218_MICBIAS_1_EN_SHIFT 0 +#define DA7218_MICBIAS_1_EN_MASK (0x1 << 0) +#define DA7218_MICBIAS_2_EN_SHIFT 4 +#define DA7218_MICBIAS_2_EN_MASK (0x1 << 4) + + +/* + * General defines & data + */ + +/* Register inversion */ +#define DA7218_NO_INVERT 0 +#define DA7218_INVERT 1 + +/* Byte related defines */ +#define DA7218_BYTE_SHIFT 8 +#define DA7218_BYTE_MASK 0xFF +#define DA7218_2BYTE_SHIFT 16 +#define DA7218_2BYTE_MASK 0xFFFF + +/* PLL Output Frequencies */ +#define DA7218_PLL_FREQ_OUT_90316 90316800 +#define DA7218_PLL_FREQ_OUT_98304 98304000 + +/* ALC Calibration */ +#define DA7218_ALC_CALIB_DELAY_MIN 2500 +#define DA7218_ALC_CALIB_DELAY_MAX 5000 +#define DA7218_ALC_CALIB_MAX_TRIES 5 + +/* Ref Oscillator */ +#define DA7218_REF_OSC_CHECK_DELAY_MIN 5000 +#define DA7218_REF_OSC_CHECK_DELAY_MAX 10000 +#define DA7218_REF_OSC_CHECK_TRIES 4 + +/* SRM */ +#define DA7218_SRM_CHECK_DELAY 50 +#define DA7218_SRM_CHECK_TRIES 8 + +/* Mic Level Detect */ +#define DA7218_MIC_LVL_DET_DELAY 50 + +enum da7218_biq_cfg { + DA7218_BIQ_CFG_DATA = 0, + DA7218_BIQ_CFG_ADDR, + DA7218_BIQ_CFG_SIZE, +}; + +enum da7218_clk_src { + DA7218_CLKSRC_MCLK = 0, + DA7218_CLKSRC_MCLK_SQR, +}; + +enum da7218_sys_clk { + DA7218_SYSCLK_MCLK = 0, + DA7218_SYSCLK_PLL, + DA7218_SYSCLK_PLL_SRM, + DA7218_SYSCLK_PLL_32KHZ +}; + +enum da7218_dev_id { + DA7217_DEV_ID = 0, + DA7218_DEV_ID, +}; + +/* Regulators */ +enum da7218_supplies { + DA7218_SUPPLY_VDD = 0, + DA7218_SUPPLY_VDDMIC, + DA7218_SUPPLY_VDDIO, + DA7218_NUM_SUPPLIES, +}; + +/* Private data */ +struct da7218_priv { + struct da7218_pdata *pdata; + + struct regulator_bulk_data supplies[DA7218_NUM_SUPPLIES]; + struct regmap *regmap; + int dev_id; + + struct snd_soc_jack *jack; + int irq; + + struct clk *mclk; + unsigned int mclk_rate; + + bool hp_single_supply; + bool master; + u8 alc_en; + u8 in_filt_en; + u8 mic_lvl_det_en; + + u8 biq_5stage_coeff[DA7218_OUT_1_BIQ_5STAGE_CFG_SIZE]; + u8 stbiq_3stage_coeff[DA7218_SIDETONE_BIQ_3STAGE_CFG_SIZE]; +}; + +/* HP detect control */ +int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* _DA7218_H */ -- 2.6.2

1 0

[alsa-devel] Applied "ASoC: rsnd: Add device tree support for r8a77{79, 93, 94}" to the asoc tree
by Mark Brown 30 Nov '15

30 Nov '15

The patch ASoC: rsnd: Add device tree support for r8a77{79,93,94} 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 4eb404d00f58bbb2d116fdd2563dd45ff0cbdef1 Mon Sep 17 00:00:00 2001 From: Simon Horman <horms+renesas(a)verge.net.au> Date: Mon, 30 Nov 2015 15:02:52 +0900 Subject: [PATCH] ASoC: rsnd: Add device tree support for r8a77{79,93,94} Simply document new compat strings. There appears to be no need for a driver updates. Signed-off-by: Simon Horman <horms+renesas(a)verge.net.au> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index bbcb3272977d..59cebfbd7b6a 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -7,8 +7,11 @@ Required properties: "renesas,rcar_sound-gen3" if generation3 Examples with soctypes are: - "renesas,rcar_sound-r8a7778" (R-Car M1A) + - "renesas,rcar_sound-r8a7779" (R-Car H1) - "renesas,rcar_sound-r8a7790" (R-Car H2) - "renesas,rcar_sound-r8a7791" (R-Car M2-W) + - "renesas,rcar_sound-r8a7793" (R-Car M2-N) + - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - reg : Should contain the register physical address. required register is -- 2.6.2

1 0

[alsa-devel] System with multiple arizona (wm5102) codecs
by Pavel Machek 30 Nov '15

30 Nov '15

Hi! I've got an embedded system with two arizona / wm5102 codecs. Unfortunately, kernel does not seem to be ready for that configuration. In particular, drivers/regulator/arizona-ldo1.c and drivers/regulator/arizona-micsupp.c register system-wide "MICVDD" and "LDO1" regulators, but with two codecs in the system, we really have wm5102-codec.1.MICVDD, wm5102-codec.2.MICVDD, wm5102-codec.1.LDO1 and wm5102-codec.2.LDO1. That got me second codec working in two-codec configuration, but first one still stops working as soon as two codecs are enabled. If you have idea what else needs fixing, let me know. Best regards, Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

6 43

[alsa-devel] [PATCH v3 0/2] ALSA: compress: Add procfs info file for compressed nodes
by Richard Fitzgerald 30 Nov '15

30 Nov '15

This updates the compress core code to create an 'info' file under procfs for each compressed node, like the PCM core does for PCM nodes. Based off Takashi's for-next Richard Fitzgerald (2): ALSA: compress: Add procfs info file for compressed nodes ALSA: compress: Pass id string to snd_compress_new include/sound/compress_driver.h | 7 +++- sound/core/compress_offload.c | 86 ++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-compress.c | 8 +++- 3 files changed, 97 insertions(+), 4 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH] ALSA: i2c: constify snd_i2c_ops structures
by Julia Lawall 30 Nov '15

30 Nov '15

The snd_i2c_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- include/sound/i2c.h | 2 +- sound/i2c/i2c.c | 2 +- sound/pci/ice1712/delta.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/i2c.h b/include/sound/i2c.h index d125ff8..835254d 100644 --- a/include/sound/i2c.h +++ b/include/sound/i2c.h @@ -66,7 +66,7 @@ struct snd_i2c_bus { struct snd_i2c_bit_ops *bit; void *ops; } hw_ops; /* lowlevel operations */ - struct snd_i2c_ops *ops; /* midlevel operations */ + const struct snd_i2c_ops *ops; /* midlevel operations */ unsigned long private_value; void *private_data; diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index 4677037..ef2a9af 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -39,7 +39,7 @@ static int snd_i2c_bit_readbytes(struct snd_i2c_device *device, static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr); -static struct snd_i2c_ops snd_i2c_bit_ops = { +static const struct snd_i2c_ops snd_i2c_bit_ops = { .sendbytes = snd_i2c_bit_sendbytes, .readbytes = snd_i2c_bit_readbytes, .probeaddr = snd_i2c_bit_probeaddr, diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 496dbd0..3bfdc78 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -174,7 +174,7 @@ static int ap_cs8427_probeaddr(struct snd_i2c_bus *bus, unsigned short addr) return -ENOENT; } -static struct snd_i2c_ops ap_cs8427_i2c_ops = { +static const struct snd_i2c_ops ap_cs8427_i2c_ops = { .sendbytes = ap_cs8427_sendbytes, .readbytes = ap_cs8427_readbytes, .probeaddr = ap_cs8427_probeaddr,

2 1

[alsa-devel] [PATCH] ALSA: pcm: constify action_ops structures
by Julia Lawall 30 Nov '15

30 Nov '15

The action_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/core/pcm_native.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a8b27cd..fadd3eb 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -875,7 +875,7 @@ struct action_ops { * Note: the stream state might be changed also on failure * Note2: call with calling stream lock + link lock */ -static int snd_pcm_action_group(struct action_ops *ops, +static int snd_pcm_action_group(const struct action_ops *ops, struct snd_pcm_substream *substream, int state, int do_lock) { @@ -932,7 +932,7 @@ static int snd_pcm_action_group(struct action_ops *ops, /* * Note: call with stream lock */ -static int snd_pcm_action_single(struct action_ops *ops, +static int snd_pcm_action_single(const struct action_ops *ops, struct snd_pcm_substream *substream, int state) { @@ -952,7 +952,7 @@ static int snd_pcm_action_single(struct action_ops *ops, /* * Note: call with stream lock */ -static int snd_pcm_action(struct action_ops *ops, +static int snd_pcm_action(const struct action_ops *ops, struct snd_pcm_substream *substream, int state) { @@ -984,7 +984,7 @@ static int snd_pcm_action(struct action_ops *ops, /* * Note: don't use any locks before */ -static int snd_pcm_action_lock_irq(struct action_ops *ops, +static int snd_pcm_action_lock_irq(const struct action_ops *ops, struct snd_pcm_substream *substream, int state) { @@ -998,7 +998,7 @@ static int snd_pcm_action_lock_irq(struct action_ops *ops, /* */ -static int snd_pcm_action_nonatomic(struct action_ops *ops, +static int snd_pcm_action_nonatomic(const struct action_ops *ops, struct snd_pcm_substream *substream, int state) { @@ -1056,7 +1056,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } -static struct action_ops snd_pcm_action_start = { +static const struct action_ops snd_pcm_action_start = { .pre_action = snd_pcm_pre_start, .do_action = snd_pcm_do_start, .undo_action = snd_pcm_undo_start, @@ -1107,7 +1107,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) wake_up(&runtime->tsleep); } -static struct action_ops snd_pcm_action_stop = { +static const struct action_ops snd_pcm_action_stop = { .pre_action = snd_pcm_pre_stop, .do_action = snd_pcm_do_stop, .post_action = snd_pcm_post_stop @@ -1224,7 +1224,7 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) } } -static struct action_ops snd_pcm_action_pause = { +static const struct action_ops snd_pcm_action_pause = { .pre_action = snd_pcm_pre_pause, .do_action = snd_pcm_do_pause, .undo_action = snd_pcm_undo_pause, @@ -1273,7 +1273,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) wake_up(&runtime->tsleep); } -static struct action_ops snd_pcm_action_suspend = { +static const struct action_ops snd_pcm_action_suspend = { .pre_action = snd_pcm_pre_suspend, .do_action = snd_pcm_do_suspend, .post_action = snd_pcm_post_suspend @@ -1375,7 +1375,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } -static struct action_ops snd_pcm_action_resume = { +static const struct action_ops snd_pcm_action_resume = { .pre_action = snd_pcm_pre_resume, .do_action = snd_pcm_do_resume, .undo_action = snd_pcm_undo_resume, @@ -1478,7 +1478,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, int state) snd_pcm_playback_silence(substream, ULONG_MAX); } -static struct action_ops snd_pcm_action_reset = { +static const struct action_ops snd_pcm_action_reset = { .pre_action = snd_pcm_pre_reset, .do_action = snd_pcm_do_reset, .post_action = snd_pcm_post_reset @@ -1522,7 +1522,7 @@ static void snd_pcm_post_prepare(struct snd_pcm_substream *substream, int state) snd_pcm_set_state(substream, SNDRV_PCM_STATE_PREPARED); } -static struct action_ops snd_pcm_action_prepare = { +static const struct action_ops snd_pcm_action_prepare = { .pre_action = snd_pcm_pre_prepare, .do_action = snd_pcm_do_prepare, .post_action = snd_pcm_post_prepare @@ -1618,7 +1618,7 @@ static void snd_pcm_post_drain_init(struct snd_pcm_substream *substream, int sta { } -static struct action_ops snd_pcm_action_drain_init = { +static const struct action_ops snd_pcm_action_drain_init = { .pre_action = snd_pcm_pre_drain_init, .do_action = snd_pcm_do_drain_init, .post_action = snd_pcm_post_drain_init

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Add device tree support for r8a77{79, 93, 94}
by Simon Horman 30 Nov '15

30 Nov '15

Simply document new compat strings. There appears to be no need for a driver updates. Signed-off-by: Simon Horman <horms+renesas(a)verge.net.au> --- The r8a7792 is omitted from this change as according to the documentation the hardware does not appear to be present on that SoC. --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index bbcb3272977d..59cebfbd7b6a 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -7,8 +7,11 @@ Required properties: "renesas,rcar_sound-gen3" if generation3 Examples with soctypes are: - "renesas,rcar_sound-r8a7778" (R-Car M1A) + - "renesas,rcar_sound-r8a7779" (R-Car H1) - "renesas,rcar_sound-r8a7790" (R-Car H2) - "renesas,rcar_sound-r8a7791" (R-Car M2-W) + - "renesas,rcar_sound-r8a7793" (R-Car M2-N) + - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - reg : Should contain the register physical address. required register is -- 2.1.4

2 1

[alsa-devel] Question about TDM "slot"
by Kuninori Morimoto 30 Nov '15

30 Nov '15

Hi I would like to ask you about what does TDM "slot" mean on snd_soc_of_parse_tdm_slot() / snd_soc_dai_set_tdm_slot(). Is slot = 3 in this case ? | |--- 2ch --> | | | CPU |--- 2ch --> | Codec | | |--- 2ch --> | | total 6ch If this is correct, I have other question about slot_width. snd_soc_dai_set_tdm_slot comments say slot_width: Width in bits for each slot. Does this mean it will be 2ch x system word bit ? I know some codec chip have below style. Does it unexpected ? | |--- 8ch --> | | | CPU |--- 4ch --> | Codec | total 12ch Best regards --- Kuninori Morimoto

2 11

[alsa-devel] [PATCH 1/2] ASoC: fsl_sai: add tdm slots operation support
by Zidan Wang 29 Nov '15

29 Nov '15

Add tdm slots operation support. If tdm slots and slot width have been configured in machine driver, we should use these values. Otherwise, using relevant channels and word length to set slots and slot width. SAI will generate BCLK depends on sample rate, slots and slot width. And there may be unused BCLK cycles before each LRCLK transition. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 31 +++++++++++++++++++++++++------ sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4b042fd..c01322c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -126,6 +126,17 @@ out: return IRQ_HANDLED; } +static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + sai->slots = slots; + sai->slot_width = slot_width; + + return 0; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -395,11 +406,19 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + u32 slots = (channels == 1) ? 2 : channels; + u32 slot_width = word_width; int ret; + if (sai->slots) + slots = sai->slots; + + if (sai->slot_width) + slot_width = sai->slot_width; + if (!sai->is_slave_mode) { ret = fsl_sai_set_bclk(cpu_dai, tx, - 2 * word_width * params_rate(params)); + slots * slot_width * params_rate(params)); if (ret) return ret; @@ -411,21 +430,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, sai->mclk_streams |= BIT(substream->stream); } - } if (!sai->is_dsp_mode) - val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + val_cr4 |= FSL_SAI_CR4_SYWD(slot_width); - val_cr5 |= FSL_SAI_CR5_WNW(word_width); - val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 |= FSL_SAI_CR5_WNW(slot_width); + val_cr5 |= FSL_SAI_CR5_W0W(slot_width); if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); - val_cr4 |= FSL_SAI_CR4_FRSZ(channels); + val_cr4 |= FSL_SAI_CR4_FRSZ(slots); /* * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will @@ -592,6 +610,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, + .set_tdm_slot = fsl_sai_set_dai_tdm_slot, .hw_params = fsl_sai_hw_params, .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b95fbc3..d9ed7be 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -143,6 +143,9 @@ struct fsl_sai { unsigned int mclk_id[2]; unsigned int mclk_streams; + unsigned int slots; + unsigned int slot_width; + struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- 1.9.1

4 6

[alsa-devel] [PATCH 0/4 v5] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Nov '15

27 Nov '15

Hopefully last version? :-) * Added commit text about duplicate events (patch 4/4) * Added locks in bind/unbind on i915 side (patch 2/4) * Fixed docbook comments in i915 struct (patch 1/4) * Removed port_mst_streaming parameter (patch 1/4) * Added Reviewed-by - 1 & 2 by Jani, 3 & 4 by Takashi. Hopefully this was correct, otherwise feel free to adjust yourself before committing. Both Jani and Takashi seem okay with this going into 4.3. Nobody has said which tree you prefer to take it through, so how about Takashi merging it? David Henningsson (4): drm/i915: Add audio pin sense / ELD callback drm/i915: Call audio pin/ELD notify function ALSA: hda - allow codecs to access the i915 pin/ELD callback ALSA: hda - Wake the codec up on pin/ELD notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 27 ++++++++++++++++++++++++--- include/drm/i915_component.h | 16 ++++++++++++++++ include/sound/hda_i915.h | 7 +++++++ sound/hda/hdac_i915.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++++++- 6 files changed, 79 insertions(+), 4 deletions(-) -- 1.9.1

7 37

[alsa-devel] [PATCH] ASoC: arizona: Add 32uS delay after putting FLL into freerun
by Charles Keepax 27 Nov '15

27 Nov '15

When switching between two clock sources using the FLL freerun to smooth the transition we should wait 32uS after putting the FLL into freerun before we proceed. In practice we appear to be getting enough delay from the surrounding code, but better to make it explicit. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e76ecc7..8cd448c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2215,6 +2215,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + udelay(32); } /* -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: wm_adsp: Expand the list of available firmwares
by Charles Keepax 27 Nov '15

27 Nov '15

Expand the list of available firmware names to include a good selection of generic uses for the DSP cores. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 49 +++++++++++++++++++++++++++++++------------ 1 files changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0bb415a..905ae99 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -201,27 +201,48 @@ static void wm_adsp_buf_free(struct list_head *list) } } -#define WM_ADSP_NUM_FW 4 - -#define WM_ADSP_FW_MBC_VSS 0 -#define WM_ADSP_FW_TX 1 -#define WM_ADSP_FW_TX_SPK 2 -#define WM_ADSP_FW_RX_ANC 3 +#define WM_ADSP_FW_MBC_VSS 0 +#define WM_ADSP_FW_HIFI 1 +#define WM_ADSP_FW_TX 2 +#define WM_ADSP_FW_TX_SPK 3 +#define WM_ADSP_FW_RX 4 +#define WM_ADSP_FW_RX_ANC 5 +#define WM_ADSP_FW_CTRL 6 +#define WM_ADSP_FW_ASR 7 +#define WM_ADSP_FW_TRACE 8 +#define WM_ADSP_FW_SPK_PROT 9 +#define WM_ADSP_FW_MISC 10 + +#define WM_ADSP_NUM_FW 11 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { - [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", - [WM_ADSP_FW_TX] = "Tx", - [WM_ADSP_FW_TX_SPK] = "Tx Speaker", - [WM_ADSP_FW_RX_ANC] = "Rx ANC", + [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", + [WM_ADSP_FW_HIFI] = "MasterHiFi", + [WM_ADSP_FW_TX] = "Tx", + [WM_ADSP_FW_TX_SPK] = "Tx Speaker", + [WM_ADSP_FW_RX] = "Rx", + [WM_ADSP_FW_RX_ANC] = "Rx ANC", + [WM_ADSP_FW_CTRL] = "Voice Ctrl", + [WM_ADSP_FW_ASR] = "ASR Assist", + [WM_ADSP_FW_TRACE] = "Dbg Trace", + [WM_ADSP_FW_SPK_PROT] = "Protection", + [WM_ADSP_FW_MISC] = "Misc", }; static struct { const char *file; } wm_adsp_fw[WM_ADSP_NUM_FW] = { - [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, - [WM_ADSP_FW_TX] = { .file = "tx" }, - [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, - [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, + [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, + [WM_ADSP_FW_HIFI] = { .file = "hifi" }, + [WM_ADSP_FW_TX] = { .file = "tx" }, + [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, + [WM_ADSP_FW_RX] = { .file = "rx" }, + [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, + [WM_ADSP_FW_CTRL] = { .file = "ctrl" }, + [WM_ADSP_FW_ASR] = { .file = "asr" }, + [WM_ADSP_FW_TRACE] = { .file = "trace" }, + [WM_ADSP_FW_SPK_PROT] = { .file = "spk-prot" }, + [WM_ADSP_FW_MISC] = { .file = "misc" }, }; struct wm_coeff_ctl_ops { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: indicate register name for debug
by Kuninori Morimoto 27 Nov '15

27 Nov '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd driver is indicating how to use regmap debug method on gen.c comment area. regmap debug method indicates address and value, but rsnd driver is using too many IPs (SSI/SSIU/SRC/CTU/MIX/DVC/CMD), and address. Thus, we would like to know more useful information for debugging. This patch indicates address name for debugging. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/gen.c | 52 +++++++++++++++++++++++++++++-------------------- 1 file changed, 31 insertions(+), 21 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 84f8bb2..15d7706 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -31,29 +31,33 @@ struct rsnd_gen { /* RSND_REG_MAX base */ struct regmap_field *regs[RSND_REG_MAX]; + const char *reg_name[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) +#define rsnd_reg_name(gen, id) ((gen)->reg_name[id]) struct rsnd_regmap_field_conf { int idx; unsigned int reg_offset; unsigned int id_offset; + const char *reg_name; }; -#define RSND_REG_SET(id, offset, _id_offset) \ +#define RSND_REG_SET(id, offset, _id_offset, n) \ { \ .idx = id, \ .reg_offset = offset, \ .id_offset = _id_offset, \ + .reg_name = n, \ } /* single address mapping */ #define RSND_GEN_S_REG(id, offset) \ - RSND_REG_SET(RSND_REG_##id, offset, 0) + RSND_REG_SET(RSND_REG_##id, offset, 0, #id) /* multi address mapping */ #define RSND_GEN_M_REG(id, offset, _id_offset) \ - RSND_REG_SET(RSND_REG_##id, offset, _id_offset) + RSND_REG_SET(RSND_REG_##id, offset, _id_offset, #id) /* * basic function @@ -83,8 +87,9 @@ u32 rsnd_read(struct rsnd_priv *priv, regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); - dev_dbg(dev, "r %s[%d] - %4d : %08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val); + dev_dbg(dev, "r %s[%d] - %-18s (%4d) : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + rsnd_reg_name(gen, reg), reg, val); return val; } @@ -99,10 +104,11 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "w %s[%d] - %4d : %08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); - regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); + + dev_dbg(dev, "w %s[%d] - %-18s (%4d) : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + rsnd_reg_name(gen, reg), reg, data); } void rsnd_force_write(struct rsnd_priv *priv, @@ -115,10 +121,11 @@ void rsnd_force_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "w %s[%d] - %4d : %08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); - regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); + + dev_dbg(dev, "w %s[%d] - %-18s (%4d) : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + rsnd_reg_name(gen, reg), reg, data); } void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, @@ -130,11 +137,13 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "b %s[%d] - %4d : %08x/%08x\n", - rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data, mask); - regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), mask, data); + + dev_dbg(dev, "b %s[%d] - %-18s (%4d) : %08x/%08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + rsnd_reg_name(gen, reg), reg, data, mask); + } phys_addr_t rsnd_gen_get_phy_addr(struct rsnd_priv *priv, int reg_id) @@ -150,7 +159,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, int id_size, int reg_id, const char *name, - struct rsnd_regmap_field_conf *conf, + const struct rsnd_regmap_field_conf *conf, int conf_size) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); @@ -203,6 +212,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, /* RSND_REG_MAX base */ gen->regs[conf[i].idx] = regs; + gen->reg_name[conf[i].idx] = conf[i].reg_name; } return 0; @@ -213,7 +223,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, */ static int rsnd_gen2_probe(struct rsnd_priv *priv) { - struct rsnd_regmap_field_conf conf_ssiu[] = { + const static struct rsnd_regmap_field_conf conf_ssiu[] = { RSND_GEN_S_REG(SSI_MODE0, 0x800), RSND_GEN_S_REG(SSI_MODE1, 0x804), /* FIXME: it needs SSI_MODE2/3 in the future */ @@ -223,7 +233,7 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), }; - struct rsnd_regmap_field_conf conf_scu[] = { + const static struct rsnd_regmap_field_conf conf_scu[] = { RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x0, 0x20), RSND_GEN_M_REG(SRC_BUSIF_DALIGN,0x8, 0x20), RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), @@ -267,7 +277,7 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100), RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100), }; - struct rsnd_regmap_field_conf conf_adg[] = { + const static struct rsnd_regmap_field_conf conf_adg[] = { RSND_GEN_S_REG(BRRA, 0x00), RSND_GEN_S_REG(BRRB, 0x04), RSND_GEN_S_REG(SSICKR, 0x08), @@ -287,7 +297,7 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), }; - struct rsnd_regmap_field_conf conf_ssi[] = { + const static struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), RSND_GEN_M_REG(SSISR, 0x04, 0x40), RSND_GEN_M_REG(SSITDR, 0x08, 0x40), @@ -318,14 +328,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) static int rsnd_gen1_probe(struct rsnd_priv *priv) { - struct rsnd_regmap_field_conf conf_adg[] = { + const static struct rsnd_regmap_field_conf conf_adg[] = { RSND_GEN_S_REG(BRRA, 0x00), RSND_GEN_S_REG(BRRB, 0x04), RSND_GEN_S_REG(SSICKR, 0x08), RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), }; - struct rsnd_regmap_field_conf conf_ssi[] = { + const static struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), RSND_GEN_M_REG(SSISR, 0x04, 0x40), RSND_GEN_M_REG(SSITDR, 0x08, 0x40), -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup semantics of rsnd_src_record_error()
by Kuninori Morimoto 27 Nov '15

27 Nov '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_src_error_record() should recorde error, but it clears error too. this patch fixes up semantic of rsnd_src_error_record that it records error but doesn't clear error. And this patch renames rsnd_src_error_clear() to rsnd_src_status_clear() rsnd_src_error_record() to rsnd_src_record_error() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index c103aa7..6d93c4e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -300,7 +300,7 @@ static void rsnd_src_irq_ctrol(struct rsnd_mod *mod, int enable) rsnd_mod_bset(mod, SCU_SYS_INT_EN1, sys_int_mask, sys_int_val); } -static void rsnd_src_error_clear(struct rsnd_mod *mod) +static void rsnd_src_status_clear(struct rsnd_mod *mod) { u32 val = OUF_SRC(rsnd_mod_id(mod)); @@ -308,7 +308,7 @@ static void rsnd_src_error_clear(struct rsnd_mod *mod) rsnd_mod_bset(mod, SCU_SYS_STATUS1, val, val); } -static bool rsnd_src_error_record(struct rsnd_mod *mod) +static bool rsnd_src_record_error(struct rsnd_mod *mod) { struct rsnd_src *src = rsnd_mod_to_src(mod); u32 val0, val1; @@ -332,9 +332,6 @@ static bool rsnd_src_error_record(struct rsnd_mod *mod) ret = true; } - /* clear error static */ - rsnd_src_error_clear(mod); - return ret; } @@ -383,7 +380,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, rsnd_src_set_convert_rate(io, mod); - rsnd_src_error_clear(mod); + rsnd_src_status_clear(mod); rsnd_src_irq_enable(mod); @@ -434,7 +431,7 @@ static void __rsnd_src_interrupt(struct rsnd_mod *mod, if (!rsnd_io_is_working(io)) goto rsnd_src_interrupt_out; - if (rsnd_src_error_record(mod)) { + if (rsnd_src_record_error(mod)) { dev_dbg(dev, "%s[%d] restart\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); @@ -450,7 +447,9 @@ static void __rsnd_src_interrupt(struct rsnd_mod *mod, rsnd_mod_name(mod), rsnd_mod_id(mod)); } + rsnd_src_status_clear(mod); rsnd_src_interrupt_out: + spin_unlock(&priv->lock); } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup semantics of rsnd_src/ssi_record_error()
by Kuninori Morimoto 27 Nov '15

27 Nov '15

Hi Mark These patches fixup semantics of rsnd_src_record_error() / rsnd_ssi_record_error() to be more understandable code Kuninori Morimoto (2): ASoC: rsnd: tidyup semantics of rsnd_src_record_error() ASoC: rsnd: tidyup semantics of rsnd_ssi_record_error() sound/soc/sh/rcar/src.c | 12 +++++------- sound/soc/sh/rcar/ssi.c | 7 ++----- 2 files changed, 7 insertions(+), 12 deletions(-) Subject: [PATCH 0/13] ASoC: rsnd: remove SRC Gen1 and platform boot support Date: Tue, 10 Nov 2015 14:03:14 +0900

2 4

[alsa-devel] [GIT PULL] ASoC fixes for v4.4
by Mark Brown 27 Nov '15

27 Nov '15

The following changes since commit 1ec218373b8ebda821aec00bb156a9c94fad9cd4: Linux 4.4-rc2 (2015-11-22 16:45:59 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.4-rc2 for you to fetch changes up to 923f1cbf2e45f6292a01de959031a8efeabede78: Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/st', 'asoc/fix/sun4i-codec', 'asoc/fix/topology', 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus (2015-11-25 17:57:54 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.4 Quite a large batch of fixes have come in since the merge window, mainly driver specific ones but there's a couple of core ones: - A fix for DAPM resume on active streams to ensure everything ends up cleanly in the right state. - Reset the DAPM cache when freeing widgets to fix a crash on driver remove and reload. The PM functions for nau8825 are new code which fix crashes on resume. ---------------------------------------------------------------- Also the rt5677 changes look big but are very mechanical (making two simple changes repeatedly). Adam Sampson (1): ASoC: sun4i-codec: use consistent names for PA controls Andreas Dannenberg (1): ASoC: davinci-mcasp: Fix TDM slot rx/tx mask associations Arnd Bergmann (2): ASoC: fix rockchip 64-bit build warning ASoC: fsl: clarify ac97 dependency Bard Liao (4): ASoC: rl6231: avoid using divisible by 3 for DMIC clk ASoC: rt5670: fix wrong bit def for pll src ASoC: rt5645: Add dmi_system_id "Google Wizpig" ASoC: rt5645: Add dmi_system_id "Google Terra" Jeeja KP (1): ASoC: core: Change power state before rechecking endpoint John Keeping (1): ASoC: es8328: Fix shifts for mixer switches John Lin (3): ASoC: rl6231: fix range of DMIC clock ASoC: rt5645: Increase the delay time to imporve the HP pop noise ASoC: rt5645: Add struct dmi_system_id "Google Edgar" for Chrome OS JongHo Kim (1): ASoC: wm8960: Fix the Input PGA Mute switch Jyri Sarha (1): ASoC: dapm: Reset dapm wcache after freeing damp widgets Kuninori Morimoto (2): ASoC: rsnd: fixup SCU_SYS_INT_EN1 address ASoC: rsnd: make sure SRC In Rate feature enablement Mark Brown (7): Merge tag 'asoc-v4.3-rc6' into asoc-fix-rcar Merge remote-tracking branch 'asoc/fix/arizona' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl', 'asoc/fix/fsl-sai' and 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/nau8825', 'asoc/fix/ops', 'asoc/fix/rcar', 'asoc/fix/rl6231', 'asoc/fix/rockchip' and 'asoc/fix/rt5670' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/st', 'asoc/fix/sun4i-codec', 'asoc/fix/topology', 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus Moise Gergaud (3): ASoC: sti: remove wrong error message ASoC: sti: rename ST proprietary DT properties ASoC: sti: set player private data Oder Chiou (2): ASoC: rt5645: Power up the RC clock to make sure the speaker volume adjust properly ASoC: rt5677: Avoid the pop sound that comes from the filter power Omair M Abdullah (1): ASoC: topology: fix info callback for TLV byte control Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix master capture only mode Randy Dunlap (1): ASoC: fix kernel-doc warnings in sound/soc/soc-ops.c Richard Fitzgerald (1): ASoC: arizona: fix range of OPCLK_REF Sachin Pandhare (1): ASoC: wm8962: correct addresses for HPF_C_0/1 Sjoerd Simons (1): ASoC: rockchip: Fix incorrect VDW value for 24 bit Stefan Agner (1): ASoC: fsl_sai: fix Rx synchrounous mode Sudip Mukherjee (1): ASoC: Intel: Skylake: fix memory leak Vinod Koul (1): ASoC: Intel: Skylake: Add I2C depends for SKL machine Yong Zhi (1): ASoC: nau8825: add pm function include/sound/soc-dapm.h | 1 + sound/soc/codecs/arizona.c | 16 ++---- sound/soc/codecs/es8328.c | 16 +++--- sound/soc/codecs/nau8825.c | 31 ++++++++++ sound/soc/codecs/rl6231.c | 6 +- sound/soc/codecs/rt5645.c | 61 ++++++++++++++++++-- sound/soc/codecs/rt5670.h | 12 ++-- sound/soc/codecs/rt5677.c | 100 ++++++++++++++++++++------------- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8962.c | 4 +- sound/soc/davinci/davinci-mcasp.c | 12 +++- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/fsl_sai.c | 3 +- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/rockchip/rockchip_spdif.h | 6 +- sound/soc/sh/rcar/gen.c | 2 +- sound/soc/sh/rcar/src.c | 7 +++ sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 7 +++ sound/soc/soc-ops.c | 2 +- sound/soc/soc-topology.c | 3 +- sound/soc/sti/uniperif_player.c | 9 ++- sound/soc/sti/uniperif_reader.c | 3 +- sound/soc/sunxi/sun4i-codec.c | 27 ++++----- 26 files changed, 238 insertions(+), 105 deletions(-)

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: work around KEF X300A firmware bug
by Patrick Laughlin 27 Nov '15

27 Nov '15

Thank you for the help regarding this bug. I am not a programmer - could you tell me how to install this fix? Id there a newer version of the firmware you can send me? Thanks you! Patrick

2 1

[alsa-devel] [PATCH] ASoC: topology: Fix not to keep a reference to tplg fw
by Subhransu S. Prusty 27 Nov '15

27 Nov '15

During element creation, the name of some of the elements point to memory referenced in tplg fw. If the tplg fw is released after tplg is parsed by framework, kernel panic happens during creation of elements while card initialization. Issue is caught with id#87b5ed8ecb9fe05a696e1c0b53c7a49ea66432c1 So create a copy of the memory and assign to names instead. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-topology.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 6963ba2..61eb1de 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -709,7 +709,7 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, be->hdr.name, be->hdr.access); memset(&kc, 0, sizeof(kc)); - kc.name = be->hdr.name; + kc.name = kstrdup(be->hdr.name, GFP_KERNEL); kc.private_value = (long)sbe; kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc.access = be->hdr.access; @@ -789,7 +789,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, mc->hdr.name, mc->hdr.access); memset(&kc, 0, sizeof(kc)); - kc.name = mc->hdr.name; + kc.name = kstrdup(mc->hdr.name, GFP_KERNEL); kc.private_value = (long)sm; kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc.access = mc->hdr.access; @@ -935,7 +935,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, ec->hdr.name, ec->items); memset(&kc, 0, sizeof(kc)); - kc.name = ec->hdr.name; + kc.name = kstrdup(ec->hdr.name, GFP_KERNEL); kc.private_value = (long)se; kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc.access = ec->hdr.access; @@ -1105,8 +1105,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) return -EINVAL; - route.source = elem->source; - route.sink = elem->sink; + route.source = kstrdup(elem->source, GFP_KERNEL); + route.sink = kstrdup(elem->sink, GFP_KERNEL); route.connected = NULL; /* set to NULL atm for tplg users */ if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0) route.control = NULL; @@ -1149,7 +1149,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); - kc[i].name = mc->hdr.name; + kc[i].name = kstrdup(mc->hdr.name, GFP_KERNEL); kc[i].private_value = (long)sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = mc->hdr.access; @@ -1228,7 +1228,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); - kc->name = ec->hdr.name; + kc->name = kstrdup(ec->hdr.name, GFP_KERNEL); kc->private_value = (long)se; kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc->access = ec->hdr.access; @@ -1330,7 +1330,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - kc[i].name = be->hdr.name; + kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access; -- 1.9.1

4 16

Re: [alsa-devel] [PATCH] alsa-mixer: sb-omni-surround-5.1.conf: remove analog-surround-21, add Linux 4.3+ support
by Raymond Yau 27 Nov '15

27 Nov '15

>> >> > Do you mean 2.1 mode won't work with all usb audio since surround 5.1 of usb audio is already using route plugin >> What do you mean by "all usb audio"? In 4.1/5.1 mode LFE and all other outputs work properly, it is just for 2.1, which works as stereo. >> >> > If you don't have SPDIF output/input, you need to add your card in USB-AUDIO.conf >> Sound card has S/PDIF output, but not input. Not sure whether output works properly though. I do not have speakers or any other hardware with S/PDIF input to test it. >> >>> >>> > >>> > In 2.1 mode LFE is not actually working at all, so it is removed. >>> >>> Do you mean 2.1 mode won't work with all usb audio since surround 5.1 of usb audio is already using route plugin >>> >>> USB-Audio.pcm.surround51.0 { >>> @args [ CARD ] >>> @args.CARD { type string } >>> @func refer >>> name { >>> @func concat >>> strings [ >>> "cards.USB-Audio." >>> { @func card_name card $CARD } >>> ".pcm.surround51:CARD=" $CARD >>> ] >>> } >>> default { >>> type route >>> ttable.0.0 1 >>> ttable.1.1 1 >>> ttable.2.4 1 >>> ttable.3.5 1 >>> ttable.4.2 1 >>> ttable.5.3 1 >>> slave { >>> pcm { >>> type hw >>> card $CARD >>> device 0 >>> } >>> channels 6 >>> } >>> } >>> } >>> http://git.alsa-project.org/?p=alsa-lib.git;a=blob_plain;f=src/conf/pcm/sur… type route slave.pcm { @func refer name { @func concat strings [ "cards." { @func card_driver card $CARD } ".pcm.surround51." $DEV ":CARD=" $CARD ] } } ttable.0.FL 1 ttable.1.FR 1 ttable.2.LFE 1 hint { description "2.1 Surround output to Front and Subwoofer speakers" device $DEV } 2.1 seem use 5.1 and route plugin If you mean route plugin cannot be used twice, you need to remove surround 2.1 from USB-Audio.conf

1 0

[alsa-devel] Intel SST and HP ElitePad 1000 G2
by Jorge Fernandez Monteagudo 26 Nov '15

26 Nov '15

Hi, This is my first email in this list, and my last chance to make my tablet audio system work I'm trying to get my tablet audio system working but I'm run out of ideas. The HP ElitePad 1000 G2 it's an Atom Z3795 1.60GHz Bay Trail-T tablet. I'm using a kernel 4.2.5 with all the SST configs enabled: CONFIG_SND_SST_MFLD_PLATFORM=m CONFIG_SND_SST_IPC=m CONFIG_SND_SST_IPC_ACPI=m CONFIG_SND_SOC_INTEL_SST=m CONFIG_SND_SOC_INTEL_SST_ACPI=m CONFIG_SND_SOC_INTEL_HASWELL=m CONFIG_SND_SOC_INTEL_BAYTRAIL=m CONFIG_SND_SOC_INTEL_HASWELL_MACH=m CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH=m CONFIG_SND_SOC_INTEL_BROADWELL_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH=m CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH=m CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH=m # CONFIG_SND_SOC_QCOM is not set # CONFIG_SND_SOC_XTFPGA_I2S is not set CONFIG_SND_SOC_I2C_AND_SPI=m I've follow the same steps as the installation for Asus T100TA https://wiki.debian.org/InstallingDebianOn/Asus/T100TA and finally I only get the dmesg traces: sst-acpi 80860F28:00: No matching ASoC machine driver found intel-sst-acpi 80860F28:00: No matching machine driver found From the 'dsdt.dsl' I can see: Device (I2C2) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F41" /* Intel Baytrail I2C Host Controller */) // _HID: Hardware ID Name (_CID, "80860F41" /* Intel Baytrail I2C Host Controller */) // _CID: Compatible ID Name (_DEP, Package (0x01) // _DEP: Dependencies { PEPD }) Name (_DDN, "Intel(R) I2C Controller #2 - 80860F42") // _DDN: DOS Device Name Name (_UID, 0x02) // _UID: Unique ID Name (RBUF, ResourceTemplate () { Memory32Fixed (ReadWrite, 0x00000000, // Address Base 0x00001000, // Address Length _Y12) Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000021, } FixedDMA (0x0012, 0x0002, Width32bit, ) FixedDMA (0x0013, 0x0003, Width32bit, ) }) Method (_HRV, 0, NotSerialized) // _HRV: Hardware Revision { Return (SOCS) /* \SOCS */ } Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { CreateDWordField (RBUF, \_SB.I2C2._Y12._BAS, B0BA) // _BAS: Base Address CreateDWordField (RBUF, \_SB.I2C2._Y12._LEN, B0LN) // _LEN: Length Store (I20A, B0BA) /* \_SB_.I2C2._CRS.B0BA */ Store (I20L, B0LN) /* \_SB_.I2C2._CRS.B0LN */ Return (RBUF) /* \_SB_.I2C2.RBUF */ } Method (_STA, 0, NotSerialized) // _STA: Status { If (LOr (LEqual (I20A, Zero), LEqual (L22D, One))) { Return (Zero) } Return (0x0F) } Method (_PS3, 0, NotSerialized) // _PS3: Power State 3 { Or (PSAT, 0x03, PSAT) /* \_SB_.I2C2.PSAT */ Or (PSAT, Zero, PSAT) /* \_SB_.I2C2.PSAT */ } Method (_PS0, 0, NotSerialized) // _PS0: Power State 0 { And (PSAT, 0xFFFFFFFC, PSAT) /* \_SB_.I2C2.PSAT */ Or (PSAT, Zero, PSAT) /* \_SB_.I2C2.PSAT */ } OperationRegion (KEYS, SystemMemory, I21A, 0x0100) Field (KEYS, DWordAcc, NoLock, WriteAsZeros) { Offset (0x84), PSAT, 32 } Device (RTEK) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "INTCCFFD") // _HID: Hardware ID Name (_CID, "INTCCFFD") // _CID: Compatible ID Name (_DDN, "RTEK Codec Controller ") // _DDN: DOS Device Name Name (_UID, 0x02) // _UID: Unique ID Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { I2cSerialBus (0x001C, ControllerInitiated, 0x00061A80, AddressingMode7Bit, "\\_SB.I2C2", 0x00, ResourceConsumer, , ) GpioInt (Edge, ActiveBoth, SharedAndWake, PullNone, 0x0000, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x000E } GpioInt (Edge, ActiveBoth, SharedAndWake, PullNone, 0x0000, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x0003 } GpioInt (Edge, ActiveBoth, SharedAndWake, PullNone, 0x0000, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x0000 } GpioIo (Shared, PullDefault, 0x0000, 0x0000, IoRestrictionInputOnly, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x000E } GpioIo (Shared, PullDefault, 0x0000, 0x0000, IoRestrictionInputOnly, "\\_SB.GPO0", 0x00, ResourceConsumer, , ) { // Pin list 0x0003, 0x0000 } }) Return (SBUF) /* \_SB_.I2C2.RTEK._CRS.SBUF */ } What am I missing? I suppose I need some changes in the kernel sources but I don't know where to start. Any help is welcome Regards, Jorge Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o CONFIDENCIAL. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. This message is intended exclusively for its addressee and may contain information that is CONFIDENTIAL and protected by professional privilege. If you are not the intended recipient you are hereby notified that any dissemination, copy or disclosure of this communication is strictly prohibited by law. If this message has been received in error, please immediately notify us via e-mail and delete it.

3 10

[alsa-devel] [PATCH] ASoC: sigmadsp: Add check for return value of i2c_master_send
by pascal.huerst＠gmail.com 26 Nov '15

26 Nov '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> i2c_master_send returns the number of bytes transmitted, which leads to an error in the firmware download function. Check for the return value and only return it directly, if it is an actual error number. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

2 1

Re: [alsa-devel] ASoC: hdac-hdmi: Add hdmi driver
by Dan Carpenter 26 Nov '15

26 Nov '15

Hello Subhransu S. Prusty, The patch 18382ead3640: "ASoC: hdac-hdmi: Add hdmi driver" from Nov 10, 2015, leads to the following static checker warning: sound/soc/codecs/hdac_hdmi.c:416 hdac_hdmi_parse_and_map_nid() warn: unsigned 'hdac->num_nodes' is never less than zero. sound/soc/codecs/hdac_hdmi.c 409 static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev) 410 { 411 hda_nid_t nid; 412 int i; 413 struct hdac_device *hdac = &edev->hdac; 414 struct hdac_hdmi_priv *hdmi = edev->private_data; 415 int cvt_nid = 0, pin_nid = 0; 416 417 hdac->num_nodes = snd_hdac_get_sub_nodes(hdac, hdac->afg, &nid); 418 if (!nid || hdac->num_nodes < 0) { ^^^^^^^^^^^^^^^^^^^ snd_hdac_get_sub_nodes() never returns negatives. (We can't create a static checker warning for this because it's impossible to know if the author is planning to add error returns later). It returns zero if not found so maybe that was intended intsead? 419 dev_warn(&hdac->dev, "HDMI: failed to get afg sub nodes\n"); 420 return -EINVAL; 421 } regards, dan carpenter

1 0

[alsa-devel] [PATCH] ASoC: rsnd: tidyup semantics of rsnd_ssi_record_error()
by Kuninori Morimoto 26 Nov '15

26 Nov '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_ssi_record_error() should recorde error, but it clears error too. this patch fixes up semantic of rsnd_ssi_record_error that it records error but doesn't clear error. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index dd693c0..5b97ae8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -395,13 +395,9 @@ static u32 rsnd_ssi_record_error(struct rsnd_ssi *ssi) u32 status = rsnd_ssi_status_get(mod); /* under/over flow error */ - if (status & (UIRQ | OIRQ)) { + if (status & (UIRQ | OIRQ)) ssi->err++; - /* clear error status */ - rsnd_ssi_status_clear(mod); - } - return status; } @@ -536,6 +532,7 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, rsnd_mod_name(mod), rsnd_mod_id(mod)); } + rsnd_ssi_status_clear(mod); rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: rsnd: tidyup semantics of rsnd_src_record_error()
by Kuninori Morimoto 26 Nov '15

26 Nov '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_src_error_record() should recorde error, but it clears error too. this patch fixes up semantic of rsnd_src_error_record that it records error but doesn't clear error. And this patch renames rsnd_src_error_clear() to rsnd_src_status_clear() rsnd_src_error_record() to rsnd_src_record_error() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index c103aa7..6d93c4e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -300,7 +300,7 @@ static void rsnd_src_irq_ctrol(struct rsnd_mod *mod, int enable) rsnd_mod_bset(mod, SCU_SYS_INT_EN1, sys_int_mask, sys_int_val); } -static void rsnd_src_error_clear(struct rsnd_mod *mod) +static void rsnd_src_status_clear(struct rsnd_mod *mod) { u32 val = OUF_SRC(rsnd_mod_id(mod)); @@ -308,7 +308,7 @@ static void rsnd_src_error_clear(struct rsnd_mod *mod) rsnd_mod_bset(mod, SCU_SYS_STATUS1, val, val); } -static bool rsnd_src_error_record(struct rsnd_mod *mod) +static bool rsnd_src_record_error(struct rsnd_mod *mod) { struct rsnd_src *src = rsnd_mod_to_src(mod); u32 val0, val1; @@ -332,9 +332,6 @@ static bool rsnd_src_error_record(struct rsnd_mod *mod) ret = true; } - /* clear error static */ - rsnd_src_error_clear(mod); - return ret; } @@ -383,7 +380,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, rsnd_src_set_convert_rate(io, mod); - rsnd_src_error_clear(mod); + rsnd_src_status_clear(mod); rsnd_src_irq_enable(mod); @@ -434,7 +431,7 @@ static void __rsnd_src_interrupt(struct rsnd_mod *mod, if (!rsnd_io_is_working(io)) goto rsnd_src_interrupt_out; - if (rsnd_src_error_record(mod)) { + if (rsnd_src_record_error(mod)) { dev_dbg(dev, "%s[%d] restart\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); @@ -450,7 +447,9 @@ static void __rsnd_src_interrupt(struct rsnd_mod *mod, rsnd_mod_name(mod), rsnd_mod_id(mod)); } + rsnd_src_status_clear(mod); rsnd_src_interrupt_out: + spin_unlock(&priv->lock); } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: rt5645: Add dmi_system_id "Google Terra"
by Bard Liao 26 Nov '15

26 Nov '15

Add platform specific data for Terra project. Signed-off-by: Luke_Yin(a)asus.com <Luke_Yin(a)asus.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 2c4f06f..7b140cc 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3572,6 +3572,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"), }, }, + { + .ident = "Google Terra", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Terra"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: convert num_nodes to int
by Sudip Mukherjee 26 Nov '15

26 Nov '15

All uses of num_nodes are considering it as a signed integer and that is very much clear when we try to save the error value in it and later try to compare it with less than 0. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- include/sound/hdaudio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index e2b712c..c5fad14 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -71,7 +71,7 @@ struct hdac_device { unsigned int flags, unsigned int *res); /* widgets */ - unsigned int num_nodes; + int num_nodes; hda_nid_t start_nid, end_nid; /* misc flags */ -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ASoC: rcar: remove unused variable
by Mark Brown 26 Nov '15

26 Nov '15

On Wed, Nov 25, 2015 at 11:54:06PM +0000, Kuninori Morimoto wrote: > It is possible to remove this patch from topic/rcar, > and, apply it to topic/pcm-list ? > ccae40fa973f5b0eec5d5ad27e81c5a2b95f7b7c > (" ASoC: rcar: remove unused variable") I did that already.

2 1

[alsa-devel] [PATCH] ASoC: Atmel: ClassD: add GCK's parent clock in DT binding
by Songjun Wu 26 Nov '15

26 Nov '15

Set GCK's parent as audio clock. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- .../devicetree/bindings/sound/atmel-classd.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt index 0018451..549e701 100644 --- a/Documentation/devicetree/bindings/sound/atmel-classd.txt +++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt @@ -16,6 +16,10 @@ Required properties: Required elements: "pclk", "gclk" and "aclk". - clocks Please refer to clock-bindings.txt. +- assigned-clocks + Should be <&classd_gclk>. +- assigned-clock-parents + Should be <&audio_pll_pmc>. Optional properties: - pinctrl-names, pinctrl-0 @@ -43,6 +47,8 @@ classd: classd@fc048000 { dma-names = "tx"; clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>; clock-names = "pclk", "gclk", "aclk"; + assigned-clocks = <&classd_gclk>; + assigned-clock-parents = <&audio_pll_pmc>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_classd_default>; -- 1.7.9.5

3 3

[alsa-devel] [PATCH v2] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by timothyc.howe＠gmail.com 25 Nov '15

25 Nov '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 930 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 272 ++++++ 5 files changed, 1230 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5899744..a3c7911 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -429,6 +430,10 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9c921c7..60432ac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -244,6 +245,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..d91ce73 +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,930 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, THMS_PDN }, + { CS53L30_MCLKCTL, MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, INTERNAL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, ASP_RATE_48K }, + { CS53L30_ASP1_CTL, ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, MUTEP_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, INBIAS_CTLX_DFLT }, + { CS53L30_INBIAS_CTL2, INBIAS_CTLX_DFLT }, + { CS53L30_DMIC1_STR_CTL, DMIC1_STEREO_DFLT }, + { CS53L30_DMIC2_STR_CTL, DMIC2_STEREO_DFLT }, + { CS53L30_ADCDMIC1_CTL1, ADC1_ON_CH_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, ADC1_DMIC1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, ADC2_ON_CH_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +static int input1_sel_values[] = { DMIC1_ON_CH_AB_IN, DMIC1_ON_CH_A_IN, + DMIC1_ON_CH_B_IN, ADC1_ON_CH_AB_IN, ADC1_ON_CH_A_IN, ADC1_ON_CH_B_IN, + DMIC1_OFF_ADC1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +static int input2_sel_values[] = { DMIC2_ON_CH_AB_IN, DMIC2_ON_CH_A_IN, + DMIC2_ON_CH_B_IN, ADC2_ON_CH_AB_IN, ADC2_ON_CH_A_IN, ADC2_ON_CH_B_IN, + DMIC2_OFF_ADC2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 1); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, + CS53L30_ASP1_CTL, ASP1_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, + CS53L30_ASP2_CTL, ASP2_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= ASP_SCLK_INV; + else + asp_config_ctl &= ~ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = + cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_write(priv->regmap, CS53L30_ASPCFG_CTL, priv->asp_config_ctl); + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= INTERNAL_FS_RATIO; + else + int_sr_ctl &= ~INTERNAL_FS_RATIO; + regmap_write(priv->regmap, CS53L30_INT_SR_CTL, int_sr_ctl); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= MCLK_INT_SCALE; + else + mclk_ctl &= ~MCLK_INT_SCALE; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->asp_config_ctl &= ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & ASP_RATE); + regmap_write(priv->regmap, CS53L30_MCLKCTL, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & DIGSFT) + inter_max_check = PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + msleep(1); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & PDN_DONE) + break; + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, MCLK_DIS); + break; + } + dapm->bias_level = level; + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, ASP1_3ST, + (ASP1_3ST_VAL(tristate) & ASP1_3ST)); +} + +static int cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .owner = THIS_MODULE, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..f45e91f +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,272 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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 __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define PDN_ULP (1 << 7) +#define PDN_LP (1 << 6) +#define DISCHARGE_FILT (1 << 5) +#define THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define MCLK_DIS (1 << 7) +#define MCLK_INT_SCALE (1 << 6) +#define DMIC_DRIVE (1 << 5) +#define MCLK_DIV (3 << 2) +#define MCLK_DIV_DFLT (1 << 2) +#define SYNC_EN (1 << 1) + +/* CS53L30_INT_SR_CTL */ +#define INTERNAL_FS_RATIO (1 << 4) +#define INTERNAL_FS_DFLT (7 << 2) +#define MCLK_19MHZ_EN (1 << 0) + +/* CS53L30_MICBIAS_CTL */ +#define MIC4_BIAS_PDN (1 << 7) +#define MIC3_BIAS_PDN (1 << 6) +#define MIC2_BIAS_PDN (1 << 5) +#define MIC1_BIAS_PDN (1 << 4) +#define VP_MIN (1 << 2) +#define MIC_BIAS_CTRL (3 << 0) +#define MIC_BIAS_ALL_PDN 0xF0 +#define MIC_BIAS_DFLT (MIC_BIAS_ALL_PDN | VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define ASP_MS (1 << 7) +#define ASP_SCLK_INV (1 << 4) +#define ASP_RATE_48K (3 << 2) +#define ASP_RATE 0x0F +#define ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define ASP1_TDM_PDN (1 << 7) +#define ASP1_SDOUT_PDN (1 << 6) +#define ASP1_3ST (1 << 5) +#define SHIFT_LEFT (1 << 4) +#define ASP1_DRIVE (1 << 0) +#define ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define ASP1_CHX_TX_STATE (1 << 7) +#define ASP1_CHX_TX_LOC 0x3F +#define ASP1_CHX_TX_DFLT_SLT47 0x2F +#define ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define ASP2_SDOUT_PDN (1 << 6) +#define ASP2_DRIVE (1 << 0) +#define ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define DIGSFT (1 << 5) +#define SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define LRCK_50_NPW (1 << 3) +#define LRCK_TPWH (7 << 0) +#define LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define MUTE_PDN_ULP (1 << 7) +#define MUTE_PDN_LP (1 << 6) +#define MUTE_M4B_PDN (1 << 4) +#define MUTE_M3B_PDN (1 << 3) +#define MUTE_M2B_PDN (1 << 2) +#define MUTE_M1B_PDN (1 << 1) +#define MUTE_MB_ALL_PDN (1 << 0) +#define MUTEP_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define MUTE_PIN_POLARITY (1 << 7) +#define MUTE_ASP_TDM_PDN (1 << 6) +#define MUTE_ASP_SDOUT2_PDN (1 << 5) +#define MUTE_ASP_SDOUT1_PDN (1 << 4) +#define MUTE_ADC2B_PDN (1 << 3) +#define MUTE_ADC2A_PDN (1 << 2) +#define MUTE_ADC1B_PDN (1 << 1) +#define MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define IN4M_BIAS (3 << 6) +#define IN4P_BIAS (3 << 4) +#define IN3M_BIAS (3 << 2) +#define IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define IN2M_BIAS (3 << 6) +#define IN2P_BIAS (3 << 4) +#define IN1M_BIAS (3 << 2) +#define IN1P_BIAS (3 << 0) +#define INBIAS_CTLX_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define DMIC1_STEREO_ENB (1 << 5) +#define DMIC1_STEREO_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define DMIC2_STEREO_EN (1 << 5) +#define DMIC2_STEREO_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define ADC1B_PDN (1 << 7) +#define ADC1A_PDN (1 << 6) +#define DMIC1_PDN (1 << 2) +#define DMIC1_SCLK_DIV (1 << 1) +#define CH_TYPE (1 << 0) +#define DMIC1_ON_CH_AB_IN (CH_TYPE) +#define DMIC1_ON_CH_A_IN (ADC1B_PDN | CH_TYPE) +#define DMIC1_ON_CH_B_IN (ADC1A_PDN | CH_TYPE) +#define ADC1_ON_CH_AB_IN (DMIC1_PDN) +#define ADC1_ON_CH_A_IN (ADC1B_PDN | DMIC1_PDN) +#define ADC1_ON_CH_B_IN (ADC1A_PDN | DMIC1_PDN) +#define DMIC1_OFF_ADC1_OFF (ADC1A_PDN | ADC1B_PDN | DMIC1_PDN) +#define ADC1_DMIC1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define ADC1_NOTCH_DIS (1 << 7) +#define ADC1B_INV (1 << 5) +#define ADC1A_INV (1 << 4) +#define ADC1B_DIG_BOOST (1 << 1) +#define ADC1A_DIG_BOOST (1 << 0) +#define ADC1_DMIC1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define ADC1_HPF_EN (1 << 3) +#define ADC1_HPF_CF (3 << 1) +#define ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define ADC1B_NG (1 << 7) +#define ADC1A_NG (1 << 6) +#define ADC1_NG_BOOST (1 << 5) +#define ADC1_NG_THRESH (7 << 2) +#define ADC1_NG_DELAY (3 << 0) +#define ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define ADC1A_PREAMP (3 << 6) +#define ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define ADC1B_PREAMP (3 << 6) +#define ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define ADC2B_PDN (1 << 7) +#define ADC2A_PDN (1 << 6) +#define DMIC2_PDN (1 << 2) +#define DMIC2_CLKDIV (1 << 1) +#define DMIC2_ON_CH_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_A_IN (ADC2B_PDN) /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_B_IN (ADC2A_PDN) /* CH_TYPE must = 1 */ +#define ADC2_ON_CH_AB_IN (DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_A_IN (ADC2B_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_B_IN (ADC2A_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define DMIC2_OFF_ADC2_OFF (ADC2A_PDN | ADC2B_PDN | DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define ADC2_NOTCH_DIS (1 << 7) +#define ADC2B_INV (1 << 5) +#define ADC2A_INV (1 << 4) +#define ADC2B_DIG_BOOST (1 << 1) +#define ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define ADC2_HPF_EN (1 << 3) +#define ADC2_HPF_CF (3 << 1) +#define ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define PDN_DONE (1 << 7) +#define THMS_TRIP (1 << 6) +#define SYNC_DONE (1 << 5) +#define ADC2B_OVFL (1 << 4) +#define ADC2A_OVFL (1 << 3) +#define ADC1B_OVFL (1 << 2) +#define ADC1A_OVFL (1 << 1) +#define MUTE_PIN (1 << 0) +#define DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

1 0

[alsa-devel] [RFC PATCH 02/10] ASoC: Define soc_init_dai_link() to wrap link intialization.
by mengdong.lin＠intel.com 25 Nov '15

25 Nov '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Define soc_init_dai_link() to wrap link initialization, to reuse it later by snd_soc_instantiate_card() when adding new DAI links from topology in component probing phase. Move static func snd_soc_init_multicodec(), so that it can be reused by soc_init_dai_link(). This saves adding a function declaration for snd_soc_init_multicodec(). Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb78674..e9e8956 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1141,6 +1141,100 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].name = dai_link->codec_name; + dai_link->codecs[0].of_node = dai_link->codec_of_node; + dai_link->codecs[0].dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + +static int soc_init_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link) +{ + int i, ret; + + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; + } + + for (i = 0; i < link->num_codecs; i++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[i].name == + !!link->codecs[i].of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[i].dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "ASoC: Both platform name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + /* + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified + */ + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + return 0; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { @@ -2363,33 +2457,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); -static int snd_soc_init_multicodec(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - /* Legacy codec/codec_dai link is a single entry in multicodec */ - if (dai_link->codec_name || dai_link->codec_of_node || - dai_link->codec_dai_name) { - dai_link->num_codecs = 1; - - dai_link->codecs = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai_link_component), - GFP_KERNEL); - if (!dai_link->codecs) - return -ENOMEM; - - dai_link->codecs[0].name = dai_link->codec_name; - dai_link->codecs[0].of_node = dai_link->codec_of_node; - dai_link->codecs[0].dai_name = dai_link->codec_dai_name; - } - - if (!dai_link->codecs) { - dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); - return -EINVAL; - } - - return 0; -} - /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2398,7 +2465,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, j, ret; + int i, ret; struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) @@ -2407,63 +2474,11 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - ret = snd_soc_init_multicodec(card, link); + ret = soc_init_dai_link(card, link); if (ret) { - dev_err(card->dev, "ASoC: failed to init multicodec\n"); - return ret; - } - - for (j = 0; j < link->num_codecs; j++) { - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codecs[j].name == - !!link->codecs[j].of_node) { - dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* Codec DAI name must be specified */ - if (!link->codecs[j].dai_name) { - dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; - } - } - - /* - * Platform may be specified by either name or OF node, but - * can be left unspecified, and a dummy platform will be used. - */ - if (link->platform_name && link->platform_of_node) { - dev_err(card->dev, - "ASoC: Both platform name/of_node are set for %s\n", + dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); - return -EINVAL; - } - - /* - * CPU device may be specified by either name or OF node, but - * can be left unspecified, and will be matched based on DAI - * name alone.. - */ - if (link->cpu_name && link->cpu_of_node) { - dev_err(card->dev, - "ASoC: Neither/both cpu name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* - * At least one of CPU DAI name or CPU device name/node must be - * specified - */ - if (!link->cpu_dai_name && - !(link->cpu_name || link->cpu_of_node)) { - dev_err(card->dev, - "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", - link->name); - return -EINVAL; + return ret; } } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ALSA: compress: Add procfs info file for compressed nodes
by Richard Fitzgerald 25 Nov '15

25 Nov '15

This updates the compress core code to create an 'info' file under procfs for each compressed node, like the PCM core does for PCM nodes. Based off Takashi's for-next Richard Fitzgerald (2): ALSA: compress: Add procfs info file for compressed nodes ALSA: compress: Pass id string to snd_compress_new include/sound/compress_driver.h | 7 +++- sound/core/compress_offload.c | 81 ++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-compress.c | 8 +++- 3 files changed, 92 insertions(+), 4 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH V4 0/3] add spba clock for fsl audio IP
by Shengjiu Wang 25 Nov '15

25 Nov '15

add spba clock for fsl audio IP Changes in V3 - add error check for spba clock Shengjiu Wang (3): ASoC: fsl_esai: spba clock is needed by esai device ASoC: fsl_spdif: spba clk is needed by spdif device ASoC: fsl_asrc: spba clock is needed by asrc device Documentation/devicetree/bindings/sound/fsl,asrc.txt | 5 +++++ Documentation/devicetree/bindings/sound/fsl,esai.txt | 5 +++++ Documentation/devicetree/bindings/sound/fsl,spdif.txt | 5 +++++ sound/soc/fsl/fsl_asrc.c | 14 ++++++++++++++ sound/soc/fsl/fsl_asrc.h | 2 ++ sound/soc/fsl/fsl_esai.c | 17 +++++++++++++++++ sound/soc/fsl/fsl_spdif.c | 19 +++++++++++++++++++ 7 files changed, 67 insertions(+) -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: Fix typo in kernel doc comment for snd_soc_put_volsw_sx()
by Takashi Iwai 25 Nov '15

25 Nov '15

Spotted by kbuild bot: sound/soc/soc-ops.c:415: warning: No description found for parameter 'ucontrol' sound/soc/soc-ops.c:415: warning: Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index ecd38e52285a..2f67ba6d7a8f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * -- 2.6.2

2 2

[alsa-devel] [PATCH 0/3] ASoC: core: dapm more updates
by Vinod Koul 25 Nov '15

25 Nov '15

This series contains three updates we have for dapm First marks the SND_SOC_BYTES_EXT as deprecated as discussed Second one adds support for dapm sink widget. This is useful in DSP based systems where a module can act as a sink of data Last changes order of power state to before rechecking endpoint. This ensure DAPM is restored properly before userspace can access the card. Jeeja KP (1): ASoC: core: Change power state before rechecking endpoint Vinod Koul (2): ASoC: core: mark SND_SOC_BYTES_EXT as deprecated ASoC: core: add a dapm sink widget include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 6 +++--- sound/soc/soc-dapm.c | 5 +++++ 4 files changed, 15 insertions(+), 3 deletions(-) -- 1.9.1

3 13

[alsa-devel] [PATCH] ASoC: hdac_hdmi: remove always false condition
by Sudip Mukherjee 25 Nov '15

25 Nov '15

hdac->num_nodes is unsigned int and can never be less than 0. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- Or, maybe, another way can be to declare num_nodes as int. Then this test makes sense and also snd_hdac_get_sub_nodes() is returning an int. sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 205f2c2..02109aa 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -415,7 +415,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev) int cvt_nid = 0, pin_nid = 0; hdac->num_nodes = snd_hdac_get_sub_nodes(hdac, hdac->afg, &nid); - if (!nid || hdac->num_nodes < 0) { + if (!nid) { dev_warn(&hdac->dev, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: img: i2s in: Add AIF widgets
by Damien Horsley 25 Nov '15

25 Nov '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add AIF widgets to i2s input controller This allows connections to this component in the dapm graph Signed-off-by: Damien.Horsley <Damien.Horsley(a)imgtec.com> --- sound/soc/img/img-i2s-in.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 0389203..792334c 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -356,8 +356,14 @@ static int img_i2s_in_dai_probe(struct snd_soc_dai *dai) return 0; } +static const struct snd_soc_dapm_widget img_i2s_in_widgets[] = { + SND_SOC_DAPM_AIF_OUT("IMG I2S IN", "Capture", 0, SND_SOC_NOPM, 0, 0) +}; + static const struct snd_soc_component_driver img_i2s_in_component = { - .name = "img-i2s-in" + .name = "img-i2s-in", + .dapm_widgets = img_i2s_in_widgets, + .num_dapm_widgets = ARRAY_SIZE(img_i2s_in_widgets) }; static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st, @@ -441,6 +447,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) i2s->dai_driver.capture.rates = SNDRV_PCM_RATE_8000_192000; i2s->dai_driver.capture.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE; + i2s->dai_driver.capture.stream_name = "Capture"; i2s->dai_driver.ops = &img_i2s_in_dai_ops; rst = devm_reset_control_get(dev, "rst"); -- 2.1.4

2 2

[alsa-devel] [PATCH] ASoC: fsl: clarify ac97 dependency
by Arnd Bergmann 25 Nov '15

25 Nov '15

A new randconfig build failure shows that the fsl-asoc-card module must not be built-in when the AC97 driver is a loadable module: sound/built-in.o: In function `fsl_asoc_card_late_probe': :(.text+0x571d8): undefined reference to `snd_ac97_update_bits' I couldn't come up with a nice solution, so this adds another dependency on "X || !X", which is the Kconfig way of saying that we have an optional dependency on something that might be a loadable module. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 50760cad9de9 ("ASoC: fsl-asoc-card: add AC'97 support") --- The commit that introduced this was merged for 4.4-rc1, so we might want the fix to go into 4.4 as well. diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19c302b0d763..14dfdee05fd5 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -283,6 +283,8 @@ config SND_SOC_IMX_MC13783 config SND_SOC_FSL_ASOC_CARD tristate "Generic ASoC Sound Card with ASRC support" depends on OF && I2C + # enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m: + depends on SND_AC97_CODEC || SND_AC97_CODEC=n select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_ESAI

3 2

[alsa-devel] [PATCH] ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in
by Jie Yang 25 Nov '15

25 Nov '15

The previous commit ef3e199a49c8 ("ASoC: Intel: sst: only use sst-firmware when DW DMAC is available") does not fix the 0day building errors thoroughly: sound/built-in.o: In function 'dw_dma_remove' sound/built-in.o: In function 'dw_dma_probe' Here we fallback to select sst-firmware only when DW DMAC is built-in selected. We may need to refactor sst common driver and split DW related codes to platform driver, but ATM, this fallback may be the smallest fix. Please be noticed that after applying this patch, we may need select DW DMAC manually in DMA driver menu, before we can prompt and select HSW/BDW and old BYT machines. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 8 ++++---- sound/soc/intel/common/Makefile | 4 +--- sound/soc/intel/common/sst-dsp.c | 2 +- sound/soc/intel/common/sst-dsp.h | 2 +- 4 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 13a7621..99ce3bf 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE + depends on DW_DMAC_CORE=y select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 @@ -57,7 +57,7 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE + depends on DW_DMAC_CORE=y select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 @@ -68,7 +68,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE + depends on DW_DMAC_CORE=y select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 @@ -80,7 +80,7 @@ config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE + depends on DW_DMAC_CORE=y select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 658edce..3b9332e 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -2,9 +2,7 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o sst-match-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o -ifneq ($(CONFIG_DW_DMAC_CORE),) -snd-soc-sst-dsp-objs += sst-firmware.o -endif +snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += 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 diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index c9452e0..b5bbdf4 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -420,7 +420,7 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) } EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); -#if IS_ENABLED(CONFIG_DW_DMAC_CORE) +#ifdef CONFIG_DW_DMAC_CORE struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) { diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 859f0de..0b84c71 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -216,7 +216,7 @@ struct sst_pdata { void *dsp; }; -#if IS_ENABLED(CONFIG_DW_DMAC_CORE) +#ifdef CONFIG_DW_DMAC_CORE /* Initialization */ struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: rockchip: Fix incorrect VDW value for 24 bit
by Sjoerd Simons 25 Nov '15

25 Nov '15

Correct valid data word register value for 24 bit data width. The bit value should be 10 (aka 0x2), not 0x10. This fixes playback of 24 bit audio. Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> --- sound/soc/rockchip/rockchip_spdif.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h index 07f86a2..921b409 100644 --- a/sound/soc/rockchip/rockchip_spdif.h +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -28,9 +28,9 @@ #define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT) #define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT) -#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00) -#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01) -#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10) +#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x0) +#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x1) +#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x2) /* * DMACR -- 2.6.2

3 2

[alsa-devel] [PATCH v2 0/2] ALSA: compress: Add procfs info file for compressed nodes
by Richard Fitzgerald 25 Nov '15

25 Nov '15

This updates the compress core code to create an 'info' file under procfs for each compressed node, like the PCM core does for PCM nodes. Based off Takashi's for-next Richard Fitzgerald (2): ALSA: compress: Add procfs info file for compressed nodes ALSA: compress: Pass id string to snd_compress_new include/sound/compress_driver.h | 7 +++- sound/core/compress_offload.c | 81 ++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-compress.c | 8 +++- 3 files changed, 92 insertions(+), 4 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by Tim Howe 25 Nov '15

25 Nov '15

Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 930 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 272 ++++++ 5 files changed, 1230 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5899744..a3c7911 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -429,6 +430,10 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9c921c7..60432ac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -244,6 +245,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..d91ce73 +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,930 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, THMS_PDN }, + { CS53L30_MCLKCTL, MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, INTERNAL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, ASP_RATE_48K }, + { CS53L30_ASP1_CTL, ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, ASP1_CHX_TX_DFLT_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, MUTEP_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, INBIAS_CTLX_DFLT }, + { CS53L30_INBIAS_CTL2, INBIAS_CTLX_DFLT }, + { CS53L30_DMIC1_STR_CTL, DMIC1_STEREO_DFLT }, + { CS53L30_DMIC2_STR_CTL, DMIC2_STEREO_DFLT }, + { CS53L30_ADCDMIC1_CTL1, ADC1_ON_CH_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, ADC1_DMIC1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, ADC2_ON_CH_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +static int input1_sel_values[] = { DMIC1_ON_CH_AB_IN, DMIC1_ON_CH_A_IN, + DMIC1_ON_CH_B_IN, ADC1_ON_CH_AB_IN, ADC1_ON_CH_A_IN, ADC1_ON_CH_B_IN, + DMIC1_OFF_ADC1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +static int input2_sel_values[] = { DMIC2_ON_CH_AB_IN, DMIC2_ON_CH_A_IN, + DMIC2_ON_CH_B_IN, ADC2_ON_CH_AB_IN, ADC2_ON_CH_A_IN, ADC2_ON_CH_B_IN, + DMIC2_OFF_ADC2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + ADC1_DMIC1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + ASP1_3ST, 1); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, + CS53L30_ASP1_CTL, ASP1_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, + CS53L30_ASP2_CTL, ASP2_SDOUT_PDN, + 1, cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= ASP_SCLK_INV; + else + asp_config_ctl &= ~ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = + cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_write(priv->regmap, CS53L30_ASPCFG_CTL, priv->asp_config_ctl); + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= INTERNAL_FS_RATIO; + else + int_sr_ctl &= ~INTERNAL_FS_RATIO; + regmap_write(priv->regmap, CS53L30_INT_SR_CTL, int_sr_ctl); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= MCLK_INT_SCALE; + else + mclk_ctl &= ~MCLK_INT_SCALE; + + regmap_write(priv->regmap, CS53L30_MCLKCTL, mclk_ctl); + priv->asp_config_ctl &= ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & ASP_RATE); + regmap_write(priv->regmap, CS53L30_MCLKCTL, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_LP, PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & DIGSFT) + inter_max_check = PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + PDN_ULP, PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + msleep(1); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & PDN_DONE) + break; + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + PDN_DONE, PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + MCLK_DIS, MCLK_DIS); + break; + } + dapm->bias_level = level; + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, ASP1_3ST, + (ASP1_3ST_VAL(tristate) & ASP1_3ST)); +} + +static int cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .owner = THIS_MODULE, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..f45e91f +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,272 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.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 __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define PDN_ULP (1 << 7) +#define PDN_LP (1 << 6) +#define DISCHARGE_FILT (1 << 5) +#define THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define MCLK_DIS (1 << 7) +#define MCLK_INT_SCALE (1 << 6) +#define DMIC_DRIVE (1 << 5) +#define MCLK_DIV (3 << 2) +#define MCLK_DIV_DFLT (1 << 2) +#define SYNC_EN (1 << 1) + +/* CS53L30_INT_SR_CTL */ +#define INTERNAL_FS_RATIO (1 << 4) +#define INTERNAL_FS_DFLT (7 << 2) +#define MCLK_19MHZ_EN (1 << 0) + +/* CS53L30_MICBIAS_CTL */ +#define MIC4_BIAS_PDN (1 << 7) +#define MIC3_BIAS_PDN (1 << 6) +#define MIC2_BIAS_PDN (1 << 5) +#define MIC1_BIAS_PDN (1 << 4) +#define VP_MIN (1 << 2) +#define MIC_BIAS_CTRL (3 << 0) +#define MIC_BIAS_ALL_PDN 0xF0 +#define MIC_BIAS_DFLT (MIC_BIAS_ALL_PDN | VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define ASP_MS (1 << 7) +#define ASP_SCLK_INV (1 << 4) +#define ASP_RATE_48K (3 << 2) +#define ASP_RATE 0x0F +#define ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define ASP1_TDM_PDN (1 << 7) +#define ASP1_SDOUT_PDN (1 << 6) +#define ASP1_3ST (1 << 5) +#define SHIFT_LEFT (1 << 4) +#define ASP1_DRIVE (1 << 0) +#define ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define ASP1_CHX_TX_STATE (1 << 7) +#define ASP1_CHX_TX_LOC 0x3F +#define ASP1_CHX_TX_DFLT_SLT47 0x2F +#define ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define ASP2_SDOUT_PDN (1 << 6) +#define ASP2_DRIVE (1 << 0) +#define ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define DIGSFT (1 << 5) +#define SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define LRCK_50_NPW (1 << 3) +#define LRCK_TPWH (7 << 0) +#define LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define MUTE_PDN_ULP (1 << 7) +#define MUTE_PDN_LP (1 << 6) +#define MUTE_M4B_PDN (1 << 4) +#define MUTE_M3B_PDN (1 << 3) +#define MUTE_M2B_PDN (1 << 2) +#define MUTE_M1B_PDN (1 << 1) +#define MUTE_MB_ALL_PDN (1 << 0) +#define MUTEP_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define MUTE_PIN_POLARITY (1 << 7) +#define MUTE_ASP_TDM_PDN (1 << 6) +#define MUTE_ASP_SDOUT2_PDN (1 << 5) +#define MUTE_ASP_SDOUT1_PDN (1 << 4) +#define MUTE_ADC2B_PDN (1 << 3) +#define MUTE_ADC2A_PDN (1 << 2) +#define MUTE_ADC1B_PDN (1 << 1) +#define MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define IN4M_BIAS (3 << 6) +#define IN4P_BIAS (3 << 4) +#define IN3M_BIAS (3 << 2) +#define IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define IN2M_BIAS (3 << 6) +#define IN2P_BIAS (3 << 4) +#define IN1M_BIAS (3 << 2) +#define IN1P_BIAS (3 << 0) +#define INBIAS_CTLX_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define DMIC1_STEREO_ENB (1 << 5) +#define DMIC1_STEREO_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define DMIC2_STEREO_EN (1 << 5) +#define DMIC2_STEREO_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define ADC1B_PDN (1 << 7) +#define ADC1A_PDN (1 << 6) +#define DMIC1_PDN (1 << 2) +#define DMIC1_SCLK_DIV (1 << 1) +#define CH_TYPE (1 << 0) +#define DMIC1_ON_CH_AB_IN (CH_TYPE) +#define DMIC1_ON_CH_A_IN (ADC1B_PDN | CH_TYPE) +#define DMIC1_ON_CH_B_IN (ADC1A_PDN | CH_TYPE) +#define ADC1_ON_CH_AB_IN (DMIC1_PDN) +#define ADC1_ON_CH_A_IN (ADC1B_PDN | DMIC1_PDN) +#define ADC1_ON_CH_B_IN (ADC1A_PDN | DMIC1_PDN) +#define DMIC1_OFF_ADC1_OFF (ADC1A_PDN | ADC1B_PDN | DMIC1_PDN) +#define ADC1_DMIC1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define ADC1_NOTCH_DIS (1 << 7) +#define ADC1B_INV (1 << 5) +#define ADC1A_INV (1 << 4) +#define ADC1B_DIG_BOOST (1 << 1) +#define ADC1A_DIG_BOOST (1 << 0) +#define ADC1_DMIC1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define ADC1_HPF_EN (1 << 3) +#define ADC1_HPF_CF (3 << 1) +#define ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define ADC1B_NG (1 << 7) +#define ADC1A_NG (1 << 6) +#define ADC1_NG_BOOST (1 << 5) +#define ADC1_NG_THRESH (7 << 2) +#define ADC1_NG_DELAY (3 << 0) +#define ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define ADC1A_PREAMP (3 << 6) +#define ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define ADC1B_PREAMP (3 << 6) +#define ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define ADC2B_PDN (1 << 7) +#define ADC2A_PDN (1 << 6) +#define DMIC2_PDN (1 << 2) +#define DMIC2_CLKDIV (1 << 1) +#define DMIC2_ON_CH_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_A_IN (ADC2B_PDN) /* CH_TYPE must = 1 */ +#define DMIC2_ON_CH_B_IN (ADC2A_PDN) /* CH_TYPE must = 1 */ +#define ADC2_ON_CH_AB_IN (DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_A_IN (ADC2B_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define ADC2_ON_CH_B_IN (ADC2A_PDN | DMIC2_PDN) /* CH_TYPE must = 0 */ +#define DMIC2_OFF_ADC2_OFF (ADC2A_PDN | ADC2B_PDN | DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define ADC2_NOTCH_DIS (1 << 7) +#define ADC2B_INV (1 << 5) +#define ADC2A_INV (1 << 4) +#define ADC2B_DIG_BOOST (1 << 1) +#define ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define ADC2_HPF_EN (1 << 3) +#define ADC2_HPF_CF (3 << 1) +#define ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define PDN_DONE (1 << 7) +#define THMS_TRIP (1 << 6) +#define SYNC_DONE (1 << 5) +#define ADC2B_OVFL (1 << 4) +#define ADC2A_OVFL (1 << 3) +#define ADC1B_OVFL (1 << 2) +#define ADC1A_OVFL (1 << 1) +#define MUTE_PIN (1 << 0) +#define DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

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Re: [alsa-devel] [PATCH] ASoC: rcar: remove unused variable
by Arnd Bergmann 25 Nov '15

25 Nov '15

On Wednesday 25 November 2015 06:25:36 Kuninori Morimoto wrote: > > Hi Arnd, Mark > > > After a recent cleanup, the soc_card variable became unused > > and now produces a warning: > > > > soc/sh/rcar/core.c: In function '__rsnd_kctrl_new': > > soc/sh/rcar/core.c:801:23: warning: unused variable 'soc_card' [-Wunused-variable] > > > > This removes the variable. > > > > Fixes: 1a497983a5ae ("ASoC: Change the PCM runtime array to a list") > > Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> > > > > diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c > > index c6685f14b9cb..90b244c1f526 100644 > > --- a/sound/soc/sh/rcar/core.c > > +++ b/sound/soc/sh/rcar/core.c > > @@ -798,7 +798,6 @@ static int __rsnd_kctrl_new(struct rsnd_mod *mod, > > void (*update)(struct rsnd_dai_stream *io, > > struct rsnd_mod *mod)) > > { > > - struct snd_soc_card *soc_card = rtd->card; > > struct snd_card *card = rtd->card->snd_card; > > struct snd_kcontrol *kctrl; > > struct snd_kcontrol_new knew = { > > It seems this patch was accepted to topic/rcar branch, > but I got compile error > > /opt/home/morimoto/WORK/linux/sound/soc/sh/rcar/core.c: In function '__rsnd_kctrl_new': > /opt/home/morimoto/WORK/linux/sound/soc/sh/rcar/core.c:807:18: error: 'soc_card' undeclared (first use in this function) > .index = rtd - soc_card->rtd, > ^ > /opt/home/morimoto/WORK/linux/sound/soc/sh/rcar/core.c:807:18: note: each undeclared identifier is reported only once for each function it appears in > make[6]: *** [sound/soc/sh/rcar/core.o] エラー 1 > make[6]: *** 未完了のジョブを待っています.... > make[5]: *** [sound/soc/sh/rcar] エラー 2 > make[4]: *** [sound/soc/sh] エラー 2 > > This __rsnd_kctrl_new() is using "soc_card"... It looks like the commit that removed the user of that variable is in the "topic/pcm-list" branch instead of "topic/rcar". Arnd

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[alsa-devel] [RFC V2 PATCH 0/5] hdmi audio dynamically bind PCM to pin
by libin.yang＠linux.intel.com 25 Nov '15

25 Nov '15

From: Libin Yang <libin.yang(a)linux.intel.com> The patches are trying to support dynamically binding PCM to pin in HDMI/DP audio. The pin will not bind to any PCM if there is not monitor connected. When there is a monitor connected to the port (pin), driver will try to find a proper PCM to assign to the port and setup the pin if necessary. Libin Yang (5): ALSA: hda - hdmi playback without monitor in dynamic pcm bind mode ALSA: hda - hdmi operate spdif based on pcm ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug ALSA: hda - add in_use flag in hda_pcm ALSA: hda - hdmi setup pin when monitor hotplug in pcm dynamic assignment mode sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 361 ++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 343 insertions(+), 19 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH V3 0/3] add spba clock for fsl audio IP
by Shengjiu Wang 25 Nov '15

25 Nov '15

add spba clock for fsl audio IP Changes in V3 - update the comments for clock description. Shengjiu Wang (3): ASoC: fsl_esai: spba clock is needed by esai device ASoC: fsl_spdif: spba clk is needed by spdif device ASoC: fsl_asrc: spba clock is needed by asrc device Documentation/devicetree/bindings/sound/fsl,asrc.txt | 5 +++++ Documentation/devicetree/bindings/sound/fsl,esai.txt | 5 +++++ Documentation/devicetree/bindings/sound/fsl,spdif.txt | 5 +++++ sound/soc/fsl/fsl_asrc.c | 10 ++++++++++ sound/soc/fsl/fsl_asrc.h | 2 ++ sound/soc/fsl/fsl_esai.c | 13 +++++++++++++ sound/soc/fsl/fsl_spdif.c | 15 +++++++++++++++ 7 files changed, 55 insertions(+) -- 1.9.1

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[alsa-devel] [RFC][PATCH 0/8] ALSA: dice: constrain PCM substreams to current sampling transfer frequency
by Takashi Sakamoto 25 Nov '15

25 Nov '15

Hi, This patchset adds a constrain to ALSA dice driver to start PCM substreams and AMDTP packet transferring just at current sampling transfer frequency. Dice hardware doesn't allow drivers to get supported combinations between sampling rate and PCM channels. ALSA dice driver should follow to the hardware design, though current ALSA driver has some over-specifications. As a result, the driver has several issue and brings inconvenience to users. This patchset consists of two parts: * 01-05: to add constrain to current sampling transfer frequency and related code cleanup * 06-08: to ensure and stabilize AMDTP packet transmission As a result, userspace applications can request PCM substreams at current sampling transfer frequency. Therefore, when users want to start PCM substreams at different rate, they should set the rate in advance by the other ways (i.e. ffado-dbus-server/ffado-mixer). Takashi Sakamoto (8): ALSA: dice: limit to current sampling transfer frequency ALSA: dice: limit stream to current sampling transfer frequency. ALSA: dice: add MIDI ports according to current number of MIDI substreams ALSA: dice: get the number of MBLA data channel at opening PCM substream ALSA: dice: purge generating channel cache ALSA: dice: ensure phase lock before starting streaming ALSA: dice: expand timeout to wait for Dice notification ALSA: dice: wait for ensuring phase lock sound/firewire/dice/dice-midi.c | 25 +++- sound/firewire/dice/dice-pcm.c | 205 ++++++++++----------------------- sound/firewire/dice/dice-stream.c | 118 +++++++++++++------ sound/firewire/dice/dice-transaction.c | 109 ------------------ sound/firewire/dice/dice.c | 67 +---------- sound/firewire/dice/dice.h | 14 +-- 6 files changed, 162 insertions(+), 376 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 0/4] New feature and clean up for pcm1792a driver
by Raphael Poggi 24 Nov '15

24 Nov '15

From: Raphael Poggi <poggi.raph(a)gmail.com> Hi, This patch series support for the channel selection (stereo/left/right) based on device tree bindings. It also adds some registers defines and cleans up weird test on function exit. Raphael Poggi (4): sound: codecs: pcm1792a: clean function exit sound: codecs: pcm1792a: add define for register dt-bindings: sound: add binding for pcm1792a sound: codecs: pcm1792a: introduce channel side .../devicetree/bindings/sound/pcm1792a.txt | 5 + sound/soc/codecs/pcm1792a.c | 107 +++++++++++++++------ 2 files changed, 83 insertions(+), 29 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH] ALSA: hda - Fix noise on Gigabyte Z170X mobo
by Takashi Iwai 24 Nov '15

24 Nov '15

Gigabyte Z710X mobo with ALC1150 codec gets significant noises from the analog loopback routes even if their inputs are all muted. Simply kill the aamix for fixing it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=108301 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 e4f80dc4704f..9bedf7c85e29 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1759,6 +1759,7 @@ enum { ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, + ALC882_FIXUP_DISABLE_AAMIX, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1920,6 +1921,8 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action); +static void alc_fixup_disable_aamix(struct hda_codec *codec, + const struct hda_fixup *fix, int action); static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { @@ -2151,6 +2154,10 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC882_FIXUP_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2218,6 +2225,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), + SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.6.3

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Re: [alsa-devel] [PATCH v3 1/5] mfd: arizona: Support Cirrus Logic CS47L24 and WM1831
by Lee Jones 24 Nov '15

24 Nov '15

On Tue, 03 Nov 2015, Richard Fitzgerald wrote: > This patch adds the regmap configuration tables and > core MFD handling for the CS47L24 and WM1831 codecs. > > Note that compared to the other Arizona codecs, these devices > do not have an LDO1 or micsupp regulators, extcon driver, or > the DCVDD isolation control. > > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > --- > drivers/mfd/Kconfig | 18 +- > drivers/mfd/Makefile | 3 + > drivers/mfd/arizona-core.c | 74 +- > drivers/mfd/arizona-irq.c | 40 +- > drivers/mfd/arizona-spi.c | 7 + > drivers/mfd/arizona.h | 4 + > drivers/mfd/cs47l24-tables.c | 1629 ++++++++++++++++++++++++++++++++++++++ > include/linux/mfd/arizona/core.h | 3 + > 8 files changed, 1753 insertions(+), 25 deletions(-) > create mode 100644 drivers/mfd/cs47l24-tables.c Applied, thanks. > diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig > index 4d92df6..9581ebb 100644 > --- a/drivers/mfd/Kconfig > +++ b/drivers/mfd/Kconfig > @@ -1370,24 +1370,30 @@ config MFD_ARIZONA > bool > > config MFD_ARIZONA_I2C > - tristate "Wolfson Microelectronics Arizona platform with I2C" > + tristate "Cirrus Logic/Wolfson Microelectronics Arizona platform with I2C" > select MFD_ARIZONA > select MFD_CORE > select REGMAP_I2C > depends on I2C > help > - Support for the Wolfson Microelectronics Arizona platform audio SoC > - core functionality controlled via I2C. > + Support for the Cirrus Logic/Wolfson Microelectronics Arizona platform > + audio SoC core functionality controlled via I2C. > > config MFD_ARIZONA_SPI > - tristate "Wolfson Microelectronics Arizona platform with SPI" > + tristate "Cirrus Logic/Wolfson Microelectronics Arizona platform with SPI" > select MFD_ARIZONA > select MFD_CORE > select REGMAP_SPI > depends on SPI_MASTER > help > - Support for the Wolfson Microelectronics Arizona platform audio SoC > - core functionality controlled via I2C. > + Support for the Cirrus Logic/Wolfson Microelectronics Arizona platform > + audio SoC core functionality controlled via I2C. > + > +config MFD_CS47L24 > + bool "Cirrus Logic CS47L24 and WM1831" > + depends on MFD_ARIZONA > + help > + Support for Cirrus Logic CS47L24 and WM1831 low power audio SoC > > config MFD_WM5102 > bool "Wolfson Microelectronics WM5102" > diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile > index a8b76b8..3fa1df4 100644 > --- a/drivers/mfd/Makefile > +++ b/drivers/mfd/Makefile > @@ -51,6 +51,9 @@ endif > ifeq ($(CONFIG_MFD_WM8998),y) > obj-$(CONFIG_MFD_ARIZONA) += wm8998-tables.o > endif > +ifeq ($(CONFIG_MFD_CS47L24),y) > +obj-$(CONFIG_MFD_ARIZONA) += cs47l24-tables.o > +endif > obj-$(CONFIG_MFD_WM8400) += wm8400-core.o > wm831x-objs := wm831x-core.o wm831x-irq.o wm831x-otp.o > wm831x-objs += wm831x-auxadc.o > diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c > index d474732..b9489a0 100644 > --- a/drivers/mfd/arizona-core.c > +++ b/drivers/mfd/arizona-core.c > @@ -598,6 +598,12 @@ static int arizona_runtime_resume(struct device *dev) > goto err; > } > break; > + case WM1831: > + case CS47L24: > + ret = arizona_wait_for_boot(arizona); > + if (ret != 0) > + goto err; > + break; > default: > ret = arizona_wait_for_boot(arizona); > if (ret != 0) > @@ -682,6 +688,9 @@ static int arizona_runtime_suspend(struct device *dev) > } > } > break; > + case WM1831: > + case CS47L24: > + break; > default: > jd_active = arizona_is_jack_det_active(arizona); > if (jd_active < 0) > @@ -862,6 +871,8 @@ const struct of_device_id arizona_of_match[] = { > { .compatible = "wlf,wm8997", .data = (void *)WM8997 }, > { .compatible = "wlf,wm8998", .data = (void *)WM8998 }, > { .compatible = "wlf,wm1814", .data = (void *)WM1814 }, > + { .compatible = "wlf,wm1831", .data = (void *)WM1831 }, > + { .compatible = "cirrus,cs47l24", .data = (void *)CS47L24 }, > {}, > }; > EXPORT_SYMBOL_GPL(arizona_of_match); > @@ -919,6 +930,23 @@ static const struct mfd_cell wm5110_devs[] = { > }, > }; > > +static const char * const cs47l24_supplies[] = { > + "MICVDD", > + "CPVDD", > + "SPKVDD", > +}; > + > +static const struct mfd_cell cs47l24_devs[] = { > + { .name = "arizona-gpio" }, > + { .name = "arizona-haptics" }, > + { .name = "arizona-pwm" }, > + { > + .name = "cs47l24-codec", > + .parent_supplies = cs47l24_supplies, > + .num_parent_supplies = ARRAY_SIZE(cs47l24_supplies), > + }, > +}; > + > static const char * const wm8997_supplies[] = { > "MICVDD", > "DBVDD2", > @@ -963,7 +991,7 @@ static const struct mfd_cell wm8998_devs[] = { > int arizona_dev_init(struct arizona *arizona) > { > struct device *dev = arizona->dev; > - const char *type_name; > + const char *type_name = NULL; > unsigned int reg, val, mask; > int (*apply_patch)(struct arizona *) = NULL; > const struct mfd_cell *subdevs = NULL; > @@ -987,6 +1015,8 @@ int arizona_dev_init(struct arizona *arizona) > case WM8997: > case WM8998: > case WM1814: > + case WM1831: > + case CS47L24: > for (i = 0; i < ARRAY_SIZE(wm5102_core_supplies); i++) > arizona->core_supplies[i].supply > = wm5102_core_supplies[i]; > @@ -1001,11 +1031,18 @@ int arizona_dev_init(struct arizona *arizona) > /* Mark DCVDD as external, LDO1 driver will clear if internal */ > arizona->external_dcvdd = true; > > - ret = mfd_add_devices(arizona->dev, -1, early_devs, > - ARRAY_SIZE(early_devs), NULL, 0, NULL); > - if (ret != 0) { > - dev_err(dev, "Failed to add early children: %d\n", ret); > - return ret; > + switch (arizona->type) { > + case WM1831: > + case CS47L24: > + break; /* No LDO1 regulator */ > + default: > + ret = mfd_add_devices(arizona->dev, -1, early_devs, > + ARRAY_SIZE(early_devs), NULL, 0, NULL); > + if (ret != 0) { > + dev_err(dev, "Failed to add early children: %d\n", ret); > + return ret; > + } > + break; > } > > ret = devm_regulator_bulk_get(dev, arizona->num_core_supplies, > @@ -1069,6 +1106,7 @@ int arizona_dev_init(struct arizona *arizona) > case 0x5102: > case 0x5110: > case 0x6349: > + case 0x6363: > case 0x8997: > break; > default: > @@ -1167,6 +1205,30 @@ int arizona_dev_init(struct arizona *arizona) > n_subdevs = ARRAY_SIZE(wm5110_devs); > } > break; > + case 0x6363: > + if (IS_ENABLED(CONFIG_MFD_CS47L24)) { > + switch (arizona->type) { > + case CS47L24: > + type_name = "CS47L24"; > + break; > + > + case WM1831: > + type_name = "WM1831"; > + break; > + > + default: > + dev_warn(arizona->dev, > + "CS47L24 registered as %d\n", > + arizona->type); > + arizona->type = CS47L24; > + break; > + } > + > + apply_patch = cs47l24_patch; > + subdevs = cs47l24_devs; > + n_subdevs = ARRAY_SIZE(cs47l24_devs); > + } > + break; > case 0x8997: > if (IS_ENABLED(CONFIG_MFD_WM8997)) { > type_name = "WM8997"; > diff --git a/drivers/mfd/arizona-irq.c b/drivers/mfd/arizona-irq.c > index 3d425e9..682bc86 100644 > --- a/drivers/mfd/arizona-irq.c > +++ b/drivers/mfd/arizona-irq.c > @@ -30,11 +30,13 @@ static int arizona_map_irq(struct arizona *arizona, int irq) > { > int ret; > > - ret = regmap_irq_get_virq(arizona->aod_irq_chip, irq); > - if (ret < 0) > - ret = regmap_irq_get_virq(arizona->irq_chip, irq); > + if (arizona->aod_irq_chip) { > + ret = regmap_irq_get_virq(arizona->aod_irq_chip, irq); > + if (ret >= 0) > + return ret; > + } > > - return ret; > + return regmap_irq_get_virq(arizona->irq_chip, irq); > } > > int arizona_request_irq(struct arizona *arizona, int irq, char *name, > @@ -107,8 +109,8 @@ static irqreturn_t arizona_irq_thread(int irq, void *data) > do { > poll = false; > > - /* Always handle the AoD domain */ > - handle_nested_irq(irq_find_mapping(arizona->virq, 0)); > + if (arizona->aod_irq_chip) > + handle_nested_irq(irq_find_mapping(arizona->virq, 0)); > > /* > * Check if one of the main interrupts is asserted and only > @@ -219,6 +221,15 @@ int arizona_irq_init(struct arizona *arizona) > arizona->ctrlif_error = false; > break; > #endif > +#ifdef CONFIG_MFD_CS47L24 > + case WM1831: > + case CS47L24: > + aod = NULL; > + irq = &cs47l24_irq; > + > + arizona->ctrlif_error = false; > + break; > +#endif > #ifdef CONFIG_MFD_WM8997 > case WM8997: > aod = &wm8997_aod; > @@ -291,13 +302,16 @@ int arizona_irq_init(struct arizona *arizona) > goto err; > } > > - ret = regmap_add_irq_chip(arizona->regmap, > - irq_create_mapping(arizona->virq, 0), > - IRQF_ONESHOT, 0, aod, > - &arizona->aod_irq_chip); > - if (ret != 0) { > - dev_err(arizona->dev, "Failed to add AOD IRQs: %d\n", ret); > - goto err_domain; > + if (aod) { > + ret = regmap_add_irq_chip(arizona->regmap, > + irq_create_mapping(arizona->virq, 0), > + IRQF_ONESHOT, 0, aod, > + &arizona->aod_irq_chip); > + if (ret != 0) { > + dev_err(arizona->dev, > + "Failed to add AOD IRQs: %d\n", ret); > + goto err_domain; > + } > } > > ret = regmap_add_irq_chip(arizona->regmap, > diff --git a/drivers/mfd/arizona-spi.c b/drivers/mfd/arizona-spi.c > index 8cffb1c..b006771 100644 > --- a/drivers/mfd/arizona-spi.c > +++ b/drivers/mfd/arizona-spi.c > @@ -46,6 +46,11 @@ static int arizona_spi_probe(struct spi_device *spi) > if (IS_ENABLED(CONFIG_MFD_WM5110)) > regmap_config = &wm5110_spi_regmap; > break; > + case WM1831: > + case CS47L24: > + if (IS_ENABLED(CONFIG_MFD_CS47L24)) > + regmap_config = &cs47l24_spi_regmap; > + break; > default: > dev_err(&spi->dev, "Unknown device type %ld\n", type); > return -EINVAL; > @@ -89,6 +94,8 @@ static const struct spi_device_id arizona_spi_ids[] = { > { "wm5102", WM5102 }, > { "wm5110", WM5110 }, > { "wm8280", WM8280 }, > + { "wm1831", WM1831 }, > + { "cs47l24", CS47L24 }, > { }, > }; > MODULE_DEVICE_TABLE(spi, arizona_spi_ids); > diff --git a/drivers/mfd/arizona.h b/drivers/mfd/arizona.h > index 3af12e9..198e9ce 100644 > --- a/drivers/mfd/arizona.h > +++ b/drivers/mfd/arizona.h > @@ -25,6 +25,8 @@ extern const struct regmap_config wm5102_spi_regmap; > extern const struct regmap_config wm5110_i2c_regmap; > extern const struct regmap_config wm5110_spi_regmap; > > +extern const struct regmap_config cs47l24_spi_regmap; > + > extern const struct regmap_config wm8997_i2c_regmap; > > extern const struct regmap_config wm8998_i2c_regmap; > @@ -40,6 +42,8 @@ extern const struct regmap_irq_chip wm5110_aod; > extern const struct regmap_irq_chip wm5110_irq; > extern const struct regmap_irq_chip wm5110_revd_irq; > > +extern const struct regmap_irq_chip cs47l24_irq; > + > extern const struct regmap_irq_chip wm8997_aod; > extern const struct regmap_irq_chip wm8997_irq; > > diff --git a/drivers/mfd/cs47l24-tables.c b/drivers/mfd/cs47l24-tables.c > new file mode 100644 > index 0000000..8708006 > --- /dev/null > +++ b/drivers/mfd/cs47l24-tables.c > @@ -0,0 +1,1629 @@ > +/* > + * Data tables for CS47L24 codec > + * > + * Copyright 2015 Cirrus Logic, Inc. > + * > + * Author: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.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/module.h> > + > +#include <linux/mfd/arizona/core.h> > +#include <linux/mfd/arizona/registers.h> > +#include <linux/device.h> > + > +#include "arizona.h" > + > +#define CS47L24_NUM_ISR 5 > + > +static const struct reg_sequence cs47l24_reva_patch[] = { > + { 0x80, 0x3 }, > + { 0x27C, 0x0010 }, > + { 0x221, 0x0070 }, > + { 0x80, 0x0 }, > +}; > + > +int cs47l24_patch(struct arizona *arizona) > +{ > + return regmap_register_patch(arizona->regmap, > + cs47l24_reva_patch, > + ARRAY_SIZE(cs47l24_reva_patch)); > +} > +EXPORT_SYMBOL_GPL(cs47l24_patch); > + > +static const struct regmap_irq cs47l24_irqs[ARIZONA_NUM_IRQ] = { > + [ARIZONA_IRQ_GP2] = { .reg_offset = 0, .mask = ARIZONA_GP2_EINT1 }, > + [ARIZONA_IRQ_GP1] = { .reg_offset = 0, .mask = ARIZONA_GP1_EINT1 }, > + > + [ARIZONA_IRQ_DSP3_RAM_RDY] = { > + .reg_offset = 1, .mask = ARIZONA_DSP3_RAM_RDY_EINT1 > + }, > + [ARIZONA_IRQ_DSP2_RAM_RDY] = { > + .reg_offset = 1, .mask = ARIZONA_DSP2_RAM_RDY_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ8] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ8_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ7] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ7_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ6] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ6_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ5] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ5_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ4] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ4_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ3] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ3_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ2] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ2_EINT1 > + }, > + [ARIZONA_IRQ_DSP_IRQ1] = { > + .reg_offset = 1, .mask = ARIZONA_DSP_IRQ1_EINT1 > + }, > + > + [ARIZONA_IRQ_SPK_OVERHEAT_WARN] = { > + .reg_offset = 2, .mask = ARIZONA_SPK_OVERHEAT_WARN_EINT1 > + }, > + [ARIZONA_IRQ_SPK_OVERHEAT] = { > + .reg_offset = 2, .mask = ARIZONA_SPK_OVERHEAT_EINT1 > + }, > + [ARIZONA_IRQ_WSEQ_DONE] = { > + .reg_offset = 2, .mask = ARIZONA_WSEQ_DONE_EINT1 > + }, > + [ARIZONA_IRQ_DRC2_SIG_DET] = { > + .reg_offset = 2, .mask = ARIZONA_DRC2_SIG_DET_EINT1 > + }, > + [ARIZONA_IRQ_DRC1_SIG_DET] = { > + .reg_offset = 2, .mask = ARIZONA_DRC1_SIG_DET_EINT1 > + }, > + [ARIZONA_IRQ_ASRC2_LOCK] = { > + .reg_offset = 2, .mask = ARIZONA_ASRC2_LOCK_EINT1 > + }, > + [ARIZONA_IRQ_ASRC1_LOCK] = { > + .reg_offset = 2, .mask = ARIZONA_ASRC1_LOCK_EINT1 > + }, > + [ARIZONA_IRQ_UNDERCLOCKED] = { > + .reg_offset = 2, .mask = ARIZONA_UNDERCLOCKED_EINT1 > + }, > + [ARIZONA_IRQ_OVERCLOCKED] = { > + .reg_offset = 2, .mask = ARIZONA_OVERCLOCKED_EINT1 > + }, > + [ARIZONA_IRQ_FLL2_LOCK] = { > + .reg_offset = 2, .mask = ARIZONA_FLL2_LOCK_EINT1 > + }, > + [ARIZONA_IRQ_FLL1_LOCK] = { > + .reg_offset = 2, .mask = ARIZONA_FLL1_LOCK_EINT1 > + }, > + [ARIZONA_IRQ_CLKGEN_ERR] = { > + .reg_offset = 2, .mask = ARIZONA_CLKGEN_ERR_EINT1 > + }, > + [ARIZONA_IRQ_CLKGEN_ERR_ASYNC] = { > + .reg_offset = 2, .mask = ARIZONA_CLKGEN_ERR_ASYNC_EINT1 > + }, > + > + [ARIZONA_IRQ_CTRLIF_ERR] = { > + .reg_offset = 3, .mask = ARIZONA_V2_CTRLIF_ERR_EINT1 > + }, > + [ARIZONA_IRQ_MIXER_DROPPED_SAMPLES] = { > + .reg_offset = 3, .mask = ARIZONA_V2_MIXER_DROPPED_SAMPLE_EINT1 > + }, > + [ARIZONA_IRQ_ASYNC_CLK_ENA_LOW] = { > + .reg_offset = 3, .mask = ARIZONA_V2_ASYNC_CLK_ENA_LOW_EINT1 > + }, > + [ARIZONA_IRQ_SYSCLK_ENA_LOW] = { > + .reg_offset = 3, .mask = ARIZONA_V2_SYSCLK_ENA_LOW_EINT1 > + }, > + [ARIZONA_IRQ_ISRC1_CFG_ERR] = { > + .reg_offset = 3, .mask = ARIZONA_V2_ISRC1_CFG_ERR_EINT1 > + }, > + [ARIZONA_IRQ_ISRC2_CFG_ERR] = { > + .reg_offset = 3, .mask = ARIZONA_V2_ISRC2_CFG_ERR_EINT1 > + }, > + [ARIZONA_IRQ_ISRC3_CFG_ERR] = { > + .reg_offset = 3, .mask = ARIZONA_V2_ISRC3_CFG_ERR_EINT1 > + }, > + [ARIZONA_IRQ_HP1R_DONE] = { > + .reg_offset = 3, .mask = ARIZONA_HP1R_DONE_EINT1 > + }, > + [ARIZONA_IRQ_HP1L_DONE] = { > + .reg_offset = 3, .mask = ARIZONA_HP1L_DONE_EINT1 > + }, > + > + [ARIZONA_IRQ_BOOT_DONE] = { > + .reg_offset = 4, .mask = ARIZONA_BOOT_DONE_EINT1 > + }, > + [ARIZONA_IRQ_ASRC_CFG_ERR] = { > + .reg_offset = 4, .mask = ARIZONA_V2_ASRC_CFG_ERR_EINT1 > + }, > + [ARIZONA_IRQ_FLL2_CLOCK_OK] = { > + .reg_offset = 4, .mask = ARIZONA_FLL2_CLOCK_OK_EINT1 > + }, > + [ARIZONA_IRQ_FLL1_CLOCK_OK] = { > + .reg_offset = 4, .mask = ARIZONA_FLL1_CLOCK_OK_EINT1 > + }, > + > + [ARIZONA_IRQ_DSP_SHARED_WR_COLL] = { > + .reg_offset = 5, .mask = ARIZONA_DSP_SHARED_WR_COLL_EINT1 > + }, > + [ARIZONA_IRQ_SPK_SHUTDOWN] = { > + .reg_offset = 5, .mask = ARIZONA_SPK_SHUTDOWN_EINT1 > + }, > + [ARIZONA_IRQ_SPK1R_SHORT] = { > + .reg_offset = 5, .mask = ARIZONA_SPK1R_SHORT_EINT1 > + }, > + [ARIZONA_IRQ_SPK1L_SHORT] = { > + .reg_offset = 5, .mask = ARIZONA_SPK1L_SHORT_EINT1 > + }, > + [ARIZONA_IRQ_HP1R_SC_POS] = { > + .reg_offset = 5, .mask = ARIZONA_HP1R_SC_POS_EINT1 > + }, > + [ARIZONA_IRQ_HP1L_SC_POS] = { > + .reg_offset = 5, .mask = ARIZONA_HP1L_SC_POS_EINT1 > + }, > +}; > + > +const struct regmap_irq_chip cs47l24_irq = { > + .name = "cs47l24 IRQ", > + .status_base = ARIZONA_INTERRUPT_STATUS_1, > + .mask_base = ARIZONA_INTERRUPT_STATUS_1_MASK, > + .ack_base = ARIZONA_INTERRUPT_STATUS_1, > + .num_regs = 6, > + .irqs = cs47l24_irqs, > + .num_irqs = ARRAY_SIZE(cs47l24_irqs), > +}; > +EXPORT_SYMBOL_GPL(cs47l24_irq); > + > +static const struct reg_default cs47l24_reg_default[] = { > + { 0x00000008, 0x0019 }, /* R8 - Ctrl IF SPI CFG 1 */ > + { 0x00000020, 0x0000 }, /* R32 - Tone Generator 1 */ > + { 0x00000021, 0x1000 }, /* R33 - Tone Generator 2 */ > + { 0x00000022, 0x0000 }, /* R34 - Tone Generator 3 */ > + { 0x00000023, 0x1000 }, /* R35 - Tone Generator 4 */ > + { 0x00000024, 0x0000 }, /* R36 - Tone Generator 5 */ > + { 0x00000030, 0x0000 }, /* R48 - PWM Drive 1 */ > + { 0x00000031, 0x0100 }, /* R49 - PWM Drive 2 */ > + { 0x00000032, 0x0100 }, /* R50 - PWM Drive 3 */ > + { 0x00000041, 0x0000 }, /* R65 - Sequence control */ > + { 0x00000061, 0x01FF }, /* R97 - Sample Rate Sequence Select 1 */ > + { 0x00000062, 0x01FF }, /* R98 - Sample Rate Sequence Select 2 */ > + { 0x00000063, 0x01FF }, /* R99 - Sample Rate Sequence Select 3 */ > + { 0x00000064, 0x01FF }, /* R100 - Sample Rate Sequence Select 4 */ > + { 0x00000070, 0x0000 }, /* R112 - Comfort Noise Generator */ > + { 0x00000090, 0x0000 }, /* R144 - Haptics Control 1 */ > + { 0x00000091, 0x7FFF }, /* R145 - Haptics Control 2 */ > + { 0x00000092, 0x0000 }, /* R146 - Haptics phase 1 intensity */ > + { 0x00000093, 0x0000 }, /* R147 - Haptics phase 1 duration */ > + { 0x00000094, 0x0000 }, /* R148 - Haptics phase 2 intensity */ > + { 0x00000095, 0x0000 }, /* R149 - Haptics phase 2 duration */ > + { 0x00000096, 0x0000 }, /* R150 - Haptics phase 3 intensity */ > + { 0x00000097, 0x0000 }, /* R151 - Haptics phase 3 duration */ > + { 0x00000100, 0x0002 }, /* R256 - Clock 32k 1 */ > + { 0x00000101, 0x0504 }, /* R257 - System Clock 1 */ > + { 0x00000102, 0x0011 }, /* R258 - Sample rate 1 */ > + { 0x00000103, 0x0011 }, /* R259 - Sample rate 2 */ > + { 0x00000104, 0x0011 }, /* R260 - Sample rate 3 */ > + { 0x00000112, 0x0305 }, /* R274 - Async clock 1 */ > + { 0x00000113, 0x0011 }, /* R275 - Async sample rate 1 */ > + { 0x00000114, 0x0011 }, /* R276 - Async sample rate 2 */ > + { 0x00000149, 0x0000 }, /* R329 - Output system clock */ > + { 0x0000014A, 0x0000 }, /* R330 - Output async clock */ > + { 0x00000152, 0x0000 }, /* R338 - Rate Estimator 1 */ > + { 0x00000153, 0x0000 }, /* R339 - Rate Estimator 2 */ > + { 0x00000154, 0x0000 }, /* R340 - Rate Estimator 3 */ > + { 0x00000155, 0x0000 }, /* R341 - Rate Estimator 4 */ > + { 0x00000156, 0x0000 }, /* R342 - Rate Estimator 5 */ > + { 0x00000171, 0x0002 }, /* R369 - FLL1 Control 1 */ > + { 0x00000172, 0x0008 }, /* R370 - FLL1 Control 2 */ > + { 0x00000173, 0x0018 }, /* R371 - FLL1 Control 3 */ > + { 0x00000174, 0x007D }, /* R372 - FLL1 Control 4 */ > + { 0x00000175, 0x0006 }, /* R373 - FLL1 Control 5 */ > + { 0x00000176, 0x0000 }, /* R374 - FLL1 Control 6 */ > + { 0x00000177, 0x0281 }, /* R375 - FLL1 Loop Filter Test 1 */ > + { 0x00000178, 0x0000 }, /* R376 - FLL1 NCO Test 0 */ > + { 0x00000179, 0x0000 }, /* R376 - FLL1 Control 7 */ > + { 0x00000181, 0x0000 }, /* R385 - FLL1 Synchroniser 1 */ > + { 0x00000182, 0x0000 }, /* R386 - FLL1 Synchroniser 2 */ > + { 0x00000183, 0x0000 }, /* R387 - FLL1 Synchroniser 3 */ > + { 0x00000184, 0x0000 }, /* R388 - FLL1 Synchroniser 4 */ > + { 0x00000185, 0x0000 }, /* R389 - FLL1 Synchroniser 5 */ > + { 0x00000186, 0x0000 }, /* R390 - FLL1 Synchroniser 6 */ > + { 0x00000187, 0x0001 }, /* R390 - FLL1 Synchroniser 7 */ > + { 0x00000189, 0x0000 }, /* R393 - FLL1 Spread Spectrum */ > + { 0x0000018A, 0x000C }, /* R394 - FLL1 GPIO Clock */ > + { 0x00000191, 0x0002 }, /* R401 - FLL2 Control 1 */ > + { 0x00000192, 0x0008 }, /* R402 - FLL2 Control 2 */ > + { 0x00000193, 0x0018 }, /* R403 - FLL2 Control 3 */ > + { 0x00000194, 0x007D }, /* R404 - FLL2 Control 4 */ > + { 0x00000195, 0x000C }, /* R405 - FLL2 Control 5 */ > + { 0x00000196, 0x0000 }, /* R406 - FLL2 Control 6 */ > + { 0x00000197, 0x0000 }, /* R407 - FLL2 Loop Filter Test 1 */ > + { 0x00000198, 0x0000 }, /* R408 - FLL2 NCO Test 0 */ > + { 0x00000199, 0x0000 }, /* R408 - FLL2 Control 7 */ > + { 0x000001A1, 0x0000 }, /* R417 - FLL2 Synchroniser 1 */ > + { 0x000001A2, 0x0000 }, /* R418 - FLL2 Synchroniser 2 */ > + { 0x000001A3, 0x0000 }, /* R419 - FLL2 Synchroniser 3 */ > + { 0x000001A4, 0x0000 }, /* R420 - FLL2 Synchroniser 4 */ > + { 0x000001A5, 0x0000 }, /* R421 - FLL2 Synchroniser 5 */ > + { 0x000001A6, 0x0000 }, /* R422 - FLL2 Synchroniser 6 */ > + { 0x000001A7, 0x0001 }, /* R422 - FLL2 Synchroniser 7 */ > + { 0x000001A9, 0x0000 }, /* R425 - FLL2 Spread Spectrum */ > + { 0x000001AA, 0x000C }, /* R426 - FLL2 GPIO Clock */ > + { 0x00000218, 0x00E6 }, /* R536 - Mic Bias Ctrl 1 */ > + { 0x00000219, 0x00E6 }, /* R537 - Mic Bias Ctrl 2 */ > + { 0x00000300, 0x0000 }, /* R768 - Input Enables */ > + { 0x00000308, 0x0000 }, /* R776 - Input Rate */ > + { 0x00000309, 0x0022 }, /* R777 - Input Volume Ramp */ > + { 0x0000030C, 0x0002 }, /* R780 - HPF Control */ > + { 0x00000310, 0x2000 }, /* R784 - IN1L Control */ > + { 0x00000311, 0x0180 }, /* R785 - ADC Digital Volume 1L */ > + { 0x00000312, 0x0000 }, /* R786 - DMIC1L Control */ > + { 0x00000314, 0x0000 }, /* R788 - IN1R Control */ > + { 0x00000315, 0x0180 }, /* R789 - ADC Digital Volume 1R */ > + { 0x00000316, 0x0000 }, /* R790 - DMIC1R Control */ > + { 0x00000318, 0x2000 }, /* R792 - IN2L Control */ > + { 0x00000319, 0x0180 }, /* R793 - ADC Digital Volume 2L */ > + { 0x0000031A, 0x0000 }, /* R794 - DMIC2L Control */ > + { 0x0000031C, 0x0000 }, /* R796 - IN2R Control */ > + { 0x0000031D, 0x0180 }, /* R797 - ADC Digital Volume 2R */ > + { 0x0000031E, 0x0000 }, /* R798 - DMIC2R Control */ > + { 0x00000400, 0x0000 }, /* R1024 - Output Enables 1 */ > + { 0x00000408, 0x0000 }, /* R1032 - Output Rate 1 */ > + { 0x00000409, 0x0022 }, /* R1033 - Output Volume Ramp */ > + { 0x00000410, 0x0080 }, /* R1040 - Output Path Config 1L */ > + { 0x00000411, 0x0180 }, /* R1041 - DAC Digital Volume 1L */ > + { 0x00000412, 0x0081 }, /* R1042 - DAC Volume Limit 1L */ > + { 0x00000413, 0x0001 }, /* R1043 - Noise Gate Select 1L */ > + { 0x00000415, 0x0180 }, /* R1045 - DAC Digital Volume 1R */ > + { 0x00000416, 0x0081 }, /* R1046 - DAC Volume Limit 1R */ > + { 0x00000417, 0x0002 }, /* R1047 - Noise Gate Select 1R */ > + { 0x00000429, 0x0180 }, /* R1065 - DAC Digital Volume 4L */ > + { 0x0000042A, 0x0081 }, /* R1066 - Out Volume 4L */ > + { 0x0000042B, 0x0040 }, /* R1067 - Noise Gate Select 4L */ > + { 0x00000450, 0x0000 }, /* R1104 - DAC AEC Control 1 */ > + { 0x00000458, 0x0000 }, /* R1112 - Noise Gate Control */ > + { 0x000004A0, 0x3480 }, /* R1184 - HP1 Short Circuit Ctrl */ > + { 0x00000500, 0x000C }, /* R1280 - AIF1 BCLK Ctrl */ > + { 0x00000501, 0x0008 }, /* R1281 - AIF1 Tx Pin Ctrl */ > + { 0x00000502, 0x0000 }, /* R1282 - AIF1 Rx Pin Ctrl */ > + { 0x00000503, 0x0000 }, /* R1283 - AIF1 Rate Ctrl */ > + { 0x00000504, 0x0000 }, /* R1284 - AIF1 Format */ > + { 0x00000506, 0x0040 }, /* R1286 - AIF1 Rx BCLK Rate */ > + { 0x00000507, 0x1818 }, /* R1287 - AIF1 Frame Ctrl 1 */ > + { 0x00000508, 0x1818 }, /* R1288 - AIF1 Frame Ctrl 2 */ > + { 0x00000509, 0x0000 }, /* R1289 - AIF1 Frame Ctrl 3 */ > + { 0x0000050A, 0x0001 }, /* R1290 - AIF1 Frame Ctrl 4 */ > + { 0x0000050B, 0x0002 }, /* R1291 - AIF1 Frame Ctrl 5 */ > + { 0x0000050C, 0x0003 }, /* R1292 - AIF1 Frame Ctrl 6 */ > + { 0x0000050D, 0x0004 }, /* R1293 - AIF1 Frame Ctrl 7 */ > + { 0x0000050E, 0x0005 }, /* R1294 - AIF1 Frame Ctrl 8 */ > + { 0x0000050F, 0x0006 }, /* R1295 - AIF1 Frame Ctrl 9 */ > + { 0x00000510, 0x0007 }, /* R1296 - AIF1 Frame Ctrl 10 */ > + { 0x00000511, 0x0000 }, /* R1297 - AIF1 Frame Ctrl 11 */ > + { 0x00000512, 0x0001 }, /* R1298 - AIF1 Frame Ctrl 12 */ > + { 0x00000513, 0x0002 }, /* R1299 - AIF1 Frame Ctrl 13 */ > + { 0x00000514, 0x0003 }, /* R1300 - AIF1 Frame Ctrl 14 */ > + { 0x00000515, 0x0004 }, /* R1301 - AIF1 Frame Ctrl 15 */ > + { 0x00000516, 0x0005 }, /* R1302 - AIF1 Frame Ctrl 16 */ > + { 0x00000517, 0x0006 }, /* R1303 - AIF1 Frame Ctrl 17 */ > + { 0x00000518, 0x0007 }, /* R1304 - AIF1 Frame Ctrl 18 */ > + { 0x00000519, 0x0000 }, /* R1305 - AIF1 Tx Enables */ > + { 0x0000051A, 0x0000 }, /* R1306 - AIF1 Rx Enables */ > + { 0x00000540, 0x000C }, /* R1344 - AIF2 BCLK Ctrl */ > + { 0x00000541, 0x0008 }, /* R1345 - AIF2 Tx Pin Ctrl */ > + { 0x00000542, 0x0000 }, /* R1346 - AIF2 Rx Pin Ctrl */ > + { 0x00000543, 0x0000 }, /* R1347 - AIF2 Rate Ctrl */ > + { 0x00000544, 0x0000 }, /* R1348 - AIF2 Format */ > + { 0x00000546, 0x0040 }, /* R1350 - AIF2 Rx BCLK Rate */ > + { 0x00000547, 0x1818 }, /* R1351 - AIF2 Frame Ctrl 1 */ > + { 0x00000548, 0x1818 }, /* R1352 - AIF2 Frame Ctrl 2 */ > + { 0x00000549, 0x0000 }, /* R1353 - AIF2 Frame Ctrl 3 */ > + { 0x0000054A, 0x0001 }, /* R1354 - AIF2 Frame Ctrl 4 */ > + { 0x0000054B, 0x0002 }, /* R1355 - AIF2 Frame Ctrl 5 */ > + { 0x0000054C, 0x0003 }, /* R1356 - AIF2 Frame Ctrl 6 */ > + { 0x0000054D, 0x0004 }, /* R1357 - AIF2 Frame Ctrl 7 */ > + { 0x0000054E, 0x0005 }, /* R1358 - AIF2 Frame Ctrl 8 */ > + { 0x00000551, 0x0000 }, /* R1361 - AIF2 Frame Ctrl 11 */ > + { 0x00000552, 0x0001 }, /* R1362 - AIF2 Frame Ctrl 12 */ > + { 0x00000553, 0x0002 }, /* R1363 - AIF2 Frame Ctrl 13 */ > + { 0x00000554, 0x0003 }, /* R1364 - AIF2 Frame Ctrl 14 */ > + { 0x00000555, 0x0004 }, /* R1365 - AIF2 Frame Ctrl 15 */ > + { 0x00000556, 0x0005 }, /* R1366 - AIF2 Frame Ctrl 16 */ > + { 0x00000559, 0x0000 }, /* R1369 - AIF2 Tx Enables */ > + { 0x0000055A, 0x0000 }, /* R1370 - AIF2 Rx Enables */ > + { 0x00000580, 0x000C }, /* R1408 - AIF3 BCLK Ctrl */ > + { 0x00000581, 0x0008 }, /* R1409 - AIF3 Tx Pin Ctrl */ > + { 0x00000582, 0x0000 }, /* R1410 - AIF3 Rx Pin Ctrl */ > + { 0x00000583, 0x0000 }, /* R1411 - AIF3 Rate Ctrl */ > + { 0x00000584, 0x0000 }, /* R1412 - AIF3 Format */ > + { 0x00000586, 0x0040 }, /* R1414 - AIF3 Rx BCLK Rate */ > + { 0x00000587, 0x1818 }, /* R1415 - AIF3 Frame Ctrl 1 */ > + { 0x00000588, 0x1818 }, /* R1416 - AIF3 Frame Ctrl 2 */ > + { 0x00000589, 0x0000 }, /* R1417 - AIF3 Frame Ctrl 3 */ > + { 0x0000058A, 0x0001 }, /* R1418 - AIF3 Frame Ctrl 4 */ > + { 0x00000591, 0x0000 }, /* R1425 - AIF3 Frame Ctrl 11 */ > + { 0x00000592, 0x0001 }, /* R1426 - AIF3 Frame Ctrl 12 */ > + { 0x00000599, 0x0000 }, /* R1433 - AIF3 Tx Enables */ > + { 0x0000059A, 0x0000 }, /* R1434 - AIF3 Rx Enables */ > + { 0x00000640, 0x0000 }, /* R1600 - PWM1MIX Input 1 Source */ > + { 0x00000641, 0x0080 }, /* R1601 - PWM1MIX Input 1 Volume */ > + { 0x00000642, 0x0000 }, /* R1602 - PWM1MIX Input 2 Source */ > + { 0x00000643, 0x0080 }, /* R1603 - PWM1MIX Input 2 Volume */ > + { 0x00000644, 0x0000 }, /* R1604 - PWM1MIX Input 3 Source */ > + { 0x00000645, 0x0080 }, /* R1605 - PWM1MIX Input 3 Volume */ > + { 0x00000646, 0x0000 }, /* R1606 - PWM1MIX Input 4 Source */ > + { 0x00000647, 0x0080 }, /* R1607 - PWM1MIX Input 4 Volume */ > + { 0x00000648, 0x0000 }, /* R1608 - PWM2MIX Input 1 Source */ > + { 0x00000649, 0x0080 }, /* R1609 - PWM2MIX Input 1 Volume */ > + { 0x0000064A, 0x0000 }, /* R1610 - PWM2MIX Input 2 Source */ > + { 0x0000064B, 0x0080 }, /* R1611 - PWM2MIX Input 2 Volume */ > + { 0x0000064C, 0x0000 }, /* R1612 - PWM2MIX Input 3 Source */ > + { 0x0000064D, 0x0080 }, /* R1613 - PWM2MIX Input 3 Volume */ > + { 0x0000064E, 0x0000 }, /* R1614 - PWM2MIX Input 4 Source */ > + { 0x0000064F, 0x0080 }, /* R1615 - PWM2MIX Input 4 Volume */ > + { 0x00000680, 0x0000 }, /* R1664 - OUT1LMIX Input 1 Source */ > + { 0x00000681, 0x0080 }, /* R1665 - OUT1LMIX Input 1 Volume */ > + { 0x00000682, 0x0000 }, /* R1666 - OUT1LMIX Input 2 Source */ > + { 0x00000683, 0x0080 }, /* R1667 - OUT1LMIX Input 2 Volume */ > + { 0x00000684, 0x0000 }, /* R1668 - OUT1LMIX Input 3 Source */ > + { 0x00000685, 0x0080 }, /* R1669 - OUT1LMIX Input 3 Volume */ > + { 0x00000686, 0x0000 }, /* R1670 - OUT1LMIX Input 4 Source */ > + { 0x00000687, 0x0080 }, /* R1671 - OUT1LMIX Input 4 Volume */ > + { 0x00000688, 0x0000 }, /* R1672 - OUT1RMIX Input 1 Source */ > + { 0x00000689, 0x0080 }, /* R1673 - OUT1RMIX Input 1 Volume */ > + { 0x0000068A, 0x0000 }, /* R1674 - OUT1RMIX Input 2 Source */ > + { 0x0000068B, 0x0080 }, /* R1675 - OUT1RMIX Input 2 Volume */ > + { 0x0000068C, 0x0000 }, /* R1676 - OUT1RMIX Input 3 Source */ > + { 0x0000068D, 0x0080 }, /* R1677 - OUT1RMIX Input 3 Volume */ > + { 0x0000068E, 0x0000 }, /* R1678 - OUT1RMIX Input 4 Source */ > + { 0x0000068F, 0x0080 }, /* R1679 - OUT1RMIX Input 4 Volume */ > + { 0x000006B0, 0x0000 }, /* R1712 - OUT4LMIX Input 1 Source */ > + { 0x000006B1, 0x0080 }, /* R1713 - OUT4LMIX Input 1 Volume */ > + { 0x000006B2, 0x0000 }, /* R1714 - OUT4LMIX Input 2 Source */ > + { 0x000006B3, 0x0080 }, /* R1715 - OUT4LMIX Input 2 Volume */ > + { 0x000006B4, 0x0000 }, /* R1716 - OUT4LMIX Input 3 Source */ > + { 0x000006B5, 0x0080 }, /* R1717 - OUT4LMIX Input 3 Volume */ > + { 0x000006B6, 0x0000 }, /* R1718 - OUT4LMIX Input 4 Source */ > + { 0x000006B7, 0x0080 }, /* R1719 - OUT4LMIX Input 4 Volume */ > + { 0x00000700, 0x0000 }, /* R1792 - AIF1TX1MIX Input 1 Source */ > + { 0x00000701, 0x0080 }, /* R1793 - AIF1TX1MIX Input 1 Volume */ > + { 0x00000702, 0x0000 }, /* R1794 - AIF1TX1MIX Input 2 Source */ > + { 0x00000703, 0x0080 }, /* R1795 - AIF1TX1MIX Input 2 Volume */ > + { 0x00000704, 0x0000 }, /* R1796 - AIF1TX1MIX Input 3 Source */ > + { 0x00000705, 0x0080 }, /* R1797 - AIF1TX1MIX Input 3 Volume */ > + { 0x00000706, 0x0000 }, /* R1798 - AIF1TX1MIX Input 4 Source */ > + { 0x00000707, 0x0080 }, /* R1799 - AIF1TX1MIX Input 4 Volume */ > + { 0x00000708, 0x0000 }, /* R1800 - AIF1TX2MIX Input 1 Source */ > + { 0x00000709, 0x0080 }, /* R1801 - AIF1TX2MIX Input 1 Volume */ > + { 0x0000070A, 0x0000 }, /* R1802 - AIF1TX2MIX Input 2 Source */ > + { 0x0000070B, 0x0080 }, /* R1803 - AIF1TX2MIX Input 2 Volume */ > + { 0x0000070C, 0x0000 }, /* R1804 - AIF1TX2MIX Input 3 Source */ > + { 0x0000070D, 0x0080 }, /* R1805 - AIF1TX2MIX Input 3 Volume */ > + { 0x0000070E, 0x0000 }, /* R1806 - AIF1TX2MIX Input 4 Source */ > + { 0x0000070F, 0x0080 }, /* R1807 - AIF1TX2MIX Input 4 Volume */ > + { 0x00000710, 0x0000 }, /* R1808 - AIF1TX3MIX Input 1 Source */ > + { 0x00000711, 0x0080 }, /* R1809 - AIF1TX3MIX Input 1 Volume */ > + { 0x00000712, 0x0000 }, /* R1810 - AIF1TX3MIX Input 2 Source */ > + { 0x00000713, 0x0080 }, /* R1811 - AIF1TX3MIX Input 2 Volume */ > + { 0x00000714, 0x0000 }, /* R1812 - AIF1TX3MIX Input 3 Source */ > + { 0x00000715, 0x0080 }, /* R1813 - AIF1TX3MIX Input 3 Volume */ > + { 0x00000716, 0x0000 }, /* R1814 - AIF1TX3MIX Input 4 Source */ > + { 0x00000717, 0x0080 }, /* R1815 - AIF1TX3MIX Input 4 Volume */ > + { 0x00000718, 0x0000 }, /* R1816 - AIF1TX4MIX Input 1 Source */ > + { 0x00000719, 0x0080 }, /* R1817 - AIF1TX4MIX Input 1 Volume */ > + { 0x0000071A, 0x0000 }, /* R1818 - AIF1TX4MIX Input 2 Source */ > + { 0x0000071B, 0x0080 }, /* R1819 - AIF1TX4MIX Input 2 Volume */ > + { 0x0000071C, 0x0000 }, /* R1820 - AIF1TX4MIX Input 3 Source */ > + { 0x0000071D, 0x0080 }, /* R1821 - AIF1TX4MIX Input 3 Volume */ > + { 0x0000071E, 0x0000 }, /* R1822 - AIF1TX4MIX Input 4 Source */ > + { 0x0000071F, 0x0080 }, /* R1823 - AIF1TX4MIX Input 4 Volume */ > + { 0x00000720, 0x0000 }, /* R1824 - AIF1TX5MIX Input 1 Source */ > + { 0x00000721, 0x0080 }, /* R1825 - AIF1TX5MIX Input 1 Volume */ > + { 0x00000722, 0x0000 }, /* R1826 - AIF1TX5MIX Input 2 Source */ > + { 0x00000723, 0x0080 }, /* R1827 - AIF1TX5MIX Input 2 Volume */ > + { 0x00000724, 0x0000 }, /* R1828 - AIF1TX5MIX Input 3 Source */ > + { 0x00000725, 0x0080 }, /* R1829 - AIF1TX5MIX Input 3 Volume */ > + { 0x00000726, 0x0000 }, /* R1830 - AIF1TX5MIX Input 4 Source */ > + { 0x00000727, 0x0080 }, /* R1831 - AIF1TX5MIX Input 4 Volume */ > + { 0x00000728, 0x0000 }, /* R1832 - AIF1TX6MIX Input 1 Source */ > + { 0x00000729, 0x0080 }, /* R1833 - AIF1TX6MIX Input 1 Volume */ > + { 0x0000072A, 0x0000 }, /* R1834 - AIF1TX6MIX Input 2 Source */ > + { 0x0000072B, 0x0080 }, /* R1835 - AIF1TX6MIX Input 2 Volume */ > + { 0x0000072C, 0x0000 }, /* R1836 - AIF1TX6MIX Input 3 Source */ > + { 0x0000072D, 0x0080 }, /* R1837 - AIF1TX6MIX Input 3 Volume */ > + { 0x0000072E, 0x0000 }, /* R1838 - AIF1TX6MIX Input 4 Source */ > + { 0x0000072F, 0x0080 }, /* R1839 - AIF1TX6MIX Input 4 Volume */ > + { 0x00000730, 0x0000 }, /* R1840 - AIF1TX7MIX Input 1 Source */ > + { 0x00000731, 0x0080 }, /* R1841 - AIF1TX7MIX Input 1 Volume */ > + { 0x00000732, 0x0000 }, /* R1842 - AIF1TX7MIX Input 2 Source */ > + { 0x00000733, 0x0080 }, /* R1843 - AIF1TX7MIX Input 2 Volume */ > + { 0x00000734, 0x0000 }, /* R1844 - AIF1TX7MIX Input 3 Source */ > + { 0x00000735, 0x0080 }, /* R1845 - AIF1TX7MIX Input 3 Volume */ > + { 0x00000736, 0x0000 }, /* R1846 - AIF1TX7MIX Input 4 Source */ > + { 0x00000737, 0x0080 }, /* R1847 - AIF1TX7MIX Input 4 Volume */ > + { 0x00000738, 0x0000 }, /* R1848 - AIF1TX8MIX Input 1 Source */ > + { 0x00000739, 0x0080 }, /* R1849 - AIF1TX8MIX Input 1 Volume */ > + { 0x0000073A, 0x0000 }, /* R1850 - AIF1TX8MIX Input 2 Source */ > + { 0x0000073B, 0x0080 }, /* R1851 - AIF1TX8MIX Input 2 Volume */ > + { 0x0000073C, 0x0000 }, /* R1852 - AIF1TX8MIX Input 3 Source */ > + { 0x0000073D, 0x0080 }, /* R1853 - AIF1TX8MIX Input 3 Volume */ > + { 0x0000073E, 0x0000 }, /* R1854 - AIF1TX8MIX Input 4 Source */ > + { 0x0000073F, 0x0080 }, /* R1855 - AIF1TX8MIX Input 4 Volume */ > + { 0x00000740, 0x0000 }, /* R1856 - AIF2TX1MIX Input 1 Source */ > + { 0x00000741, 0x0080 }, /* R1857 - AIF2TX1MIX Input 1 Volume */ > + { 0x00000742, 0x0000 }, /* R1858 - AIF2TX1MIX Input 2 Source */ > + { 0x00000743, 0x0080 }, /* R1859 - AIF2TX1MIX Input 2 Volume */ > + { 0x00000744, 0x0000 }, /* R1860 - AIF2TX1MIX Input 3 Source */ > + { 0x00000745, 0x0080 }, /* R1861 - AIF2TX1MIX Input 3 Volume */ > + { 0x00000746, 0x0000 }, /* R1862 - AIF2TX1MIX Input 4 Source */ > + { 0x00000747, 0x0080 }, /* R1863 - AIF2TX1MIX Input 4 Volume */ > + { 0x00000748, 0x0000 }, /* R1864 - AIF2TX2MIX Input 1 Source */ > + { 0x00000749, 0x0080 }, /* R1865 - AIF2TX2MIX Input 1 Volume */ > + { 0x0000074A, 0x0000 }, /* R1866 - AIF2TX2MIX Input 2 Source */ > + { 0x0000074B, 0x0080 }, /* R1867 - AIF2TX2MIX Input 2 Volume */ > + { 0x0000074C, 0x0000 }, /* R1868 - AIF2TX2MIX Input 3 Source */ > + { 0x0000074D, 0x0080 }, /* R1869 - AIF2TX2MIX Input 3 Volume */ > + { 0x0000074E, 0x0000 }, /* R1870 - AIF2TX2MIX Input 4 Source */ > + { 0x0000074F, 0x0080 }, /* R1871 - AIF2TX2MIX Input 4 Volume */ > + { 0x00000750, 0x0000 }, /* R1872 - AIF2TX3MIX Input 1 Source */ > + { 0x00000751, 0x0080 }, /* R1873 - AIF2TX3MIX Input 1 Volume */ > + { 0x00000752, 0x0000 }, /* R1874 - AIF2TX3MIX Input 2 Source */ > + { 0x00000753, 0x0080 }, /* R1875 - AIF2TX3MIX Input 2 Volume */ > + { 0x00000754, 0x0000 }, /* R1876 - AIF2TX3MIX Input 3 Source */ > + { 0x00000755, 0x0080 }, /* R1877 - AIF2TX3MIX Input 3 Volume */ > + { 0x00000756, 0x0000 }, /* R1878 - AIF2TX3MIX Input 4 Source */ > + { 0x00000757, 0x0080 }, /* R1879 - AIF2TX3MIX Input 4 Volume */ > + { 0x00000758, 0x0000 }, /* R1880 - AIF2TX4MIX Input 1 Source */ > + { 0x00000759, 0x0080 }, /* R1881 - AIF2TX4MIX Input 1 Volume */ > + { 0x0000075A, 0x0000 }, /* R1882 - AIF2TX4MIX Input 2 Source */ > + { 0x0000075B, 0x0080 }, /* R1883 - AIF2TX4MIX Input 2 Volume */ > + { 0x0000075C, 0x0000 }, /* R1884 - AIF2TX4MIX Input 3 Source */ > + { 0x0000075D, 0x0080 }, /* R1885 - AIF2TX4MIX Input 3 Volume */ > + { 0x0000075E, 0x0000 }, /* R1886 - AIF2TX4MIX Input 4 Source */ > + { 0x0000075F, 0x0080 }, /* R1887 - AIF2TX4MIX Input 4 Volume */ > + { 0x00000760, 0x0000 }, /* R1888 - AIF2TX5MIX Input 1 Source */ > + { 0x00000761, 0x0080 }, /* R1889 - AIF2TX5MIX Input 1 Volume */ > + { 0x00000762, 0x0000 }, /* R1890 - AIF2TX5MIX Input 2 Source */ > + { 0x00000763, 0x0080 }, /* R1891 - AIF2TX5MIX Input 2 Volume */ > + { 0x00000764, 0x0000 }, /* R1892 - AIF2TX5MIX Input 3 Source */ > + { 0x00000765, 0x0080 }, /* R1893 - AIF2TX5MIX Input 3 Volume */ > + { 0x00000766, 0x0000 }, /* R1894 - AIF2TX5MIX Input 4 Source */ > + { 0x00000767, 0x0080 }, /* R1895 - AIF2TX5MIX Input 4 Volume */ > + { 0x00000768, 0x0000 }, /* R1896 - AIF2TX6MIX Input 1 Source */ > + { 0x00000769, 0x0080 }, /* R1897 - AIF2TX6MIX Input 1 Volume */ > + { 0x0000076A, 0x0000 }, /* R1898 - AIF2TX6MIX Input 2 Source */ > + { 0x0000076B, 0x0080 }, /* R1899 - AIF2TX6MIX Input 2 Volume */ > + { 0x0000076C, 0x0000 }, /* R1900 - AIF2TX6MIX Input 3 Source */ > + { 0x0000076D, 0x0080 }, /* R1901 - AIF2TX6MIX Input 3 Volume */ > + { 0x0000076E, 0x0000 }, /* R1902 - AIF2TX6MIX Input 4 Source */ > + { 0x0000076F, 0x0080 }, /* R1903 - AIF2TX6MIX Input 4 Volume */ > + { 0x00000780, 0x0000 }, /* R1920 - AIF3TX1MIX Input 1 Source */ > + { 0x00000781, 0x0080 }, /* R1921 - AIF3TX1MIX Input 1 Volume */ > + { 0x00000782, 0x0000 }, /* R1922 - AIF3TX1MIX Input 2 Source */ > + { 0x00000783, 0x0080 }, /* R1923 - AIF3TX1MIX Input 2 Volume */ > + { 0x00000784, 0x0000 }, /* R1924 - AIF3TX1MIX Input 3 Source */ > + { 0x00000785, 0x0080 }, /* R1925 - AIF3TX1MIX Input 3 Volume */ > + { 0x00000786, 0x0000 }, /* R1926 - AIF3TX1MIX Input 4 Source */ > + { 0x00000787, 0x0080 }, /* R1927 - AIF3TX1MIX Input 4 Volume */ > + { 0x00000788, 0x0000 }, /* R1928 - AIF3TX2MIX Input 1 Source */ > + { 0x00000789, 0x0080 }, /* R1929 - AIF3TX2MIX Input 1 Volume */ > + { 0x0000078A, 0x0000 }, /* R1930 - AIF3TX2MIX Input 2 Source */ > + { 0x0000078B, 0x0080 }, /* R1931 - AIF3TX2MIX Input 2 Volume */ > + { 0x0000078C, 0x0000 }, /* R1932 - AIF3TX2MIX Input 3 Source */ > + { 0x0000078D, 0x0080 }, /* R1933 - AIF3TX2MIX Input 3 Volume */ > + { 0x0000078E, 0x0000 }, /* R1934 - AIF3TX2MIX Input 4 Source */ > + { 0x0000078F, 0x0080 }, /* R1935 - AIF3TX2MIX Input 4 Volume */ > + { 0x00000880, 0x0000 }, /* R2176 - EQ1MIX Input 1 Source */ > + { 0x00000881, 0x0080 }, /* R2177 - EQ1MIX Input 1 Volume */ > + { 0x00000882, 0x0000 }, /* R2178 - EQ1MIX Input 2 Source */ > + { 0x00000883, 0x0080 }, /* R2179 - EQ1MIX Input 2 Volume */ > + { 0x00000884, 0x0000 }, /* R2180 - EQ1MIX Input 3 Source */ > + { 0x00000885, 0x0080 }, /* R2181 - EQ1MIX Input 3 Volume */ > + { 0x00000886, 0x0000 }, /* R2182 - EQ1MIX Input 4 Source */ > + { 0x00000887, 0x0080 }, /* R2183 - EQ1MIX Input 4 Volume */ > + { 0x00000888, 0x0000 }, /* R2184 - EQ2MIX Input 1 Source */ > + { 0x00000889, 0x0080 }, /* R2185 - EQ2MIX Input 1 Volume */ > + { 0x0000088A, 0x0000 }, /* R2186 - EQ2MIX Input 2 Source */ > + { 0x0000088B, 0x0080 }, /* R2187 - EQ2MIX Input 2 Volume */ > + { 0x0000088C, 0x0000 }, /* R2188 - EQ2MIX Input 3 Source */ > + { 0x0000088D, 0x0080 }, /* R2189 - EQ2MIX Input 3 Volume */ > + { 0x0000088E, 0x0000 }, /* R2190 - EQ2MIX Input 4 Source */ > + { 0x0000088F, 0x0080 }, /* R2191 - EQ2MIX Input 4 Volume */ > + { 0x000008C0, 0x0000 }, /* R2240 - DRC1LMIX Input 1 Source */ > + { 0x000008C1, 0x0080 }, /* R2241 - DRC1LMIX Input 1 Volume */ > + { 0x000008C2, 0x0000 }, /* R2242 - DRC1LMIX Input 2 Source */ > + { 0x000008C3, 0x0080 }, /* R2243 - DRC1LMIX Input 2 Volume */ > + { 0x000008C4, 0x0000 }, /* R2244 - DRC1LMIX Input 3 Source */ > + { 0x000008C5, 0x0080 }, /* R2245 - DRC1LMIX Input 3 Volume */ > + { 0x000008C6, 0x0000 }, /* R2246 - DRC1LMIX Input 4 Source */ > + { 0x000008C7, 0x0080 }, /* R2247 - DRC1LMIX Input 4 Volume */ > + { 0x000008C8, 0x0000 }, /* R2248 - DRC1RMIX Input 1 Source */ > + { 0x000008C9, 0x0080 }, /* R2249 - DRC1RMIX Input 1 Volume */ > + { 0x000008CA, 0x0000 }, /* R2250 - DRC1RMIX Input 2 Source */ > + { 0x000008CB, 0x0080 }, /* R2251 - DRC1RMIX Input 2 Volume */ > + { 0x000008CC, 0x0000 }, /* R2252 - DRC1RMIX Input 3 Source */ > + { 0x000008CD, 0x0080 }, /* R2253 - DRC1RMIX Input 3 Volume */ > + { 0x000008CE, 0x0000 }, /* R2254 - DRC1RMIX Input 4 Source */ > + { 0x000008CF, 0x0080 }, /* R2255 - DRC1RMIX Input 4 Volume */ > + { 0x000008D0, 0x0000 }, /* R2256 - DRC2LMIX Input 1 Source */ > + { 0x000008D1, 0x0080 }, /* R2257 - DRC2LMIX Input 1 Volume */ > + { 0x000008D2, 0x0000 }, /* R2258 - DRC2LMIX Input 2 Source */ > + { 0x000008D3, 0x0080 }, /* R2259 - DRC2LMIX Input 2 Volume */ > + { 0x000008D4, 0x0000 }, /* R2260 - DRC2LMIX Input 3 Source */ > + { 0x000008D5, 0x0080 }, /* R2261 - DRC2LMIX Input 3 Volume */ > + { 0x000008D6, 0x0000 }, /* R2262 - DRC2LMIX Input 4 Source */ > + { 0x000008D7, 0x0080 }, /* R2263 - DRC2LMIX Input 4 Volume */ > + { 0x000008D8, 0x0000 }, /* R2264 - DRC2RMIX Input 1 Source */ > + { 0x000008D9, 0x0080 }, /* R2265 - DRC2RMIX Input 1 Volume */ > + { 0x000008DA, 0x0000 }, /* R2266 - DRC2RMIX Input 2 Source */ > + { 0x000008DB, 0x0080 }, /* R2267 - DRC2RMIX Input 2 Volume */ > + { 0x000008DC, 0x0000 }, /* R2268 - DRC2RMIX Input 3 Source */ > + { 0x000008DD, 0x0080 }, /* R2269 - DRC2RMIX Input 3 Volume */ > + { 0x000008DE, 0x0000 }, /* R2270 - DRC2RMIX Input 4 Source */ > + { 0x000008DF, 0x0080 }, /* R2271 - DRC2RMIX Input 4 Volume */ > + { 0x00000900, 0x0000 }, /* R2304 - HPLP1MIX Input 1 Source */ > + { 0x00000901, 0x0080 }, /* R2305 - HPLP1MIX Input 1 Volume */ > + { 0x00000902, 0x0000 }, /* R2306 - HPLP1MIX Input 2 Source */ > + { 0x00000903, 0x0080 }, /* R2307 - HPLP1MIX Input 2 Volume */ > + { 0x00000904, 0x0000 }, /* R2308 - HPLP1MIX Input 3 Source */ > + { 0x00000905, 0x0080 }, /* R2309 - HPLP1MIX Input 3 Volume */ > + { 0x00000906, 0x0000 }, /* R2310 - HPLP1MIX Input 4 Source */ > + { 0x00000907, 0x0080 }, /* R2311 - HPLP1MIX Input 4 Volume */ > + { 0x00000908, 0x0000 }, /* R2312 - HPLP2MIX Input 1 Source */ > + { 0x00000909, 0x0080 }, /* R2313 - HPLP2MIX Input 1 Volume */ > + { 0x0000090A, 0x0000 }, /* R2314 - HPLP2MIX Input 2 Source */ > + { 0x0000090B, 0x0080 }, /* R2315 - HPLP2MIX Input 2 Volume */ > + { 0x0000090C, 0x0000 }, /* R2316 - HPLP2MIX Input 3 Source */ > + { 0x0000090D, 0x0080 }, /* R2317 - HPLP2MIX Input 3 Volume */ > + { 0x0000090E, 0x0000 }, /* R2318 - HPLP2MIX Input 4 Source */ > + { 0x0000090F, 0x0080 }, /* R2319 - HPLP2MIX Input 4 Volume */ > + { 0x00000910, 0x0000 }, /* R2320 - HPLP3MIX Input 1 Source */ > + { 0x00000911, 0x0080 }, /* R2321 - HPLP3MIX Input 1 Volume */ > + { 0x00000912, 0x0000 }, /* R2322 - HPLP3MIX Input 2 Source */ > + { 0x00000913, 0x0080 }, /* R2323 - HPLP3MIX Input 2 Volume */ > + { 0x00000914, 0x0000 }, /* R2324 - HPLP3MIX Input 3 Source */ > + { 0x00000915, 0x0080 }, /* R2325 - HPLP3MIX Input 3 Volume */ > + { 0x00000916, 0x0000 }, /* R2326 - HPLP3MIX Input 4 Source */ > + { 0x00000917, 0x0080 }, /* R2327 - HPLP3MIX Input 4 Volume */ > + { 0x00000918, 0x0000 }, /* R2328 - HPLP4MIX Input 1 Source */ > + { 0x00000919, 0x0080 }, /* R2329 - HPLP4MIX Input 1 Volume */ > + { 0x0000091A, 0x0000 }, /* R2330 - HPLP4MIX Input 2 Source */ > + { 0x0000091B, 0x0080 }, /* R2331 - HPLP4MIX Input 2 Volume */ > + { 0x0000091C, 0x0000 }, /* R2332 - HPLP4MIX Input 3 Source */ > + { 0x0000091D, 0x0080 }, /* R2333 - HPLP4MIX Input 3 Volume */ > + { 0x0000091E, 0x0000 }, /* R2334 - HPLP4MIX Input 4 Source */ > + { 0x0000091F, 0x0080 }, /* R2335 - HPLP4MIX Input 4 Volume */ > + { 0x00000980, 0x0000 }, /* R2432 - DSP2LMIX Input 1 Source */ > + { 0x00000981, 0x0080 }, /* R2433 - DSP2LMIX Input 1 Volume */ > + { 0x00000982, 0x0000 }, /* R2434 - DSP2LMIX Input 2 Source */ > + { 0x00000983, 0x0080 }, /* R2435 - DSP2LMIX Input 2 Volume */ > + { 0x00000984, 0x0000 }, /* R2436 - DSP2LMIX Input 3 Source */ > + { 0x00000985, 0x0080 }, /* R2437 - DSP2LMIX Input 3 Volume */ > + { 0x00000986, 0x0000 }, /* R2438 - DSP2LMIX Input 4 Source */ > + { 0x00000987, 0x0080 }, /* R2439 - DSP2LMIX Input 4 Volume */ > + { 0x00000988, 0x0000 }, /* R2440 - DSP2RMIX Input 1 Source */ > + { 0x00000989, 0x0080 }, /* R2441 - DSP2RMIX Input 1 Volume */ > + { 0x0000098A, 0x0000 }, /* R2442 - DSP2RMIX Input 2 Source */ > + { 0x0000098B, 0x0080 }, /* R2443 - DSP2RMIX Input 2 Volume */ > + { 0x0000098C, 0x0000 }, /* R2444 - DSP2RMIX Input 3 Source */ > + { 0x0000098D, 0x0080 }, /* R2445 - DSP2RMIX Input 3 Volume */ > + { 0x0000098E, 0x0000 }, /* R2446 - DSP2RMIX Input 4 Source */ > + { 0x0000098F, 0x0080 }, /* R2447 - DSP2RMIX Input 4 Volume */ > + { 0x00000990, 0x0000 }, /* R2448 - DSP2AUX1MIX Input 1 Source */ > + { 0x00000998, 0x0000 }, /* R2456 - DSP2AUX2MIX Input 1 Source */ > + { 0x000009A0, 0x0000 }, /* R2464 - DSP2AUX3MIX Input 1 Source */ > + { 0x000009A8, 0x0000 }, /* R2472 - DSP2AUX4MIX Input 1 Source */ > + { 0x000009B0, 0x0000 }, /* R2480 - DSP2AUX5MIX Input 1 Source */ > + { 0x000009B8, 0x0000 }, /* R2488 - DSP2AUX6MIX Input 1 Source */ > + { 0x000009C0, 0x0000 }, /* R2496 - DSP3LMIX Input 1 Source */ > + { 0x000009C1, 0x0080 }, /* R2497 - DSP3LMIX Input 1 Volume */ > + { 0x000009C2, 0x0000 }, /* R2498 - DSP3LMIX Input 2 Source */ > + { 0x000009C3, 0x0080 }, /* R2499 - DSP3LMIX Input 2 Volume */ > + { 0x000009C4, 0x0000 }, /* R2500 - DSP3LMIX Input 3 Source */ > + { 0x000009C5, 0x0080 }, /* R2501 - DSP3LMIX Input 3 Volume */ > + { 0x000009C6, 0x0000 }, /* R2502 - DSP3LMIX Input 4 Source */ > + { 0x000009C7, 0x0080 }, /* R2503 - DSP3LMIX Input 4 Volume */ > + { 0x000009C8, 0x0000 }, /* R2504 - DSP3RMIX Input 1 Source */ > + { 0x000009C9, 0x0080 }, /* R2505 - DSP3RMIX Input 1 Volume */ > + { 0x000009CA, 0x0000 }, /* R2506 - DSP3RMIX Input 2 Source */ > + { 0x000009CB, 0x0080 }, /* R2507 - DSP3RMIX Input 2 Volume */ > + { 0x000009CC, 0x0000 }, /* R2508 - DSP3RMIX Input 3 Source */ > + { 0x000009CD, 0x0080 }, /* R2509 - DSP3RMIX Input 3 Volume */ > + { 0x000009CE, 0x0000 }, /* R2510 - DSP3RMIX Input 4 Source */ > + { 0x000009CF, 0x0080 }, /* R2511 - DSP3RMIX Input 4 Volume */ > + { 0x000009D0, 0x0000 }, /* R2512 - DSP3AUX1MIX Input 1 Source */ > + { 0x000009D8, 0x0000 }, /* R2520 - DSP3AUX2MIX Input 1 Source */ > + { 0x000009E0, 0x0000 }, /* R2528 - DSP3AUX3MIX Input 1 Source */ > + { 0x000009E8, 0x0000 }, /* R2536 - DSP3AUX4MIX Input 1 Source */ > + { 0x000009F0, 0x0000 }, /* R2544 - DSP3AUX5MIX Input 1 Source */ > + { 0x000009F8, 0x0000 }, /* R2552 - DSP3AUX6MIX Input 1 Source */ > + { 0x00000A80, 0x0000 }, /* R2688 - ASRC1LMIX Input 1 Source */ > + { 0x00000A88, 0x0000 }, /* R2696 - ASRC1RMIX Input 1 Source */ > + { 0x00000A90, 0x0000 }, /* R2704 - ASRC2LMIX Input 1 Source */ > + { 0x00000A98, 0x0000 }, /* R2712 - ASRC2RMIX Input 1 Source */ > + { 0x00000B00, 0x0000 }, /* R2816 - ISRC1DEC1MIX Input 1 Source */ > + { 0x00000B08, 0x0000 }, /* R2824 - ISRC1DEC2MIX Input 1 Source */ > + { 0x00000B10, 0x0000 }, /* R2832 - ISRC1DEC3MIX Input 1 Source */ > + { 0x00000B18, 0x0000 }, /* R2840 - ISRC1DEC4MIX Input 1 Source */ > + { 0x00000B20, 0x0000 }, /* R2848 - ISRC1INT1MIX Input 1 Source */ > + { 0x00000B28, 0x0000 }, /* R2856 - ISRC1INT2MIX Input 1 Source */ > + { 0x00000B30, 0x0000 }, /* R2864 - ISRC1INT3MIX Input 1 Source */ > + { 0x00000B38, 0x0000 }, /* R2872 - ISRC1INT4MIX Input 1 Source */ > + { 0x00000B40, 0x0000 }, /* R2880 - ISRC2DEC1MIX Input 1 Source */ > + { 0x00000B48, 0x0000 }, /* R2888 - ISRC2DEC2MIX Input 1 Source */ > + { 0x00000B50, 0x0000 }, /* R2896 - ISRC2DEC3MIX Input 1 Source */ > + { 0x00000B58, 0x0000 }, /* R2904 - ISRC2DEC4MIX Input 1 Source */ > + { 0x00000B60, 0x0000 }, /* R2912 - ISRC2INT1MIX Input 1 Source */ > + { 0x00000B68, 0x0000 }, /* R2920 - ISRC2INT2MIX Input 1 Source */ > + { 0x00000B70, 0x0000 }, /* R2928 - ISRC2INT3MIX Input 1 Source */ > + { 0x00000B78, 0x0000 }, /* R2936 - ISRC2INT4MIX Input 1 Source */ > + { 0x00000B80, 0x0000 }, /* R2944 - ISRC3DEC1MIX Input 1 Source */ > + { 0x00000B88, 0x0000 }, /* R2952 - ISRC3DEC2MIX Input 1 Source */ > + { 0x00000B90, 0x0000 }, /* R2960 - ISRC3DEC3MIX Input 1 Source */ > + { 0x00000B98, 0x0000 }, /* R2968 - ISRC3DEC4MIX Input 1 Source */ > + { 0x00000BA0, 0x0000 }, /* R2976 - ISRC3INT1MIX Input 1 Source */ > + { 0x00000BA8, 0x0000 }, /* R2984 - ISRC3INT2MIX Input 1 Source */ > + { 0x00000BB0, 0x0000 }, /* R2992 - ISRC3INT3MIX Input 1 Source */ > + { 0x00000BB8, 0x0000 }, /* R3000 - ISRC3INT4MIX Input 1 Source */ > + { 0x00000C00, 0xA101 }, /* R3072 - GPIO1 CTRL */ > + { 0x00000C01, 0xA101 }, /* R3073 - GPIO2 CTRL */ > + { 0x00000C0F, 0x0400 }, /* R3087 - IRQ CTRL 1 */ > + { 0x00000C10, 0x1000 }, /* R3088 - GPIO Debounce Config */ > + { 0x00000C20, 0x0002 }, /* R3104 - Misc Pad Ctrl 1 */ > + { 0x00000C21, 0x8001 }, /* R3105 - Misc Pad Ctrl 2 */ > + { 0x00000C22, 0x0000 }, /* R3106 - Misc Pad Ctrl 3 */ > + { 0x00000C23, 0x0000 }, /* R3107 - Misc Pad Ctrl 4 */ > + { 0x00000C24, 0x0000 }, /* R3108 - Misc Pad Ctrl 5 */ > + { 0x00000C25, 0x0000 }, /* R3109 - Misc Pad Ctrl 6 */ > + { 0x00000C30, 0x0404 }, /* R3120 - Misc Pad Ctrl 7 */ > + { 0x00000C32, 0x0404 }, /* R3122 - Misc Pad Ctrl 9 */ > + { 0x00000C33, 0x0404 }, /* R3123 - Misc Pad Ctrl 10 */ > + { 0x00000C34, 0x0404 }, /* R3124 - Misc Pad Ctrl 11 */ > + { 0x00000C35, 0x0404 }, /* R3125 - Misc Pad Ctrl 12 */ > + { 0x00000C36, 0x0400 }, /* R3126 - Misc Pad Ctrl 13 */ > + { 0x00000C37, 0x0404 }, /* R3127 - Misc Pad Ctrl 14 */ > + { 0x00000C39, 0x0400 }, /* R3129 - Misc Pad Ctrl 16 */ > + { 0x00000D08, 0x0007 }, /* R3336 - Interrupt Status 1 Mask */ > + { 0x00000D09, 0x06FF }, /* R3337 - Interrupt Status 2 Mask */ > + { 0x00000D0A, 0xCFEF }, /* R3338 - Interrupt Status 3 Mask */ > + { 0x00000D0B, 0xFFC3 }, /* R3339 - Interrupt Status 4 Mask */ > + { 0x00000D0C, 0x000B }, /* R3340 - Interrupt Status 5 Mask */ > + { 0x00000D0D, 0xD005 }, /* R3341 - Interrupt Status 6 Mask */ > + { 0x00000D0F, 0x0000 }, /* R3343 - Interrupt Control */ > + { 0x00000D18, 0x0007 }, /* R3352 - IRQ2 Status 1 Mask */ > + { 0x00000D19, 0x06FF }, /* R3353 - IRQ2 Status 2 Mask */ > + { 0x00000D1A, 0xCFEF }, /* R3354 - IRQ2 Status 3 Mask */ > + { 0x00000D1B, 0xFFC3 }, /* R3355 - IRQ2 Status 4 Mask */ > + { 0x00000D1C, 0x000B }, /* R3356 - IRQ2 Status 5 Mask */ > + { 0x00000D1D, 0xD005 }, /* R3357 - IRQ2 Status 6 Mask */ > + { 0x00000D1F, 0x0000 }, /* R3359 - IRQ2 Control */ > + { 0x00000E00, 0x0000 }, /* R3584 - FX_Ctrl1 */ > + { 0x00000E10, 0x6318 }, /* R3600 - EQ1_1 */ > + { 0x00000E11, 0x6300 }, /* R3601 - EQ1_2 */ > + { 0x00000E12, 0x0FC8 }, /* R3602 - EQ1_3 */ > + { 0x00000E13, 0x03FE }, /* R3603 - EQ1_4 */ > + { 0x00000E14, 0x00E0 }, /* R3604 - EQ1_5 */ > + { 0x00000E15, 0x1EC4 }, /* R3605 - EQ1_6 */ > + { 0x00000E16, 0xF136 }, /* R3606 - EQ1_7 */ > + { 0x00000E17, 0x0409 }, /* R3607 - EQ1_8 */ > + { 0x00000E18, 0x04CC }, /* R3608 - EQ1_9 */ > + { 0x00000E19, 0x1C9B }, /* R3609 - EQ1_10 */ > + { 0x00000E1A, 0xF337 }, /* R3610 - EQ1_11 */ > + { 0x00000E1B, 0x040B }, /* R3611 - EQ1_12 */ > + { 0x00000E1C, 0x0CBB }, /* R3612 - EQ1_13 */ > + { 0x00000E1D, 0x16F8 }, /* R3613 - EQ1_14 */ > + { 0x00000E1E, 0xF7D9 }, /* R3614 - EQ1_15 */ > + { 0x00000E1F, 0x040A }, /* R3615 - EQ1_16 */ > + { 0x00000E20, 0x1F14 }, /* R3616 - EQ1_17 */ > + { 0x00000E21, 0x058C }, /* R3617 - EQ1_18 */ > + { 0x00000E22, 0x0563 }, /* R3618 - EQ1_19 */ > + { 0x00000E23, 0x4000 }, /* R3619 - EQ1_20 */ > + { 0x00000E24, 0x0B75 }, /* R3620 - EQ1_21 */ > + { 0x00000E26, 0x6318 }, /* R3622 - EQ2_1 */ > + { 0x00000E27, 0x6300 }, /* R3623 - EQ2_2 */ > + { 0x00000E28, 0x0FC8 }, /* R3624 - EQ2_3 */ > + { 0x00000E29, 0x03FE }, /* R3625 - EQ2_4 */ > + { 0x00000E2A, 0x00E0 }, /* R3626 - EQ2_5 */ > + { 0x00000E2B, 0x1EC4 }, /* R3627 - EQ2_6 */ > + { 0x00000E2C, 0xF136 }, /* R3628 - EQ2_7 */ > + { 0x00000E2D, 0x0409 }, /* R3629 - EQ2_8 */ > + { 0x00000E2E, 0x04CC }, /* R3630 - EQ2_9 */ > + { 0x00000E2F, 0x1C9B }, /* R3631 - EQ2_10 */ > + { 0x00000E30, 0xF337 }, /* R3632 - EQ2_11 */ > + { 0x00000E31, 0x040B }, /* R3633 - EQ2_12 */ > + { 0x00000E32, 0x0CBB }, /* R3634 - EQ2_13 */ > + { 0x00000E33, 0x16F8 }, /* R3635 - EQ2_14 */ > + { 0x00000E34, 0xF7D9 }, /* R3636 - EQ2_15 */ > + { 0x00000E35, 0x040A }, /* R3637 - EQ2_16 */ > + { 0x00000E36, 0x1F14 }, /* R3638 - EQ2_17 */ > + { 0x00000E37, 0x058C }, /* R3639 - EQ2_18 */ > + { 0x00000E38, 0x0563 }, /* R3640 - EQ2_19 */ > + { 0x00000E39, 0x4000 }, /* R3641 - EQ2_20 */ > + { 0x00000E3A, 0x0B75 }, /* R3642 - EQ2_21 */ > + { 0x00000E80, 0x0018 }, /* R3712 - DRC1 ctrl1 */ > + { 0x00000E81, 0x0933 }, /* R3713 - DRC1 ctrl2 */ > + { 0x00000E82, 0x0018 }, /* R3714 - DRC1 ctrl3 */ > + { 0x00000E83, 0x0000 }, /* R3715 - DRC1 ctrl4 */ > + { 0x00000E84, 0x0000 }, /* R3716 - DRC1 ctrl5 */ > + { 0x00000E89, 0x0018 }, /* R3721 - DRC2 ctrl1 */ > + { 0x00000E8A, 0x0933 }, /* R3722 - DRC2 ctrl2 */ > + { 0x00000E8B, 0x0018 }, /* R3723 - DRC2 ctrl3 */ > + { 0x00000E8C, 0x0000 }, /* R3724 - DRC2 ctrl4 */ > + { 0x00000E8D, 0x0000 }, /* R3725 - DRC2 ctrl5 */ > + { 0x00000EC0, 0x0000 }, /* R3776 - HPLPF1_1 */ > + { 0x00000EC1, 0x0000 }, /* R3777 - HPLPF1_2 */ > + { 0x00000EC4, 0x0000 }, /* R3780 - HPLPF2_1 */ > + { 0x00000EC5, 0x0000 }, /* R3781 - HPLPF2_2 */ > + { 0x00000EC8, 0x0000 }, /* R3784 - HPLPF3_1 */ > + { 0x00000EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ > + { 0x00000ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ > + { 0x00000ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ > + { 0x00000EE0, 0x0000 }, /* R3808 - ASRC_ENABLE */ > + { 0x00000EE2, 0x0000 }, /* R3810 - ASRC_RATE1 */ > + { 0x00000EE3, 0x4000 }, /* R3811 - ASRC_RATE2 */ > + { 0x00000EF0, 0x0000 }, /* R3824 - ISRC 1 CTRL 1 */ > + { 0x00000EF1, 0x0000 }, /* R3825 - ISRC 1 CTRL 2 */ > + { 0x00000EF2, 0x0000 }, /* R3826 - ISRC 1 CTRL 3 */ > + { 0x00000EF3, 0x0000 }, /* R3827 - ISRC 2 CTRL 1 */ > + { 0x00000EF4, 0x0000 }, /* R3828 - ISRC 2 CTRL 2 */ > + { 0x00000EF5, 0x0000 }, /* R3829 - ISRC 2 CTRL 3 */ > + { 0x00000EF6, 0x0000 }, /* R3830 - ISRC 3 CTRL 1 */ > + { 0x00000EF7, 0x0000 }, /* R3831 - ISRC 3 CTRL 2 */ > + { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ > + { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ > + { 0x00001300, 0x0010 }, /* R4864 - DSP3 Control 1 */ > +}; > + > +static bool cs47l24_is_adsp_memory(unsigned int reg) > +{ > + switch (reg) { > + case 0x200000 ... 0x205fff: /* DSP2 PM */ > + case 0x280000 ... 0x281fff: /* DSP2 ZM */ > + case 0x290000 ... 0x2a7fff: /* DSP2 XM */ > + case 0x2a8000 ... 0x2b3fff: /* DSP2 YM */ > + case 0x300000 ... 0x308fff: /* DSP3 PM */ > + case 0x380000 ... 0x381fff: /* DSP3 ZM */ > + case 0x390000 ... 0x3a7fff: /* DSP3 XM */ > + case 0x3a8000 ... 0x3b3fff: /* DSP3 YM */ > + return true; > + default: > + return false; > + } > +} > + > +static bool cs47l24_readable_register(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case ARIZONA_SOFTWARE_RESET: > + case ARIZONA_DEVICE_REVISION: > + case ARIZONA_CTRL_IF_SPI_CFG_1: > + case ARIZONA_WRITE_SEQUENCER_CTRL_0: > + case ARIZONA_WRITE_SEQUENCER_CTRL_1: > + case ARIZONA_WRITE_SEQUENCER_CTRL_2: > + case ARIZONA_TONE_GENERATOR_1: > + case ARIZONA_TONE_GENERATOR_2: > + case ARIZONA_TONE_GENERATOR_3: > + case ARIZONA_TONE_GENERATOR_4: > + case ARIZONA_TONE_GENERATOR_5: > + case ARIZONA_PWM_DRIVE_1: > + case ARIZONA_PWM_DRIVE_2: > + case ARIZONA_PWM_DRIVE_3: > + case ARIZONA_SEQUENCE_CONTROL: > + case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_1: > + case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_2: > + case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_3: > + case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_4: > + case ARIZONA_COMFORT_NOISE_GENERATOR: > + case ARIZONA_HAPTICS_CONTROL_1: > + case ARIZONA_HAPTICS_CONTROL_2: > + case ARIZONA_HAPTICS_PHASE_1_INTENSITY: > + case ARIZONA_HAPTICS_PHASE_1_DURATION: > + case ARIZONA_HAPTICS_PHASE_2_INTENSITY: > + case ARIZONA_HAPTICS_PHASE_2_DURATION: > + case ARIZONA_HAPTICS_PHASE_3_INTENSITY: > + case ARIZONA_HAPTICS_PHASE_3_DURATION: > + case ARIZONA_HAPTICS_STATUS: > + case ARIZONA_CLOCK_32K_1: > + case ARIZONA_SYSTEM_CLOCK_1: > + case ARIZONA_SAMPLE_RATE_1: > + case ARIZONA_SAMPLE_RATE_2: > + case ARIZONA_SAMPLE_RATE_3: > + case ARIZONA_SAMPLE_RATE_1_STATUS: > + case ARIZONA_SAMPLE_RATE_2_STATUS: > + case ARIZONA_SAMPLE_RATE_3_STATUS: > + case ARIZONA_ASYNC_CLOCK_1: > + case ARIZONA_ASYNC_SAMPLE_RATE_1: > + case ARIZONA_ASYNC_SAMPLE_RATE_1_STATUS: > + case ARIZONA_ASYNC_SAMPLE_RATE_2: > + case ARIZONA_ASYNC_SAMPLE_RATE_2_STATUS: > + case ARIZONA_OUTPUT_SYSTEM_CLOCK: > + case ARIZONA_OUTPUT_ASYNC_CLOCK: > + case ARIZONA_RATE_ESTIMATOR_1: > + case ARIZONA_RATE_ESTIMATOR_2: > + case ARIZONA_RATE_ESTIMATOR_3: > + case ARIZONA_RATE_ESTIMATOR_4: > + case ARIZONA_RATE_ESTIMATOR_5: > + case ARIZONA_FLL1_CONTROL_1: > + case ARIZONA_FLL1_CONTROL_2: > + case ARIZONA_FLL1_CONTROL_3: > + case ARIZONA_FLL1_CONTROL_4: > + case ARIZONA_FLL1_CONTROL_5: > + case ARIZONA_FLL1_CONTROL_6: > + case ARIZONA_FLL1_CONTROL_7: > + case ARIZONA_FLL1_LOOP_FILTER_TEST_1: > + case ARIZONA_FLL1_NCO_TEST_0: > + case ARIZONA_FLL1_SYNCHRONISER_1: > + case ARIZONA_FLL1_SYNCHRONISER_2: > + case ARIZONA_FLL1_SYNCHRONISER_3: > + case ARIZONA_FLL1_SYNCHRONISER_4: > + case ARIZONA_FLL1_SYNCHRONISER_5: > + case ARIZONA_FLL1_SYNCHRONISER_6: > + case ARIZONA_FLL1_SYNCHRONISER_7: > + case ARIZONA_FLL1_SPREAD_SPECTRUM: > + case ARIZONA_FLL1_GPIO_CLOCK: > + case ARIZONA_FLL2_CONTROL_1: > + case ARIZONA_FLL2_CONTROL_2: > + case ARIZONA_FLL2_CONTROL_3: > + case ARIZONA_FLL2_CONTROL_4: > + case ARIZONA_FLL2_CONTROL_5: > + case ARIZONA_FLL2_CONTROL_6: > + case ARIZONA_FLL2_CONTROL_7: > + case ARIZONA_FLL2_LOOP_FILTER_TEST_1: > + case ARIZONA_FLL2_NCO_TEST_0: > + case ARIZONA_FLL2_SYNCHRONISER_1: > + case ARIZONA_FLL2_SYNCHRONISER_2: > + case ARIZONA_FLL2_SYNCHRONISER_3: > + case ARIZONA_FLL2_SYNCHRONISER_4: > + case ARIZONA_FLL2_SYNCHRONISER_5: > + case ARIZONA_FLL2_SYNCHRONISER_6: > + case ARIZONA_FLL2_SYNCHRONISER_7: > + case ARIZONA_FLL2_SPREAD_SPECTRUM: > + case ARIZONA_FLL2_GPIO_CLOCK: > + case ARIZONA_MIC_BIAS_CTRL_1: > + case ARIZONA_MIC_BIAS_CTRL_2: > + case ARIZONA_HP_CTRL_1L: > + case ARIZONA_HP_CTRL_1R: > + case ARIZONA_INPUT_ENABLES: > + case ARIZONA_INPUT_ENABLES_STATUS: > + case ARIZONA_INPUT_RATE: > + case ARIZONA_INPUT_VOLUME_RAMP: > + case ARIZONA_HPF_CONTROL: > + case ARIZONA_IN1L_CONTROL: > + case ARIZONA_ADC_DIGITAL_VOLUME_1L: > + case ARIZONA_DMIC1L_CONTROL: > + case ARIZONA_IN1R_CONTROL: > + case ARIZONA_ADC_DIGITAL_VOLUME_1R: > + case ARIZONA_DMIC1R_CONTROL: > + case ARIZONA_IN2L_CONTROL: > + case ARIZONA_ADC_DIGITAL_VOLUME_2L: > + case ARIZONA_DMIC2L_CONTROL: > + case ARIZONA_IN2R_CONTROL: > + case ARIZONA_ADC_DIGITAL_VOLUME_2R: > + case ARIZONA_DMIC2R_CONTROL: > + case ARIZONA_OUTPUT_ENABLES_1: > + case ARIZONA_OUTPUT_STATUS_1: > + case ARIZONA_RAW_OUTPUT_STATUS_1: > + case ARIZONA_OUTPUT_RATE_1: > + case ARIZONA_OUTPUT_VOLUME_RAMP: > + case ARIZONA_OUTPUT_PATH_CONFIG_1L: > + case ARIZONA_DAC_DIGITAL_VOLUME_1L: > + case ARIZONA_DAC_VOLUME_LIMIT_1L: > + case ARIZONA_NOISE_GATE_SELECT_1L: > + case ARIZONA_DAC_DIGITAL_VOLUME_1R: > + case ARIZONA_DAC_VOLUME_LIMIT_1R: > + case ARIZONA_NOISE_GATE_SELECT_1R: > + case ARIZONA_DAC_DIGITAL_VOLUME_4L: > + case ARIZONA_OUT_VOLUME_4L: > + case ARIZONA_NOISE_GATE_SELECT_4L: > + case ARIZONA_DAC_AEC_CONTROL_1: > + case ARIZONA_NOISE_GATE_CONTROL: > + case ARIZONA_HP1_SHORT_CIRCUIT_CTRL: > + case ARIZONA_AIF1_BCLK_CTRL: > + case ARIZONA_AIF1_TX_PIN_CTRL: > + case ARIZONA_AIF1_RX_PIN_CTRL: > + case ARIZONA_AIF1_RATE_CTRL: > + case ARIZONA_AIF1_FORMAT: > + case ARIZONA_AIF1_RX_BCLK_RATE: > + case ARIZONA_AIF1_FRAME_CTRL_1: > + case ARIZONA_AIF1_FRAME_CTRL_2: > + case ARIZONA_AIF1_FRAME_CTRL_3: > + case ARIZONA_AIF1_FRAME_CTRL_4: > + case ARIZONA_AIF1_FRAME_CTRL_5: > + case ARIZONA_AIF1_FRAME_CTRL_6: > + case ARIZONA_AIF1_FRAME_CTRL_7: > + case ARIZONA_AIF1_FRAME_CTRL_8: > + case ARIZONA_AIF1_FRAME_CTRL_9: > + case ARIZONA_AIF1_FRAME_CTRL_10: > + case ARIZONA_AIF1_FRAME_CTRL_11: > + case ARIZONA_AIF1_FRAME_CTRL_12: > + case ARIZONA_AIF1_FRAME_CTRL_13: > + case ARIZONA_AIF1_FRAME_CTRL_14: > + case ARIZONA_AIF1_FRAME_CTRL_15: > + case ARIZONA_AIF1_FRAME_CTRL_16: > + case ARIZONA_AIF1_FRAME_CTRL_17: > + case ARIZONA_AIF1_FRAME_CTRL_18: > + case ARIZONA_AIF1_TX_ENABLES: > + case ARIZONA_AIF1_RX_ENABLES: > + case ARIZONA_AIF2_BCLK_CTRL: > + case ARIZONA_AIF2_TX_PIN_CTRL: > + case ARIZONA_AIF2_RX_PIN_CTRL: > + case ARIZONA_AIF2_RATE_CTRL: > + case ARIZONA_AIF2_FORMAT: > + case ARIZONA_AIF2_RX_BCLK_RATE: > + case ARIZONA_AIF2_FRAME_CTRL_1: > + case ARIZONA_AIF2_FRAME_CTRL_2: > + case ARIZONA_AIF2_FRAME_CTRL_3: > + case ARIZONA_AIF2_FRAME_CTRL_4: > + case ARIZONA_AIF2_FRAME_CTRL_5: > + case ARIZONA_AIF2_FRAME_CTRL_6: > + case ARIZONA_AIF2_FRAME_CTRL_7: > + case ARIZONA_AIF2_FRAME_CTRL_8: > + case ARIZONA_AIF2_FRAME_CTRL_11: > + case ARIZONA_AIF2_FRAME_CTRL_12: > + case ARIZONA_AIF2_FRAME_CTRL_13: > + case ARIZONA_AIF2_FRAME_CTRL_14: > + case ARIZONA_AIF2_FRAME_CTRL_15: > + case ARIZONA_AIF2_FRAME_CTRL_16: > + case ARIZONA_AIF2_TX_ENABLES: > + case ARIZONA_AIF2_RX_ENABLES: > + case ARIZONA_AIF3_BCLK_CTRL: > + case ARIZONA_AIF3_TX_PIN_CTRL: > + case ARIZONA_AIF3_RX_PIN_CTRL: > + case ARIZONA_AIF3_RATE_CTRL: > + case ARIZONA_AIF3_FORMAT: > + case ARIZONA_AIF3_RX_BCLK_RATE: > + case ARIZONA_AIF3_FRAME_CTRL_1: > + case ARIZONA_AIF3_FRAME_CTRL_2: > + case ARIZONA_AIF3_FRAME_CTRL_3: > + case ARIZONA_AIF3_FRAME_CTRL_4: > + case ARIZONA_AIF3_FRAME_CTRL_11: > + case ARIZONA_AIF3_FRAME_CTRL_12: > + case ARIZONA_AIF3_TX_ENABLES: > + case ARIZONA_AIF3_RX_ENABLES: > + case ARIZONA_PWM1MIX_INPUT_1_SOURCE: > + case ARIZONA_PWM1MIX_INPUT_1_VOLUME: > + case ARIZONA_PWM1MIX_INPUT_2_SOURCE: > + case ARIZONA_PWM1MIX_INPUT_2_VOLUME: > + case ARIZONA_PWM1MIX_INPUT_3_SOURCE: > + case ARIZONA_PWM1MIX_INPUT_3_VOLUME: > + case ARIZONA_PWM1MIX_INPUT_4_SOURCE: > + case ARIZONA_PWM1MIX_INPUT_4_VOLUME: > + case ARIZONA_PWM2MIX_INPUT_1_SOURCE: > + case ARIZONA_PWM2MIX_INPUT_1_VOLUME: > + case ARIZONA_PWM2MIX_INPUT_2_SOURCE: > + case ARIZONA_PWM2MIX_INPUT_2_VOLUME: > + case ARIZONA_PWM2MIX_INPUT_3_SOURCE: > + case ARIZONA_PWM2MIX_INPUT_3_VOLUME: > + case ARIZONA_PWM2MIX_INPUT_4_SOURCE: > + case ARIZONA_PWM2MIX_INPUT_4_VOLUME: > + case ARIZONA_OUT1LMIX_INPUT_1_SOURCE: > + case ARIZONA_OUT1LMIX_INPUT_1_VOLUME: > + case ARIZONA_OUT1LMIX_INPUT_2_SOURCE: > + case ARIZONA_OUT1LMIX_INPUT_2_VOLUME: > + case ARIZONA_OUT1LMIX_INPUT_3_SOURCE: > + case ARIZONA_OUT1LMIX_INPUT_3_VOLUME: > + case ARIZONA_OUT1LMIX_INPUT_4_SOURCE: > + case ARIZONA_OUT1LMIX_INPUT_4_VOLUME: > + case ARIZONA_OUT1RMIX_INPUT_1_SOURCE: > + case ARIZONA_OUT1RMIX_INPUT_1_VOLUME: > + case ARIZONA_OUT1RMIX_INPUT_2_SOURCE: > + case ARIZONA_OUT1RMIX_INPUT_2_VOLUME: > + case ARIZONA_OUT1RMIX_INPUT_3_SOURCE: > + case ARIZONA_OUT1RMIX_INPUT_3_VOLUME: > + case ARIZONA_OUT1RMIX_INPUT_4_SOURCE: > + case ARIZONA_OUT1RMIX_INPUT_4_VOLUME: > + case ARIZONA_OUT4LMIX_INPUT_1_SOURCE: > + case ARIZONA_OUT4LMIX_INPUT_1_VOLUME: > + case ARIZONA_OUT4LMIX_INPUT_2_SOURCE: > + case ARIZONA_OUT4LMIX_INPUT_2_VOLUME: > + case ARIZONA_OUT4LMIX_INPUT_3_SOURCE: > + case ARIZONA_OUT4LMIX_INPUT_3_VOLUME: > + case ARIZONA_OUT4LMIX_INPUT_4_SOURCE: > + case ARIZONA_OUT4LMIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX1MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX1MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX1MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX1MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX1MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX1MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX1MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX2MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX2MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX2MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX2MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX2MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX2MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX2MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX3MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX3MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX3MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX3MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX3MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX3MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX3MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX4MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX4MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX4MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX4MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX4MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX4MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX4MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX5MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX5MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX5MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX5MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX5MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX5MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX5MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX6MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX6MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX6MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX6MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX6MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX6MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX6MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX7MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX7MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX7MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX7MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX7MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX7MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX7MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF1TX8MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF1TX8MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF1TX8MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF1TX8MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF1TX8MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF1TX8MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF1TX8MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF2TX1MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF2TX1MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF2TX1MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF2TX1MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF2TX1MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF2TX1MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF2TX1MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF2TX2MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF2TX2MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF2TX2MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF2TX2MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF2TX2MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF2TX2MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF2TX2MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF2TX3MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF2TX3MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF2TX3MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF2TX3MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF2TX3MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF2TX3MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF2TX3MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF2TX4MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF2TX4MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF2TX4MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF2TX4MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF2TX4MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF2TX4MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF2TX4MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF2TX5MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF2TX5MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF2TX5MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF2TX5MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF2TX5MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF2TX5MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF2TX5MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF2TX6MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF2TX6MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF2TX6MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF2TX6MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF2TX6MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF2TX6MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF2TX6MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF3TX1MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF3TX1MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF3TX1MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF3TX1MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF3TX1MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF3TX1MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF3TX1MIX_INPUT_4_VOLUME: > + case ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE: > + case ARIZONA_AIF3TX2MIX_INPUT_1_VOLUME: > + case ARIZONA_AIF3TX2MIX_INPUT_2_SOURCE: > + case ARIZONA_AIF3TX2MIX_INPUT_2_VOLUME: > + case ARIZONA_AIF3TX2MIX_INPUT_3_SOURCE: > + case ARIZONA_AIF3TX2MIX_INPUT_3_VOLUME: > + case ARIZONA_AIF3TX2MIX_INPUT_4_SOURCE: > + case ARIZONA_AIF3TX2MIX_INPUT_4_VOLUME: > + case ARIZONA_EQ1MIX_INPUT_1_SOURCE: > + case ARIZONA_EQ1MIX_INPUT_1_VOLUME: > + case ARIZONA_EQ1MIX_INPUT_2_SOURCE: > + case ARIZONA_EQ1MIX_INPUT_2_VOLUME: > + case ARIZONA_EQ1MIX_INPUT_3_SOURCE: > + case ARIZONA_EQ1MIX_INPUT_3_VOLUME: > + case ARIZONA_EQ1MIX_INPUT_4_SOURCE: > + case ARIZONA_EQ1MIX_INPUT_4_VOLUME: > + case ARIZONA_EQ2MIX_INPUT_1_SOURCE: > + case ARIZONA_EQ2MIX_INPUT_1_VOLUME: > + case ARIZONA_EQ2MIX_INPUT_2_SOURCE: > + case ARIZONA_EQ2MIX_INPUT_2_VOLUME: > + case ARIZONA_EQ2MIX_INPUT_3_SOURCE: > + case ARIZONA_EQ2MIX_INPUT_3_VOLUME: > + case ARIZONA_EQ2MIX_INPUT_4_SOURCE: > + case ARIZONA_EQ2MIX_INPUT_4_VOLUME: > + case ARIZONA_DRC1LMIX_INPUT_1_SOURCE: > + case ARIZONA_DRC1LMIX_INPUT_1_VOLUME: > + case ARIZONA_DRC1LMIX_INPUT_2_SOURCE: > + case ARIZONA_DRC1LMIX_INPUT_2_VOLUME: > + case ARIZONA_DRC1LMIX_INPUT_3_SOURCE: > + case ARIZONA_DRC1LMIX_INPUT_3_VOLUME: > + case ARIZONA_DRC1LMIX_INPUT_4_SOURCE: > + case ARIZONA_DRC1LMIX_INPUT_4_VOLUME: > + case ARIZONA_DRC1RMIX_INPUT_1_SOURCE: > + case ARIZONA_DRC1RMIX_INPUT_1_VOLUME: > + case ARIZONA_DRC1RMIX_INPUT_2_SOURCE: > + case ARIZONA_DRC1RMIX_INPUT_2_VOLUME: > + case ARIZONA_DRC1RMIX_INPUT_3_SOURCE: > + case ARIZONA_DRC1RMIX_INPUT_3_VOLUME: > + case ARIZONA_DRC1RMIX_INPUT_4_SOURCE: > + case ARIZONA_DRC1RMIX_INPUT_4_VOLUME: > + case ARIZONA_DRC2LMIX_INPUT_1_SOURCE: > + case ARIZONA_DRC2LMIX_INPUT_1_VOLUME: > + case ARIZONA_DRC2LMIX_INPUT_2_SOURCE: > + case ARIZONA_DRC2LMIX_INPUT_2_VOLUME: > + case ARIZONA_DRC2LMIX_INPUT_3_SOURCE: > + case ARIZONA_DRC2LMIX_INPUT_3_VOLUME: > + case ARIZONA_DRC2LMIX_INPUT_4_SOURCE: > + case ARIZONA_DRC2LMIX_INPUT_4_VOLUME: > + case ARIZONA_DRC2RMIX_INPUT_1_SOURCE: > + case ARIZONA_DRC2RMIX_INPUT_1_VOLUME: > + case ARIZONA_DRC2RMIX_INPUT_2_SOURCE: > + case ARIZONA_DRC2RMIX_INPUT_2_VOLUME: > + case ARIZONA_DRC2RMIX_INPUT_3_SOURCE: > + case ARIZONA_DRC2RMIX_INPUT_3_VOLUME: > + case ARIZONA_DRC2RMIX_INPUT_4_SOURCE: > + case ARIZONA_DRC2RMIX_INPUT_4_VOLUME: > + case ARIZONA_HPLP1MIX_INPUT_1_SOURCE: > + case ARIZONA_HPLP1MIX_INPUT_1_VOLUME: > + case ARIZONA_HPLP1MIX_INPUT_2_SOURCE: > + case ARIZONA_HPLP1MIX_INPUT_2_VOLUME: > + case ARIZONA_HPLP1MIX_INPUT_3_SOURCE: > + case ARIZONA_HPLP1MIX_INPUT_3_VOLUME: > + case ARIZONA_HPLP1MIX_INPUT_4_SOURCE: > + case ARIZONA_HPLP1MIX_INPUT_4_VOLUME: > + case ARIZONA_HPLP2MIX_INPUT_1_SOURCE: > + case ARIZONA_HPLP2MIX_INPUT_1_VOLUME: > + case ARIZONA_HPLP2MIX_INPUT_2_SOURCE: > + case ARIZONA_HPLP2MIX_INPUT_2_VOLUME: > + case ARIZONA_HPLP2MIX_INPUT_3_SOURCE: > + case ARIZONA_HPLP2MIX_INPUT_3_VOLUME: > + case ARIZONA_HPLP2MIX_INPUT_4_SOURCE: > + case ARIZONA_HPLP2MIX_INPUT_4_VOLUME: > + case ARIZONA_HPLP3MIX_INPUT_1_SOURCE: > + case ARIZONA_HPLP3MIX_INPUT_1_VOLUME: > + case ARIZONA_HPLP3MIX_INPUT_2_SOURCE: > + case ARIZONA_HPLP3MIX_INPUT_2_VOLUME: > + case ARIZONA_HPLP3MIX_INPUT_3_SOURCE: > + case ARIZONA_HPLP3MIX_INPUT_3_VOLUME: > + case ARIZONA_HPLP3MIX_INPUT_4_SOURCE: > + case ARIZONA_HPLP3MIX_INPUT_4_VOLUME: > + case ARIZONA_HPLP4MIX_INPUT_1_SOURCE: > + case ARIZONA_HPLP4MIX_INPUT_1_VOLUME: > + case ARIZONA_HPLP4MIX_INPUT_2_SOURCE: > + case ARIZONA_HPLP4MIX_INPUT_2_VOLUME: > + case ARIZONA_HPLP4MIX_INPUT_3_SOURCE: > + case ARIZONA_HPLP4MIX_INPUT_3_VOLUME: > + case ARIZONA_HPLP4MIX_INPUT_4_SOURCE: > + case ARIZONA_HPLP4MIX_INPUT_4_VOLUME: > + case ARIZONA_DSP2LMIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2LMIX_INPUT_1_VOLUME: > + case ARIZONA_DSP2LMIX_INPUT_2_SOURCE: > + case ARIZONA_DSP2LMIX_INPUT_2_VOLUME: > + case ARIZONA_DSP2LMIX_INPUT_3_SOURCE: > + case ARIZONA_DSP2LMIX_INPUT_3_VOLUME: > + case ARIZONA_DSP2LMIX_INPUT_4_SOURCE: > + case ARIZONA_DSP2LMIX_INPUT_4_VOLUME: > + case ARIZONA_DSP2RMIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2RMIX_INPUT_1_VOLUME: > + case ARIZONA_DSP2RMIX_INPUT_2_SOURCE: > + case ARIZONA_DSP2RMIX_INPUT_2_VOLUME: > + case ARIZONA_DSP2RMIX_INPUT_3_SOURCE: > + case ARIZONA_DSP2RMIX_INPUT_3_VOLUME: > + case ARIZONA_DSP2RMIX_INPUT_4_SOURCE: > + case ARIZONA_DSP2RMIX_INPUT_4_VOLUME: > + case ARIZONA_DSP2AUX1MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2AUX2MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2AUX3MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2AUX4MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2AUX5MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP2AUX6MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3LMIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3LMIX_INPUT_1_VOLUME: > + case ARIZONA_DSP3LMIX_INPUT_2_SOURCE: > + case ARIZONA_DSP3LMIX_INPUT_2_VOLUME: > + case ARIZONA_DSP3LMIX_INPUT_3_SOURCE: > + case ARIZONA_DSP3LMIX_INPUT_3_VOLUME: > + case ARIZONA_DSP3LMIX_INPUT_4_SOURCE: > + case ARIZONA_DSP3LMIX_INPUT_4_VOLUME: > + case ARIZONA_DSP3RMIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3RMIX_INPUT_1_VOLUME: > + case ARIZONA_DSP3RMIX_INPUT_2_SOURCE: > + case ARIZONA_DSP3RMIX_INPUT_2_VOLUME: > + case ARIZONA_DSP3RMIX_INPUT_3_SOURCE: > + case ARIZONA_DSP3RMIX_INPUT_3_VOLUME: > + case ARIZONA_DSP3RMIX_INPUT_4_SOURCE: > + case ARIZONA_DSP3RMIX_INPUT_4_VOLUME: > + case ARIZONA_DSP3AUX1MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3AUX2MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3AUX3MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3AUX4MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3AUX5MIX_INPUT_1_SOURCE: > + case ARIZONA_DSP3AUX6MIX_INPUT_1_SOURCE: > + case ARIZONA_ASRC1LMIX_INPUT_1_SOURCE: > + case ARIZONA_ASRC1RMIX_INPUT_1_SOURCE: > + case ARIZONA_ASRC2LMIX_INPUT_1_SOURCE: > + case ARIZONA_ASRC2RMIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2DEC3MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2DEC4MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2INT3MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC2INT4MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3DEC1MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3INT1MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3INT2MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3INT3MIX_INPUT_1_SOURCE: > + case ARIZONA_ISRC3INT4MIX_INPUT_1_SOURCE: > + case ARIZONA_GPIO1_CTRL: > + case ARIZONA_GPIO2_CTRL: > + case ARIZONA_IRQ_CTRL_1: > + case ARIZONA_GPIO_DEBOUNCE_CONFIG: > + case ARIZONA_MISC_PAD_CTRL_1: > + case ARIZONA_MISC_PAD_CTRL_2: > + case ARIZONA_MISC_PAD_CTRL_3: > + case ARIZONA_MISC_PAD_CTRL_4: > + case ARIZONA_MISC_PAD_CTRL_5: > + case ARIZONA_MISC_PAD_CTRL_6: > + case ARIZONA_MISC_PAD_CTRL_7: > + case ARIZONA_MISC_PAD_CTRL_9: > + case ARIZONA_MISC_PAD_CTRL_10: > + case ARIZONA_MISC_PAD_CTRL_11: > + case ARIZONA_MISC_PAD_CTRL_12: > + case ARIZONA_MISC_PAD_CTRL_13: > + case ARIZONA_MISC_PAD_CTRL_14: > + case ARIZONA_MISC_PAD_CTRL_16: > + case ARIZONA_INTERRUPT_STATUS_1: > + case ARIZONA_INTERRUPT_STATUS_2: > + case ARIZONA_INTERRUPT_STATUS_3: > + case ARIZONA_INTERRUPT_STATUS_4: > + case ARIZONA_INTERRUPT_STATUS_5: > + case ARIZONA_INTERRUPT_STATUS_6: > + case ARIZONA_INTERRUPT_STATUS_1_MASK: > + case ARIZONA_INTERRUPT_STATUS_2_MASK: > + case ARIZONA_INTERRUPT_STATUS_3_MASK: > + case ARIZONA_INTERRUPT_STATUS_4_MASK: > + case ARIZONA_INTERRUPT_STATUS_5_MASK: > + case ARIZONA_INTERRUPT_STATUS_6_MASK: > + case ARIZONA_INTERRUPT_CONTROL: > + case ARIZONA_IRQ2_STATUS_1: > + case ARIZONA_IRQ2_STATUS_2: > + case ARIZONA_IRQ2_STATUS_3: > + case ARIZONA_IRQ2_STATUS_4: > + case ARIZONA_IRQ2_STATUS_5: > + case ARIZONA_IRQ2_STATUS_6: > + case ARIZONA_IRQ2_STATUS_1_MASK: > + case ARIZONA_IRQ2_STATUS_2_MASK: > + case ARIZONA_IRQ2_STATUS_3_MASK: > + case ARIZONA_IRQ2_STATUS_4_MASK: > + case ARIZONA_IRQ2_STATUS_5_MASK: > + case ARIZONA_IRQ2_STATUS_6_MASK: > + case ARIZONA_IRQ2_CONTROL: > + case ARIZONA_INTERRUPT_RAW_STATUS_2: > + case ARIZONA_INTERRUPT_RAW_STATUS_3: > + case ARIZONA_INTERRUPT_RAW_STATUS_4: > + case ARIZONA_INTERRUPT_RAW_STATUS_5: > + case ARIZONA_INTERRUPT_RAW_STATUS_6: > + case ARIZONA_INTERRUPT_RAW_STATUS_7: > + case ARIZONA_INTERRUPT_RAW_STATUS_8: > + case ARIZONA_INTERRUPT_RAW_STATUS_9: > + case ARIZONA_IRQ_PIN_STATUS: > + case ARIZONA_FX_CTRL1: > + case ARIZONA_FX_CTRL2: > + case ARIZONA_EQ1_1: > + case ARIZONA_EQ1_2: > + case ARIZONA_EQ1_3: > + case ARIZONA_EQ1_4: > + case ARIZONA_EQ1_5: > + case ARIZONA_EQ1_6: > + case ARIZONA_EQ1_7: > + case ARIZONA_EQ1_8: > + case ARIZONA_EQ1_9: > + case ARIZONA_EQ1_10: > + case ARIZONA_EQ1_11: > + case ARIZONA_EQ1_12: > + case ARIZONA_EQ1_13: > + case ARIZONA_EQ1_14: > + case ARIZONA_EQ1_15: > + case ARIZONA_EQ1_16: > + case ARIZONA_EQ1_17: > + case ARIZONA_EQ1_18: > + case ARIZONA_EQ1_19: > + case ARIZONA_EQ1_20: > + case ARIZONA_EQ1_21: > + case ARIZONA_EQ2_1: > + case ARIZONA_EQ2_2: > + case ARIZONA_EQ2_3: > + case ARIZONA_EQ2_4: > + case ARIZONA_EQ2_5: > + case ARIZONA_EQ2_6: > + case ARIZONA_EQ2_7: > + case ARIZONA_EQ2_8: > + case ARIZONA_EQ2_9: > + case ARIZONA_EQ2_10: > + case ARIZONA_EQ2_11: > + case ARIZONA_EQ2_12: > + case ARIZONA_EQ2_13: > + case ARIZONA_EQ2_14: > + case ARIZONA_EQ2_15: > + case ARIZONA_EQ2_16: > + case ARIZONA_EQ2_17: > + case ARIZONA_EQ2_18: > + case ARIZONA_EQ2_19: > + case ARIZONA_EQ2_20: > + case ARIZONA_EQ2_21: > + case ARIZONA_DRC1_CTRL1: > + case ARIZONA_DRC1_CTRL2: > + case ARIZONA_DRC1_CTRL3: > + case ARIZONA_DRC1_CTRL4: > + case ARIZONA_DRC1_CTRL5: > + case ARIZONA_DRC2_CTRL1: > + case ARIZONA_DRC2_CTRL2: > + case ARIZONA_DRC2_CTRL3: > + case ARIZONA_DRC2_CTRL4: > + case ARIZONA_DRC2_CTRL5: > + case ARIZONA_HPLPF1_1: > + case ARIZONA_HPLPF1_2: > + case ARIZONA_HPLPF2_1: > + case ARIZONA_HPLPF2_2: > + case ARIZONA_HPLPF3_1: > + case ARIZONA_HPLPF3_2: > + case ARIZONA_HPLPF4_1: > + case ARIZONA_HPLPF4_2: > + case ARIZONA_ASRC_ENABLE: > + case ARIZONA_ASRC_STATUS: > + case ARIZONA_ASRC_RATE1: > + case ARIZONA_ASRC_RATE2: > + case ARIZONA_ISRC_1_CTRL_1: > + case ARIZONA_ISRC_1_CTRL_2: > + case ARIZONA_ISRC_1_CTRL_3: > + case ARIZONA_ISRC_2_CTRL_1: > + case ARIZONA_ISRC_2_CTRL_2: > + case ARIZONA_ISRC_2_CTRL_3: > + case ARIZONA_ISRC_3_CTRL_1: > + case ARIZONA_ISRC_3_CTRL_2: > + case ARIZONA_ISRC_3_CTRL_3: > + case ARIZONA_DSP2_CONTROL_1: > + case ARIZONA_DSP2_CLOCKING_1: > + case ARIZONA_DSP2_STATUS_1: > + case ARIZONA_DSP2_STATUS_2: > + case ARIZONA_DSP2_STATUS_3: > + case ARIZONA_DSP2_STATUS_4: > + case ARIZONA_DSP2_WDMA_BUFFER_1: > + case ARIZONA_DSP2_WDMA_BUFFER_2: > + case ARIZONA_DSP2_WDMA_BUFFER_3: > + case ARIZONA_DSP2_WDMA_BUFFER_4: > + case ARIZONA_DSP2_WDMA_BUFFER_5: > + case ARIZONA_DSP2_WDMA_BUFFER_6: > + case ARIZONA_DSP2_WDMA_BUFFER_7: > + case ARIZONA_DSP2_WDMA_BUFFER_8: > + case ARIZONA_DSP2_RDMA_BUFFER_1: > + case ARIZONA_DSP2_RDMA_BUFFER_2: > + case ARIZONA_DSP2_RDMA_BUFFER_3: > + case ARIZONA_DSP2_RDMA_BUFFER_4: > + case ARIZONA_DSP2_RDMA_BUFFER_5: > + case ARIZONA_DSP2_RDMA_BUFFER_6: > + case ARIZONA_DSP2_WDMA_CONFIG_1: > + case ARIZONA_DSP2_WDMA_CONFIG_2: > + case ARIZONA_DSP2_WDMA_OFFSET_1: > + case ARIZONA_DSP2_RDMA_CONFIG_1: > + case ARIZONA_DSP2_RDMA_OFFSET_1: > + case ARIZONA_DSP2_EXTERNAL_START_SELECT_1: > + case ARIZONA_DSP2_SCRATCH_0: > + case ARIZONA_DSP2_SCRATCH_1: > + case ARIZONA_DSP2_SCRATCH_2: > + case ARIZONA_DSP2_SCRATCH_3: > + case ARIZONA_DSP3_CONTROL_1: > + case ARIZONA_DSP3_CLOCKING_1: > + case ARIZONA_DSP3_STATUS_1: > + case ARIZONA_DSP3_STATUS_2: > + case ARIZONA_DSP3_STATUS_3: > + case ARIZONA_DSP3_STATUS_4: > + case ARIZONA_DSP3_WDMA_BUFFER_1: > + case ARIZONA_DSP3_WDMA_BUFFER_2: > + case ARIZONA_DSP3_WDMA_BUFFER_3: > + case ARIZONA_DSP3_WDMA_BUFFER_4: > + case ARIZONA_DSP3_WDMA_BUFFER_5: > + case ARIZONA_DSP3_WDMA_BUFFER_6: > + case ARIZONA_DSP3_WDMA_BUFFER_7: > + case ARIZONA_DSP3_WDMA_BUFFER_8: > + case ARIZONA_DSP3_RDMA_BUFFER_1: > + case ARIZONA_DSP3_RDMA_BUFFER_2: > + case ARIZONA_DSP3_RDMA_BUFFER_3: > + case ARIZONA_DSP3_RDMA_BUFFER_4: > + case ARIZONA_DSP3_RDMA_BUFFER_5: > + case ARIZONA_DSP3_RDMA_BUFFER_6: > + case ARIZONA_DSP3_WDMA_CONFIG_1: > + case ARIZONA_DSP3_WDMA_CONFIG_2: > + case ARIZONA_DSP3_WDMA_OFFSET_1: > + case ARIZONA_DSP3_RDMA_CONFIG_1: > + case ARIZONA_DSP3_RDMA_OFFSET_1: > + case ARIZONA_DSP3_EXTERNAL_START_SELECT_1: > + case ARIZONA_DSP3_SCRATCH_0: > + case ARIZONA_DSP3_SCRATCH_1: > + case ARIZONA_DSP3_SCRATCH_2: > + case ARIZONA_DSP3_SCRATCH_3: > + return true; > + default: > + return cs47l24_is_adsp_memory(reg); > + } > +} > + > +static bool cs47l24_volatile_register(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case ARIZONA_SOFTWARE_RESET: > + case ARIZONA_DEVICE_REVISION: > + case ARIZONA_WRITE_SEQUENCER_CTRL_0: > + case ARIZONA_WRITE_SEQUENCER_CTRL_1: > + case ARIZONA_WRITE_SEQUENCER_CTRL_2: > + case ARIZONA_HAPTICS_STATUS: > + case ARIZONA_SAMPLE_RATE_1_STATUS: > + case ARIZONA_SAMPLE_RATE_2_STATUS: > + case ARIZONA_SAMPLE_RATE_3_STATUS: > + case ARIZONA_ASYNC_SAMPLE_RATE_1_STATUS: > + case ARIZONA_ASYNC_SAMPLE_RATE_2_STATUS: > + case ARIZONA_HP_CTRL_1L: > + case ARIZONA_HP_CTRL_1R: > + case ARIZONA_INPUT_ENABLES_STATUS: > + case ARIZONA_OUTPUT_STATUS_1: > + case ARIZONA_RAW_OUTPUT_STATUS_1: > + case ARIZONA_INTERRUPT_STATUS_1: > + case ARIZONA_INTERRUPT_STATUS_2: > + case ARIZONA_INTERRUPT_STATUS_3: > + case ARIZONA_INTERRUPT_STATUS_4: > + case ARIZONA_INTERRUPT_STATUS_5: > + case ARIZONA_INTERRUPT_STATUS_6: > + case ARIZONA_IRQ2_STATUS_1: > + case ARIZONA_IRQ2_STATUS_2: > + case ARIZONA_IRQ2_STATUS_3: > + case ARIZONA_IRQ2_STATUS_4: > + case ARIZONA_IRQ2_STATUS_5: > + case ARIZONA_IRQ2_STATUS_6: > + case ARIZONA_INTERRUPT_RAW_STATUS_2: > + case ARIZONA_INTERRUPT_RAW_STATUS_3: > + case ARIZONA_INTERRUPT_RAW_STATUS_4: > + case ARIZONA_INTERRUPT_RAW_STATUS_5: > + case ARIZONA_INTERRUPT_RAW_STATUS_6: > + case ARIZONA_INTERRUPT_RAW_STATUS_7: > + case ARIZONA_INTERRUPT_RAW_STATUS_8: > + case ARIZONA_INTERRUPT_RAW_STATUS_9: > + case ARIZONA_IRQ_PIN_STATUS: > + case ARIZONA_FX_CTRL2: > + case ARIZONA_ASRC_STATUS: > + case ARIZONA_DSP2_STATUS_1: > + case ARIZONA_DSP2_STATUS_2: > + case ARIZONA_DSP2_STATUS_3: > + case ARIZONA_DSP2_STATUS_4: > + case ARIZONA_DSP2_WDMA_BUFFER_1: > + case ARIZONA_DSP2_WDMA_BUFFER_2: > + case ARIZONA_DSP2_WDMA_BUFFER_3: > + case ARIZONA_DSP2_WDMA_BUFFER_4: > + case ARIZONA_DSP2_WDMA_BUFFER_5: > + case ARIZONA_DSP2_WDMA_BUFFER_6: > + case ARIZONA_DSP2_WDMA_BUFFER_7: > + case ARIZONA_DSP2_WDMA_BUFFER_8: > + case ARIZONA_DSP2_RDMA_BUFFER_1: > + case ARIZONA_DSP2_RDMA_BUFFER_2: > + case ARIZONA_DSP2_RDMA_BUFFER_3: > + case ARIZONA_DSP2_RDMA_BUFFER_4: > + case ARIZONA_DSP2_RDMA_BUFFER_5: > + case ARIZONA_DSP2_RDMA_BUFFER_6: > + case ARIZONA_DSP2_WDMA_CONFIG_1: > + case ARIZONA_DSP2_WDMA_CONFIG_2: > + case ARIZONA_DSP2_WDMA_OFFSET_1: > + case ARIZONA_DSP2_RDMA_CONFIG_1: > + case ARIZONA_DSP2_RDMA_OFFSET_1: > + case ARIZONA_DSP2_EXTERNAL_START_SELECT_1: > + case ARIZONA_DSP2_SCRATCH_0: > + case ARIZONA_DSP2_SCRATCH_1: > + case ARIZONA_DSP2_SCRATCH_2: > + case ARIZONA_DSP2_SCRATCH_3: > + case ARIZONA_DSP2_CLOCKING_1: > + case ARIZONA_DSP3_STATUS_1: > + case ARIZONA_DSP3_STATUS_2: > + case ARIZONA_DSP3_STATUS_3: > + case ARIZONA_DSP3_STATUS_4: > + case ARIZONA_DSP3_WDMA_BUFFER_1: > + case ARIZONA_DSP3_WDMA_BUFFER_2: > + case ARIZONA_DSP3_WDMA_BUFFER_3: > + case ARIZONA_DSP3_WDMA_BUFFER_4: > + case ARIZONA_DSP3_WDMA_BUFFER_5: > + case ARIZONA_DSP3_WDMA_BUFFER_6: > + case ARIZONA_DSP3_WDMA_BUFFER_7: > + case ARIZONA_DSP3_WDMA_BUFFER_8: > + case ARIZONA_DSP3_RDMA_BUFFER_1: > + case ARIZONA_DSP3_RDMA_BUFFER_2: > + case ARIZONA_DSP3_RDMA_BUFFER_3: > + case ARIZONA_DSP3_RDMA_BUFFER_4: > + case ARIZONA_DSP3_RDMA_BUFFER_5: > + case ARIZONA_DSP3_RDMA_BUFFER_6: > + case ARIZONA_DSP3_WDMA_CONFIG_1: > + case ARIZONA_DSP3_WDMA_CONFIG_2: > + case ARIZONA_DSP3_WDMA_OFFSET_1: > + case ARIZONA_DSP3_RDMA_CONFIG_1: > + case ARIZONA_DSP3_RDMA_OFFSET_1: > + case ARIZONA_DSP3_EXTERNAL_START_SELECT_1: > + case ARIZONA_DSP3_SCRATCH_0: > + case ARIZONA_DSP3_SCRATCH_1: > + case ARIZONA_DSP3_SCRATCH_2: > + case ARIZONA_DSP3_SCRATCH_3: > + case ARIZONA_DSP3_CLOCKING_1: > + return true; > + default: > + return cs47l24_is_adsp_memory(reg); > + } > +} > + > +#define CS47L24_MAX_REGISTER 0x3b3fff > + > +const struct regmap_config cs47l24_spi_regmap = { > + .reg_bits = 32, > + .pad_bits = 16, > + .val_bits = 16, > + .reg_format_endian = REGMAP_ENDIAN_BIG, > + .val_format_endian = REGMAP_ENDIAN_BIG, > + > + .max_register = CS47L24_MAX_REGISTER, > + .readable_reg = cs47l24_readable_register, > + .volatile_reg = cs47l24_volatile_register, > + > + .cache_type = REGCACHE_RBTREE, > + .reg_defaults = cs47l24_reg_default, > + .num_reg_defaults = ARRAY_SIZE(cs47l24_reg_default), > +}; > +EXPORT_SYMBOL_GPL(cs47l24_spi_regmap); > + > diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h > index 79e607e..d55a422 100644 > --- a/include/linux/mfd/arizona/core.h > +++ b/include/linux/mfd/arizona/core.h > @@ -27,6 +27,8 @@ enum arizona_type { > WM8280 = 4, > WM8998 = 5, > WM1814 = 6, > + WM1831 = 7, > + CS47L24 = 8, > }; > > #define ARIZONA_IRQ_GP1 0 > @@ -166,6 +168,7 @@ static inline int wm5102_patch(struct arizona *arizona) > #endif > > int wm5110_patch(struct arizona *arizona); > +int cs47l24_patch(struct arizona *arizona); > int wm8997_patch(struct arizona *arizona); > int wm8998_patch(struct arizona *arizona); > -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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[alsa-devel] Simple-card without codec for testing purpose
by arnaud.mouiche＠invoxia.com 24 Nov '15

24 Nov '15

Hello, I'm starting debugging / testing all the fsl_ssi issues (see Caleb's previous post). First step I just need to setup a DTS entry for "simple-card" + fake codec to generate arbitrary PCM bus output (1 to 16 channels, various TDM, various sampling rate and format and master/slave role) I will then plug my logical analyser, or do some loopback, or plug a FPGA to generate/check some patterns. Is there such "fake codec" somewhere. Indeed, something that look like the bt-sco codec, but without rate/format/channel limitations ? Or does the "simple-card" even needs one ? Regards, Arnaud

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[alsa-devel] [PATCH 0/5] Add acpi_dev_present
by Lukas Wunner 24 Nov '15

24 Nov '15

Hi everyone, I need to detect the presence of apple-gmux (ACPI HID "APP000B") in several DRM drivers to handle backlight, runtime pm and deferred probing properly. There's an idiom in use by 7 other drivers to detect presence of a particular ACPI HID which involves calling acpi_get_devices() and passing it a minimal callback that does no further filtering. However this approach results in lots of duplicate code. This series adds acpi_dev_present(), the ACPI equivalent to pci_dev_present(). It takes a HID string and returns a bool, allowing us to simplify the drivers considerably by removing all the duplicate callbacks and other boilerplate. I've pushed these patches to GitHub to ease reviewing: https://github.com/l1k/linux/commits/acpi_dev_present Should this series find acceptance, it would probably be best if we could agree to merge everything through a single repository, e.g. acpi (git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm). or alternatively alsa (which contains 5 of the 7 drivers involved). Consequently I would like to invite the driver maintainers to kindly provide acks (or raise objections, if any). The series should not result in any functional changes, however I lack the hardware to actually verify this for every driver involved. I did test it with apple-gmux though. Thanks, Lukas Lukas Wunner (5): ACPICA: Add acpi_dev_present eeepc-wmi: Use acpi_dev_present acer-wmi: Use acpi_dev_present ALSA: hda - Use acpi_dev_present ASoC: Intel: Use acpi_dev_present drivers/acpi/acpica/nsxfeval.c | 46 ++++++++++++++++++++++++++++ drivers/platform/x86/acer-wmi.c | 16 +++------- drivers/platform/x86/eeepc-wmi.c | 24 ++------------- include/acpi/acpixf.h | 7 +++++ sound/pci/hda/thinkpad_helper.c | 17 ++-------- sound/soc/intel/atom/sst/sst_acpi.c | 12 +------- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 17 ++-------- sound/soc/intel/boards/cht_bsw_rt5645.c | 13 +------- sound/soc/intel/common/sst-acpi.c | 12 +------- 9 files changed, 66 insertions(+), 98 deletions(-) -- 1.8.5.2 (Apple Git-48)

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[alsa-devel] [PATCH v1 1/1] sst: replace custom implementation of readq / writeq
by Andy Shevchenko 24 Nov '15

24 Nov '15

The readq() and writeq() helpers are available in the asm-generic/io-64-nonatomic-hi-lo.h and asm-generic/io-64-nonatomic-lo-hi.h headers. Replace custom implementation by the generic helpers. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_pvt.c | 16 ++++++---------- sound/soc/intel/common/sst-dsp.c | 9 ++++----- 2 files changed, 10 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fe..6cc222d 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -31,7 +31,10 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> + +#include <asm-generic/io-64-nonatomic-lo-hi.h> #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" #include "../../common/sst-dsp.h" @@ -49,25 +52,18 @@ u32 sst_shim_read(void __iomem *addr, int offset) u64 sst_reg_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - - return val; + return readq(addr + offset); } int sst_shim_write64(void __iomem *addr, int offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); return 0; } u64 sst_shim_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } void sst_set_fw_state_locked( diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236..5274482 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -22,6 +22,8 @@ #include <linux/io.h> #include <linux/delay.h> +#include <asm-generic/io-64-nonatomic-lo-hi.h> + #include "sst-dsp.h" #include "sst-dsp-priv.h" @@ -43,16 +45,13 @@ EXPORT_SYMBOL_GPL(sst_shim32_read); void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_write64); u64 sst_shim32_read64(void __iomem *addr, u32 offset) { - u64 val; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_read64); -- 2.5.0

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[alsa-devel] [PATCH] ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3
by Hui Wang 24 Nov '15

24 Nov '15

We have a machine Dell XPS 13 with the codec alc256, after resume back from S3, the headphone has noise when play sound. Through comparing with the coeff vaule before and after S3, we found restoring a coeff register will help remove noise. BugLink: https://bugs.launchpad.net/bugs/1519168 Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- Kailang said one of the bits in that register is: Enable/Disable pass-through path for FRONT 14h And disable this bit should help remove headphone noise. sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53f6a02..e4f80dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4587,6 +4587,7 @@ enum { ALC292_FIXUP_DISABLE_AAMIX, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5167,6 +5168,17 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x36}, + {0x20, AC_VERB_SET_PROC_COEF, 0x1737}, + {} + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5206,6 +5218,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fix kernel-doc warnings in sound/soc/soc-ops.c
by Randy Dunlap 24 Nov '15

24 Nov '15

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc warnings in soc-ops.c: ..//sound/soc/soc-ops.c:415: warning: No description found for parameter 'ucontrol' ..//sound/soc/soc-ops.c:415: warning: Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-44-rc2.orig/sound/soc/soc-ops.c +++ lnx-44-rc2/sound/soc/soc-ops.c @@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. *

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