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[alsa-devel] [PATCH v7 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark, Rob These are v7 of OF-graph sound card support. Main change from v6 "label" doesn't allow [prefix]label (in [6/9]) 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): of-graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 73 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 600 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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

17 May '17

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

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup Document
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark Cc Geert These patches tidyup Document for Renesas sound driver Kuninori Morimoto (2): ASoC: rsnd: move "renesas,rsrc-card" to "simple-scu-audio-card" on Document ASoC: rsnd: add missing clocks/clock-names on Document .../devicetree/bindings/sound/renesas,rsnd.txt | 34 ++++++++++++---------- 1 file changed, 19 insertions(+), 15 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: hdmi-codec: fix spelling mistake: "deteced" -> "detected"
by Colin King 17 May '17

17 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..a3f15149afcf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -819,7 +819,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) mutex_unlock(&hdmi_mutex); if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) { - dev_err(dev, "too many hdmi codec are deteced\n"); + dev_err(dev, "too many hdmi codec are detected\n"); return -EINVAL; } -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rsnd: remove un-used OUF_SRCO/I macro
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 20b5b2e..a8ab26d 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -12,10 +12,6 @@ #define SRC_NAME "src" -/* SRCx_STATUS */ -#define OUF_SRCO ((1 << 12) | (1 << 13)) -#define OUF_SRCI ((1 << 9) | (1 << 8)) - /* SCU_SYSTEM_STATUS0/1 */ #define OUF_SRC(id) ((1 << (id + 16)) | (1 << id)) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: remove mismatch explain comment from src.c
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Before, src.c cares SRC and SSIU, but now SSIU is cared by ssiu.c by commit c7f69ab5364d ("ASoC: rsnd: use mod base common method on SSIU"). This patch removes mismatched explanation for SRC/SSIU from src.c Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a8ab26d..e4bde0c 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -51,20 +51,6 @@ struct rsnd_src { * */ -/* - * src.c is caring... - * - * Gen1 - * - * [mem] -> [SRU] -> [SSI] - * |--------| - * - * Gen2 - * - * [mem] -> [SRC] -> [SSIU] -> [SSI] - * |-----------------| - */ - static void rsnd_src_activation(struct rsnd_mod *mod) { rsnd_mod_write(mod, SRC_SWRSR, 0); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: clkout-lr-asynchronous is AUDIO_CLKOUT related property
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> clkout-lr-asynchronous is AUDIO_CLKOUT related property, thus, it should be handled by rsnd_adg_get_clkout(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 66203d1..cb4eab0 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -480,6 +480,9 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, if (req_rate[0] % 48000 == 0) adg->flags = AUDIO_OUT_48; + if (of_get_property(np, "clkout-lr-asynchronous", NULL)) + adg->flags = LRCLK_ASYNC; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. @@ -578,7 +581,6 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *np = dev->of_node; int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); @@ -595,9 +597,6 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); - if (of_get_property(np, "clkout-lr-asynchronous", NULL)) - adg->flags = LRCLK_ASYNC; - priv->adg = adg; rsnd_adg_clk_enable(priv); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: remove error pointer for adg->clkout[i]
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit d7f298197a22f11b38059f257842dac7c30a564c ("ASoC: rcar: fixup of_clk_add_provider() usage for multi clkout") added adg->clkout[i] = ERR_PTR(-ENOENT), but, adg->clkout[i] are assumed NULL or clk pointer. This patch fix it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index cb4eab0..36bece8 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -556,7 +556,6 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, req_rate[0]); - adg->clkout[i] = ERR_PTR(-ENOENT); if (!IS_ERR(clk)) adg->clkout[i] = clk; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: need to call nolock_stop if nolock_start was failed
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_dai_call() is counting the called number of start/stop functions. nolock_stop should be called if nolock_start was failed. Otherwise, nolock_start doesn't called in 2nd play. This patch solved this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8c1f4e2..e16f723 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -766,11 +766,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + int ret; /* * call rsnd_dai_call without spinlock */ - return rsnd_dai_call(nolock_start, io, priv); + ret = rsnd_dai_call(nolock_start, io, priv); + if (ret < 0) + rsnd_dai_call(nolock_stop, io, priv); + + return ret; } static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, -- 1.9.1

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Re: [alsa-devel] ASoC: rsnd: fix sound route path when using SRC6/SRC9
by Geert Uytterhoeven 17 May '17

17 May '17

Hi Morimoto-san, On Fri, Mar 31, 2017 at 9:32 PM, Linux Kernel Mailing List <linux-kernel(a)vger.kernel.org> wrote: > Web: https://git.kernel.org/torvalds/c/a1c2ff53726907aff5feb37e4cfd45c1ff626431 > Commit: a1c2ff53726907aff5feb37e4cfd45c1ff626431 > Parent: 4b30eebfc35c67771b5f58d9274d3e321b72d7a8 > Refname: refs/heads/master > Author: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> > AuthorDate: Wed Mar 1 03:51:00 2017 +0000 > Committer: Mark Brown <broonie(a)kernel.org> > CommitDate: Mon Mar 13 12:58:07 2017 +0000 > > ASoC: rsnd: fix sound route path when using SRC6/SRC9 > > This patch fixes the problem that the missing value of the route path > setting table and incorrect values are set in the CMD_ROUTE_SELECT > register. > > Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> > [Kuninori: shared data on MIX and non-MIX case] > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > --- > sound/soc/sh/rcar/cmd.c | 36 ++++++++++++++++++++---------------- > 1 file changed, 20 insertions(+), 16 deletions(-) > > diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c > index abb5eaac854a..7d92a24b7cfa 100644 > --- a/sound/soc/sh/rcar/cmd.c > +++ b/sound/soc/sh/rcar/cmd.c > @@ -31,23 +31,24 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, > struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); > struct device *dev = rsnd_priv_to_dev(priv); > u32 data; > + u32 path[] = { > + [1] = 1 << 0, > + [5] = 1 << 8, > + [6] = 1 << 12, > + [9] = 1 << 15, > + }; > > if (!mix && !dvc) > return 0; > > + if (ARRAY_SIZE(path) < rsnd_mod_id(mod) + 1) > + return -ENXIO; > + > if (mix) { > struct rsnd_dai *rdai; > struct rsnd_mod *src; > struct rsnd_dai_stream *tio; > int i; > - u32 path[] = { > - [0] = 0, > - [1] = 1 << 0, > - [2] = 0, > - [3] = 0, > - [4] = 0, > - [5] = 1 << 8 > - }; > > /* > * it is assuming that integrater is well understanding about > @@ -70,16 +71,19 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, > } else { > struct rsnd_mod *src = rsnd_io_to_mod_src(io); > > - u32 path[] = { > - [0] = 0x30000, > - [1] = 0x30001, > - [2] = 0x40000, > - [3] = 0x10000, > - [4] = 0x20000, > - [5] = 0x40100 > + u8 cmd_case[] = { > + [0] = 0x3, > + [1] = 0x3, > + [2] = 0x4, > + [3] = 0x1, > + [4] = 0x2, > + [5] = 0x4, > + [6] = 0x1, > + [9] = 0x2, > }; > > - data = path[rsnd_mod_id(src)]; > + data = path[rsnd_mod_id(src)] | With gcc 4.9.0: sound/soc/sh/rcar/cmd.c: In function 'rsnd_cmd_init': sound/soc/sh/rcar/cmd.c:85:14: warning: array subscript is below array bounds [-Warray-bounds] data = path[rsnd_mod_id(src)] | > + cmd_case[rsnd_mod_id(src)] << 16; > } > > dev_dbg(dev, "ctu/mix path = 0x%08x", data); 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 -next] ASoC: rt5665: make local symbol rt5665_i2c_driver static
by Wei Yongjun 17 May '17

17 May '17

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fixes the following sparse warnings: sound/soc/codecs/rt5665.c:4928:19: warning: symbol 'rt5665_i2c_driver' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 26bf157..bb440bd 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4925,7 +4925,7 @@ static struct acpi_device_id rt5665_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5665_acpi_match); #endif -struct i2c_driver rt5665_i2c_driver = { +static struct i2c_driver rt5665_i2c_driver = { .driver = { .name = "rt5665", .of_match_table = of_match_ptr(rt5665_of_match),

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[alsa-devel] [PATCH] ASoC: fsi: Move inline fsi_stream_is_play() before use
by Geert Uytterhoeven 17 May '17

17 May '17

With gcc 4.1.2: sound/soc/sh/fsi.c:304: warning: ‘fsi_stream_is_play’ declared inline after being called sound/soc/sh/fsi.c:304: warning: previous declaration of ‘fsi_stream_is_play’ was here Move fsi_stream_is_play() up to fix this, removing the need for a forward declaration as well. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/sh/fsi.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ead520182e2684fb..7c4bdd82bb955bbb 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -301,7 +301,12 @@ struct fsi_master { spinlock_t lock; }; -static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io); +static inline int fsi_stream_is_play(struct fsi_priv *fsi, + struct fsi_stream *io) +{ + return &fsi->playback == io; +} + /* * basic read write function @@ -489,12 +494,6 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi) /* * fsi_stream_xx() function */ -static inline int fsi_stream_is_play(struct fsi_priv *fsi, - struct fsi_stream *io) -{ - return &fsi->playback == io; -} - static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, struct snd_pcm_substream *substream) { -- 2.7.4

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[alsa-devel] [PATCH] ASoC: atmel: Use IS_ENABLED()
by Takashi Iwai 17 May '17

17 May '17

Simplify the ifdef conditions with IS_ENABLED() macro. No functional changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/atmel/atmel-pcm.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 6eaf081cad50..4b27aed40a51 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -83,8 +83,7 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -#if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ - defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) +#if IS_ENABLED(CONFIG_SND_ATMEL_SOC_PDC) int atmel_pcm_pdc_platform_register(struct device *dev); void atmel_pcm_pdc_platform_unregister(struct device *dev); #else @@ -97,8 +96,7 @@ static inline void atmel_pcm_pdc_platform_unregister(struct device *dev) } #endif -#if defined(CONFIG_SND_ATMEL_SOC_DMA) || \ - defined(CONFIG_SND_ATMEL_SOC_DMA_MODULE) +#if IS_ENABLED(CONFIG_SND_ATMEL_SOC_DMA) int atmel_pcm_dma_platform_register(struct device *dev); void atmel_pcm_dma_platform_unregister(struct device *dev); #else -- 2.12.2

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[alsa-devel] [PATCH] ASoC: rsnd: don't use PDTA bit for 24bit on SSI
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SSI uses PDTA bit which indicates data that Input/Output data are Right-Aligned. But, 24bit sound should be Left-Aligned in this HW. Because Linux is using Right-Aligned data, and HW uses Left-Aligned data, current 24bit data is missing lower 8bit. To fix this issue, this patch removes PDTA bit, and shift 8bit in necessary module Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/cmd.c | 1 + sound/soc/sh/rcar/core.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/src.c | 12 ++++++++++-- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rcar/ssiu.c | 3 ++- 7 files changed, 68 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 7d92a24..d879c01 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -89,6 +89,7 @@ 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_MODE, rsnd_get_busif_shift(io, mod) | 1); 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 e36a6b5..6264b39 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -343,6 +343,57 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) return 0x76543210; } +u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) +{ + enum rsnd_mod_type playback_mods[] = { + RSND_MOD_SRC, + RSND_MOD_CMD, + RSND_MOD_SSIU, + }; + enum rsnd_mod_type capture_mods[] = { + RSND_MOD_CMD, + RSND_MOD_SRC, + RSND_MOD_SSIU, + }; + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_mod *tmod = NULL; + enum rsnd_mod_type *mods = + rsnd_io_is_play(io) ? + playback_mods : capture_mods; + int i; + + /* + * This is needed for 24bit data + * We need to shift 8bit + * + * Linux 24bit data is located as 0x00****** + * HW 24bit data is located as 0x******00 + * + */ + switch (runtime->sample_bits) { + case 16: + return 0; + case 32: + break; + } + + for (i = 0; i < ARRAY_SIZE(playback_mods); i++) { + tmod = rsnd_io_to_mod(io, mods[i]); + if (tmod) + break; + } + + if (tmod != mod) + return 0; + + if (rsnd_io_is_play(io)) + return (0 << 20) | /* shift to Left */ + (8 << 16); /* 8bit */ + else + return (1 << 20) | /* shift to Right */ + (8 << 16); /* 8bit */ +} + /* * rsnd_dai functions */ diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 99f93a17..ee00e35 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -238,6 +238,7 @@ 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_MODE, 0x184, 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), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index af6d57b..cc929fc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -74,6 +74,7 @@ enum rsnd_reg { RSND_REG_SCU_SYS_INT_EN0, RSND_REG_SCU_SYS_INT_EN1, RSND_REG_CMD_CTRL, + RSND_REG_CMD_BUSIF_MODE, RSND_REG_CMD_BUSIF_DALIGN, RSND_REG_CMD_ROUTE_SLCT, RSND_REG_CMDOUT_TIMSEL, @@ -207,6 +208,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io); +u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); /* * R-Car DMA diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e395fcb..7aa239e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -172,11 +172,13 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + int is_play = rsnd_io_is_play(io); int use_src = 0; u32 fin, fout; u32 ifscr, fsrate, adinr; u32 cr, route; u32 bsdsr, bsisr; + u32 i_busif, o_busif, tmp; uint ratio; if (!runtime) @@ -252,6 +254,11 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, break; } + /* BUSIF_MODE */ + tmp = rsnd_get_busif_shift(io, mod); + i_busif = ( is_play ? tmp : 0) | 1; + o_busif = (!is_play ? tmp : 0) | 1; + rsnd_mod_write(mod, SRC_ROUTE_MODE0, route); rsnd_mod_write(mod, SRC_SRCIR, 1); /* initialize */ @@ -263,8 +270,9 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, rsnd_mod_write(mod, SRC_BSISR, bsisr); rsnd_mod_write(mod, SRC_SRCIR, 0); /* cancel initialize */ - rsnd_mod_write(mod, SRC_I_BUSIF_MODE, 1); - rsnd_mod_write(mod, SRC_O_BUSIF_MODE, 1); + rsnd_mod_write(mod, SRC_I_BUSIF_MODE, i_busif); + rsnd_mod_write(mod, SRC_O_BUSIF_MODE, o_busif); + rsnd_mod_write(mod, SRC_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); rsnd_adg_set_src_timesel_gen2(mod, io, fin, fout); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d0602c1..5960042 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -319,7 +319,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, * always use 32bit system word. * see also rsnd_ssi_master_clk_enable() */ - cr_own = FORCE | SWL_32 | PDTA; + cr_own = FORCE | SWL_32; if (rdai->bit_clk_inv) cr_own |= SCKP; diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 13d648e..bed2c9c 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -145,7 +145,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, (rsnd_io_is_play(io) ? rsnd_runtime_channel_after_ctu(io) : rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE, 1); + rsnd_mod_write(mod, SSI_BUSIF_MODE, + rsnd_get_busif_shift(io, mod) | 1); rsnd_mod_write(mod, SSI_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Add ACPI match ID
by Oder Chiou 17 May '17

17 May '17

This patch adds the ACPI match ID for rt5514 codec. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 6856b8c..1b6796c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include <linux/acpi.h> #include <linux/fs.h> #include <linux/module.h> #include <linux/moduleparam.h> @@ -1095,6 +1096,14 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5514_acpi_match[] = { + { "10EC5514", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match); +#endif + static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", @@ -1198,6 +1207,7 @@ static const struct dev_pm_ops rt5514_i2_pm_ops = { static struct i2c_driver rt5514_i2c_driver = { .driver = { .name = "rt5514", + .acpi_match_table = ACPI_PTR(rt5514_acpi_match), .of_match_table = of_match_ptr(rt5514_of_match), .pm = &rt5514_i2_pm_ops, }, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: atmel-classd: sync regcache when resuming
by Quentin Schulz 17 May '17

17 May '17

The PM functions used in this driver are the ones defined in sounc/soc/soc-core.c. When suspending (using snd_soc_suspend), the regcache is marked dirty but is never synced on resume. Sync regcache on resume of Atmel ClassD device. Signed-off-by: Quentin Schulz <quentin.schulz(a)free-electrons.com> --- sound/soc/atmel/atmel-classd.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 7ae46c2647d4..b7ef8c59b49a 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -301,6 +301,14 @@ static int atmel_classd_codec_probe(struct snd_soc_codec *codec) return 0; } +static int atmel_classd_codec_resume(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec); + struct atmel_classd *dd = snd_soc_card_get_drvdata(card); + + return regcache_sync(dd->regmap); +} + static struct regmap *atmel_classd_codec_get_remap(struct device *dev) { return dev_get_regmap(dev, NULL); @@ -308,6 +316,7 @@ static struct regmap *atmel_classd_codec_get_remap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_classd = { .probe = atmel_classd_codec_probe, + .resume = atmel_classd_codec_resume, .get_regmap = atmel_classd_codec_get_remap, .component_driver = { .controls = atmel_classd_snd_controls, -- 2.11.0

4 3

[alsa-devel] [PATCH v2 0/2] ALSA, dmaengine: dw: Remove AVR32 bits in drivers
by Andy Shevchenko 17 May '17

17 May '17

AVR32 is gone. Remove bits in sound/atmel and drivers/dma/dw folders related to it. Sound part hasn't been tested anyhow. Changes since v1: - fix kbuild bot complainings - append Nicolas' Ack - don't forget to send to all TWIMC Andy Shevchenko (2): ALSA: atmel: Remove AVR32 bits from the driver dmaengine: dw: Remove AVR32 bits from the driver drivers/dma/dw/Kconfig | 7 +- drivers/dma/dw/core.c | 332 +-------------------------- drivers/dma/dw/regs.h | 50 +--- include/linux/dma/dw.h | 21 -- sound/atmel/Kconfig | 13 +- sound/atmel/Makefile | 2 - sound/atmel/abdac.c | 610 ------------------------------------------------- sound/atmel/ac97c.c | 447 ++++++------------------------------ 8 files changed, 88 insertions(+), 1394 deletions(-) delete mode 100644 sound/atmel/abdac.c -- 2.11.0

5 10

[alsa-devel] [PATCH] ALSA: pcm: add const qualifier for read-only table for sampling rate
by Takashi Sakamoto 17 May '17

17 May '17

There's a read-only table for each sampling rate, while it doesn't have const qualifier and can be modified. This commit add the qualifier. As a result, a symbol for the table moves from .data section to .rodata. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/pcm_native.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d873f90..f3a3580 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1987,8 +1987,10 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, #error "Change this table" #endif -static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; +static const unsigned int rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000 +}; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), -- 2.9.3

2 4

[alsa-devel] HDA cs8409 codec support
by Joël Krähemann 17 May '17

17 May '17

Hi all First of all, I am new to kernel programming and an other attempt already failed to do so. However I got familiar with the Intel HDA Codec specification. So I did a start script to log dmesg and mixer controls of different configurations. But I am not sure if I got a functional kernel driver setup to test things. It seems there is something wrong. The codec has 8 GPIOs and first I didn't set any mask within the kernel. Now, I just compile a kernel set it to 0xff. Any help is appreciated. Finally here is my work: https://github.com/joelkraehemann/hda-tool/ Note: I have tried to subscribe using a different email address but didn't get approved message. Thank you, Joël

2 10

[alsa-devel] [PATCH v2 15/53] docs-rst: add sound book to pdf output
by Mauro Carvalho Chehab 16 May '17

16 May '17

The sound subsystem book was added without the bits required to generate PDF output. Add them. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> --- Documentation/conf.py | 2 ++ Documentation/sound/conf.py | 10 ++++++++++ 2 files changed, 12 insertions(+) create mode 100644 Documentation/sound/conf.py diff --git a/Documentation/conf.py b/Documentation/conf.py index 1d461f0c1cd0..bca751960dec 100644 --- a/Documentation/conf.py +++ b/Documentation/conf.py @@ -371,6 +371,8 @@ latex_documents = [ 'The kernel development community', 'manual'), ('security/index', 'security.tex', 'The kernel security subsystem manual', 'The kernel development community', 'manual'), + ('sound/index', 'sound.tex', 'Linux Sound Subsystem Documentation', + 'The kernel development community', 'manual'), ] # The name of an image file (relative to this directory) to place at the top of diff --git a/Documentation/sound/conf.py b/Documentation/sound/conf.py new file mode 100644 index 000000000000..3f1fc5e74e7b --- /dev/null +++ b/Documentation/sound/conf.py @@ -0,0 +1,10 @@ +# -*- coding: utf-8; mode: python -*- + +project = "Linux Sound Subsystem Documentation" + +tags.add("subproject") + +latex_documents = [ + ('index', 'sound.tex', project, + 'The kernel development community', 'manual'), +] -- 2.9.3

2 1

[alsa-devel] [PATCH alsa-lib 1/2] build: Define __USE_UNIX98 for old glibc
by Takashi Iwai 16 May '17

16 May '17

Otherwise PTHREAD_MUTEX_RECURSIVE isn't defined and we get an error with old glibc. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_local.h | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 32e6dcdf6fcd..e4f652187c49 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -35,6 +35,7 @@ #include "local.h" #ifdef THREAD_SAFE_API +#define __USE_UNIX98 1 /* for old glibc */ #include <pthread.h> #endif -- 2.13.0

1 1

[alsa-devel] [PATCH RFC 00/21] ALSA: pcm: add tracepoints for PCM params operation
by Takashi Sakamoto 16 May '17

16 May '17

Hi, In the last Audio Mini Conference held with Linux Plumber conference 2016[1], I mentioned about tracepoints for PCM params operation. This patchset is for the idea. In ALSA PCM interface, applications can get hardware capability by ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCT_HW_PARAMS in a shape of 'struct snd_pcm_hw_params'. In kernel side, relevant processing is somewhat complicated and developers sometimes have hard time to debug drivers for PCM constraints and rules. This patchset adds tracepoints for hw_params operations. When CONFIG_SND_DEBUG is enabled, you can see 'trace_hw_params_mask/trace_hw_params_interval' events of 'snd_pcm' subsystem. When applications execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCTL_HW_PARAMS, these events are probed. Developers can get how many PCM rules are added into runtime of PCM substream and which rule changed which parameters. This patchset also includes some improvements. The last three commits brings small changes to kernel/userspace interface for error handling. I'm happy to receive your comment for this patchset. For your information, low level application of SNDRV_PCM_IOCTL_HW_REFINE operation is available in my github repository[2]. [1] You can see minutes. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-November/114279.h… [2] refine-pcm-param.c https://github.com/takaswie/alsa-ioctl-test Takashi Sakamoto (21): ALSA: pcm: add a helper function to constrain mask-type parameters ALSA: pcm: add a helper function to constrain interval-type parameters ALSA: pcm: add a helper function to apply parameter rules ALSA: pcm: tracepoints for refining PCM parameters ALSA: pcm: enable parameter tracepoints at CONFIG_SND_DEBUG only ALSA: pcm: use goto statement instead of while statement to reduce indentation ALSA: pcm: remove function local variable with alternative evaluation ALSA: pcm: adaption of code formatting ALSA: pcm: add comment about application of rule to PCM parameters ALSA: pcm: constify function local and read-only table ALSA: pcm: localize snd_pcm_hw_params_choose() ALSA: pcm: add tracepoints for selection process of hardware parameters at SNDRV_PCM_IOCTL_HW_PARAMS ALSA: pcm: use helper functions to check whether parameters are determined ALSA: pcm: use helper function to refer parameter as read-only ALSA: pcm: add const qualifier for read-only table for sampling rate ALSA: pcm/oss: refer to parameters instead of copying to reduce usage of kernel stack ALSA: pcm: check type of parameter in added rule ALSA: pcm: calculate non-mask/non-interval parameters always when possible ALSA: pcm: move fixup of info flag after selecting single parameters ALSA: pcm: return error immediately for parameters refinement ALSA: pcm: return error immediately at refine ioctl include/sound/pcm.h | 1 - sound/core/Makefile | 1 + sound/core/oss/pcm_oss.c | 20 +- sound/core/oss/pcm_plugin.c | 5 +- sound/core/oss/pcm_plugin.h | 2 +- sound/core/pcm_drm_eld.c | 8 +- sound/core/pcm_lib.c | 55 +---- sound/core/pcm_native.c | 485 ++++++++++++++++++++++++++++-------------- sound/core/pcm_params_trace.h | 142 +++++++++++++ 9 files changed, 494 insertions(+), 225 deletions(-) create mode 100644 sound/core/pcm_params_trace.h -- 2.9.3

2 27

[alsa-devel] [PATCH 0/6] Cleanup and removal of set_fs() from HD-audio
by Takashi Iwai 16 May '17

16 May '17

Hi, this is the resubmission of a part of the previous RFC patchset. There is no change since then for them, it's just officially submitted for merging to 4.13. Takashi === Takashi Iwai (6): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions sound/pci/hda/hda_codec.c | 251 ++++++------------------------------------ sound/pci/hda/hda_generic.c | 46 +++++++- sound/pci/hda/hda_local.h | 61 ---------- sound/pci/hda/patch_realtek.c | 37 +++++-- 4 files changed, 103 insertions(+), 292 deletions(-) -- 2.12.2

2 7

[alsa-devel] [PATCH 0/2] Remove set_fs() from drivers
by Takashi Iwai 16 May '17

16 May '17

Hi, this is again a resubmission of a part of the previous RFC patchset. No changes for them, just submitted as the official patches for merging to 4.13. Takashi === Takashi Iwai (2): ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage sound/drivers/opl3/opl3_oss.c | 14 ----- sound/pci/emu10k1/emufx.c | 127 ++++++++++++++++++++++++------------------ 2 files changed, 73 insertions(+), 68 deletions(-) -- 2.12.2

2 3

[alsa-devel] [PATCH 00/13] Get rid of DocBook
by Mauro Carvalho Chehab 16 May '17

16 May '17

As just one book (lsm) was missing conversion, let's convert it and store as if it were a plain text file under Documentation/lsm.txt, adding a notice that it requires update. This way, as everything is now converted, we can get rid of the DocBook building system and update places that were mentioning it. PS.: We could also remove the parts of the kernel-doc script that produce DocBook outputs, but, as this might still be useful for someone, for now let's just keep the functionality there. - This patch series is based on my past 00/05 and 00/36 patch series that convert the other DocBooks to ReST, applied on the top of docs tree (next branch). The full patch series is on this tree is at: https://git.linuxtv.org//mchehab/experimental.git/log/?h=docbook2 And the HTML output at: http://www.infradead.org/~mchehab/kernel_docs/ https://mchehab.fedorapeople.org/kernel_docs/ Mauro Carvalho Chehab (13): docs-rst: convert lsm from DocBook to ReST docs: remove DocBook from the building system docs: update old references for DocBook from the documentation MAINTAINERS: update old references for DocBook directory ata: update references for libata documentation ia64, scsi: update references for the device-io book irq: update genericirq book location fs: update location of filesystems documentation lib: update location of kgdb documentation sound: fix the comments that refers to kernel-doc fs: fix the location of the kernel-api book usb: fix the comment with regards to DocBook docs-rst: get rid of Documentation/sphinx/tmplcvt script Documentation/00-INDEX | 6 +- Documentation/DocBook/.gitignore | 17 -- Documentation/DocBook/Makefile | 276 ----------------------------- Documentation/DocBook/lsm.tmpl | 265 --------------------------- Documentation/DocBook/stylesheet.xsl | 11 -- Documentation/Makefile | 125 +++++++++++++ Documentation/Makefile.sphinx | 130 -------------- Documentation/PCI/MSI-HOWTO.txt | 2 +- Documentation/admin-guide/README.rst | 6 - Documentation/doc-guide/docbook.rst | 90 ---------- Documentation/doc-guide/index.rst | 1 - Documentation/doc-guide/sphinx.rst | 5 - Documentation/fb/api.txt | 4 +- Documentation/gpu/todo.rst | 2 +- Documentation/kernel-doc-nano-HOWTO.txt | 65 ++----- Documentation/lsm.txt | 201 +++++++++++++++++++++ Documentation/process/changes.rst | 26 +-- Documentation/process/howto.rst | 8 - Documentation/process/kernel-docs.rst | 34 +--- Documentation/sphinx/tmplcvt | 28 --- Documentation/translations/ja_JP/howto.rst | 7 - Documentation/translations/ko_KR/howto.rst | 7 - MAINTAINERS | 5 +- Makefile | 11 +- arch/ia64/include/asm/io.h | 2 +- arch/ia64/sn/kernel/iomv.c | 2 +- drivers/ata/acard-ahci.c | 2 +- drivers/ata/ahci.c | 2 +- drivers/ata/ahci.h | 2 +- drivers/ata/ata_piix.c | 2 +- drivers/ata/libahci.c | 2 +- drivers/ata/libata-core.c | 2 +- drivers/ata/libata-eh.c | 2 +- drivers/ata/libata-scsi.c | 2 +- drivers/ata/libata-sff.c | 2 +- drivers/ata/libata.h | 2 +- drivers/ata/pata_pdc2027x.c | 2 +- drivers/ata/pdc_adma.c | 2 +- drivers/ata/sata_nv.c | 2 +- drivers/ata/sata_promise.c | 2 +- drivers/ata/sata_promise.h | 2 +- drivers/ata/sata_qstor.c | 2 +- drivers/ata/sata_sil.c | 2 +- drivers/ata/sata_sis.c | 2 +- drivers/ata/sata_svw.c | 2 +- drivers/ata/sata_sx4.c | 2 +- drivers/ata/sata_uli.c | 2 +- drivers/ata/sata_via.c | 2 +- drivers/ata/sata_vsc.c | 2 +- drivers/scsi/qla1280.c | 2 +- drivers/usb/gadget/Kconfig | 2 +- fs/debugfs/file.c | 2 +- fs/debugfs/inode.c | 2 +- include/linux/ata.h | 2 +- include/linux/debugfs.h | 2 +- include/linux/libata.h | 2 +- include/sound/pcm.h | 2 +- kernel/irq/chip.c | 2 +- kernel/irq/handle.c | 2 +- kernel/irq/irqdesc.c | 2 +- lib/Kconfig.debug | 2 +- lib/Kconfig.kgdb | 2 +- scripts/Makefile | 9 +- scripts/check-lc_ctype.c | 11 -- 64 files changed, 396 insertions(+), 1032 deletions(-) delete mode 100644 Documentation/DocBook/.gitignore delete mode 100644 Documentation/DocBook/Makefile delete mode 100644 Documentation/DocBook/lsm.tmpl delete mode 100644 Documentation/DocBook/stylesheet.xsl delete mode 100644 Documentation/Makefile.sphinx delete mode 100644 Documentation/doc-guide/docbook.rst create mode 100644 Documentation/lsm.txt delete mode 100755 Documentation/sphinx/tmplcvt delete mode 100644 scripts/check-lc_ctype.c -- 2.9.3

3 8

[alsa-devel] future of sounds/oss
by Christoph Hellwig 16 May '17

16 May '17

Ho Jaroslav, hi Takashi, do you know who still uses the sound/oss drivers and why? I've recently been looking into getting rid of set_fs for using copy_{from,to}_user and friends on kernel pointers, and the sound code is a big abuser, both ALSA and the legacy OSS code. But looking at the OSS code it's pretty grotty, and also appears to be pretty much unmaintained except for global cleanups. Is there any chance we could look into getting rid of it over the next few merge windows or are there people that rely on it?

7 18

[alsa-devel] [PATCH] ALSA: hda - No loopback on ALC299 codec
by Takashi Iwai 16 May '17

16 May '17

ALC299 has no loopback mixer, but the driver still tries to add a beep control over the mixer NID which leads to the error at accessing it. This patch fixes it by properly declaring mixer_nid=0 for this codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195775 Fixes: 28f1f9b26cee ("ALSA: hda/realtek - Add new codec ID ALC299") Cc: stable(a)vger.kernel.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 58df440013c5..9c22ad694534 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6465,8 +6465,11 @@ static int patch_alc269(struct hda_codec *codec) break; case 0x10ec0225: case 0x10ec0295: + spec->codec_variant = ALC269_TYPE_ALC225; + break; case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; + spec->gen.mixer_nid = 0; /* no loopback on ALC299 */ break; case 0x10ec0234: case 0x10ec0274: -- 2.12.2

1 0

Re: [alsa-devel] Distorted audio from microphone (Logitech C310)
by Greg KH 16 May '17

16 May '17

On Mon, May 15, 2017 at 03:29:09PM +0200, Domker_ wrote: > My webcam Logitech C310 have a distored microphone audio on kernel 4.10.15-1 > and 4.11.0-1. (on 4.9.27-1 and the older ones is fine). > The distortion effect is as if I spoke very, very fast (like a damn Mickey > Mouse) > > I checked and the bug occurs on both computers I have. (with the same > camera/different sound cards, hardware and different Linux distributions) > The bug always occurs when running a system with a kernel newer than 4.9.x. > > The camera has two internal microphones. > Secound kernel 4.11.0-1 (x86_64) compiled from kernel.org! > > (tested under Linux Arch and Linux Manjaro) Hm, maybe the Linux sound developers can help out here (added to cc:)

2 1

[alsa-devel] [PATCH 0/2] ASoC: sunxi: Add ADC support for A33
by Mylène Josserand 16 May '17

16 May '17

Hello everyone, This is a first version of a serie to add ADC support for Sun8i-A33. Based on asoc/for-next branch, last commit: 20d5c84bef067b7e804a163e2abca16c47125bad. The first patch adds the support of ADC and microphones in the digital part (driver sun8i-codec). The second one adds the route between analog (sun8i-codec-analog) and digital (sun8i-codec) parts to have a functional ADC route on Sun8i-A33 device tree. Tested on Allwinner R16 Parrot board with microphone 1 (external) and microphone 2 (headset). I am not sure about the bias handling so if you have any comments on it, do not hesitate. Thank you in advance! Best regards, Mylène Josserand (2): ASoC: sun8i-codec: Add ADC support for a33 ARM: dts: sun8i: Add ADC routing arch/arm/boot/dts/sun8i-a33.dtsi | 10 ++++- sound/soc/sunxi/sun8i-codec.c | 80 +++++++++++++++++++++++++++++++++++++++- 2 files changed, 87 insertions(+), 3 deletions(-) -- 2.11.0

3 7

[alsa-devel] [PATCH 1/2] ASoC: add es8316 codec driver
by Daniel Drake 15 May '17

15 May '17

Add a codec driver for the Everest ES8316, based on code provided by David Yang from Everest Semi. I limited the functionality to items where the vendor code was clear, and things that can be tested on the Weibu F3C (Intel Cherry Trail). As a result the initial implementation only supports running in slave mode at single speed (up to 48kHz sample rate) using I2S. HPD is not supported. Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8316.c | 639 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8316.h | 122 +++++++++ 4 files changed, 767 insertions(+) create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..c6286e5ba511 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 @@ -543,6 +544,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..e878306ce46e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 000000000000..aea0b981891b --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,639 @@ +/* + * es8316.c -- es8316 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com>, + * Daniel Drake <drake(a)endlessm.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/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8316.h" + +/* In slave mode at single speed, the codec is documented as accepting 5 + * MCLK/LRCK ratios, but we also add ratio 400, which is commonly used on + * Intel Cherry Trail platforms (19.2MHz MCLK, 48kHz LRCK). + */ +#define NR_SUPPORTED_MCLK_LRCK_RATIOS 6 +static const unsigned int supported_mclk_lrck_ratios[] = { + 256, 384, 400, 512, 768, 1024 +}; + +struct es8316_priv { + unsigned int sysclk; + unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; + struct snd_pcm_hw_constraint_list sysclk_constraints; +}; + +/* + * ES8316 controls + */ +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(mic_bst_tlv, 0, 1200, 0); + +static unsigned int linin_pga_tlv[] = { + TLV_DB_RANGE_HEAD(12), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(600, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(900, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(1200, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1500, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1800, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(2100, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(2400, 0, 0), +}; +static unsigned int hpout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 3, TLV_DB_SCALE_ITEM(-4800, 1200, 0), +}; +static const char * const alc_func_txt[] = { "Off", "On" }; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(ES8316_ADC_ALC1, 6, 2, alc_func_txt); + +static const char * const ng_type_txt[] = + { "Constant PGA Gain", "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(ES8316_ADC_ALC6, 6, 2, ng_type_txt); + +static const char * const adcpol_txt[] = { "Normal", "Invert" }; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(ES8316_ADC_MUTE, 1, 2, adcpol_txt); +static const char *const dacpol_txt[] = + { "Normal", "R Invert", "L Invert", "L + R Invert" }; +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8316_DAC_SET1, 0, 4, dacpol_txt); + +static const struct snd_kcontrol_new es8316_snd_controls[] = { + SOC_DOUBLE_TLV("HP Playback Volume", ES8316_CPHP_ICAL_VOL, + 4, 0, 0, 1, hpout_vol_tlv), + SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL, + 0, 4, 7, 0, hpmixer_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, + ES8316_DAC_VOLR, 0, 0xC0, 1, dac_vol_tlv), + + SOC_SINGLE("Enable DAC Soft Ramp", ES8316_DAC_SET1, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE("DAC Notch Filter", ES8316_DAC_SET2, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Mode", ES8316_DAC_SET2, 7, 1, 0), + SOC_SINGLE("DAC Volume Control-LeR", ES8316_DAC_SET2, 2, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0), + + /* +20dB D2SE PGA Control */ + SOC_SINGLE_TLV("MIC Boost", ES8316_ADC_D2SEPGA, + 0, 1, 0, mic_bst_tlv), + + /* 0-+24dB Lineinput PGA Control */ + SOC_SINGLE_TLV("Input PGA", ES8316_ADC_PGAGAIN, + 4, 8, 0, linin_pga_tlv), + + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME, + 0, 0xc0, 1, adc_vol_tlv), + SOC_SINGLE("ADC Soft Ramp", ES8316_ADC_MUTE, 4, 1, 0), + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("ADC Double FS Mode", ES8316_ADC_DMIC, 4, 1, 0), + + SOC_SINGLE("ALC Capture Target Volume", + ES8316_ADC_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Max PGA", ES8316_ADC_ALC1, 0, 28, 0), + SOC_SINGLE("ALC Capture Min PGA", ES8316_ADC_ALC2, 0, 28, 0), + SOC_ENUM("ALC Capture Function", alc_func), + SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0), + SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4, 0, 10, 0), + SOC_SINGLE("ALC Capture NG Threshold", ES8316_ADC_ALC6, 0, 31, 0), + SOC_ENUM("ALC Capture NG Type", ng_type), + SOC_SINGLE("ALC Capture NG Switch", ES8316_ADC_ALC6, 5, 1, 0), +}; + +/* Analog Input Mux */ +static const char * const es8316_analog_in_txt[] = { + "lin1-rin1", + "lin2-rin2", + "lin1-rin1 with 20db Boost", + "lin2-rin2 with 20db Boost" +}; +static const unsigned int es8316_analog_in_values[] = { 0, 1, 2, 3 }; +static const struct soc_enum es8316_analog_input_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL, 4, 3, + ARRAY_SIZE(es8316_analog_in_txt), + es8316_analog_in_txt, + es8316_analog_in_values); +static const struct snd_kcontrol_new es8316_analog_in_mux_controls = + SOC_DAPM_ENUM("Route", es8316_analog_input_enum); + +static const char * const es8316_dmic_txt[] = { + "dmic disable", + "dmic data at high level", + "dmic data at low level", +}; +static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const struct soc_enum es8316_dmic_src_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, + ARRAY_SIZE(es8316_dmic_txt), + es8316_dmic_txt, + es8316_dmic_values); +static const struct snd_kcontrol_new es8316_dmic_src_controls = + SOC_DAPM_ENUM("Route", es8316_dmic_src_enum); + +/* hp mixer mux */ +static const char * const es8316_hpmux_texts[] = { + "lin1-rin1", + "lin2-rin2", + "lin-rin with Boost", + "lin-rin with Boost and PGA" +}; + +static const unsigned int es8316_hpmux_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_left_hpmux_enum, ES8316_HPMIX_SEL, + 4, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_left_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum); + +static SOC_ENUM_SINGLE_DECL(es8316_right_hpmux_enum, ES8316_HPMIX_SEL, + 0, es8316_hpmux_texts); + +static const struct snd_kcontrol_new es8316_right_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum); + +/* headphone Output Mixer */ +static const struct snd_kcontrol_new es8316_out_left_mix[] = { + SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH, 6, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", ES8316_HPMIX_SWITCH, 7, 1, 0), +}; +static const struct snd_kcontrol_new es8316_out_right_mix[] = { + SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH, 2, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", ES8316_HPMIX_SWITCH, 3, 1, 0), +}; + +/* DAC data source mux */ +static const char * const es8316_dacsrc_texts[] = { + "LDATA TO LDAC, RDATA TO RDAC", + "LDATA TO LDAC, LDATA TO RDAC", + "RDATA TO LDAC, RDATA TO RDAC", + "RDATA TO LDAC, LDATA TO RDAC", +}; + +static const unsigned int es8316_dacsrc_values[] = { 0, 1, 2, 3 }; + +static SOC_ENUM_SINGLE_DECL(es8316_dacsrc_mux_enum, ES8316_DAC_SET1, + 6, es8316_dacsrc_texts); + +static const struct snd_kcontrol_new es8316_dacsrc_mux_controls = + SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum); + +static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_MICBIAS("micbias", SND_SOC_NOPM, 0, 0), + + /* Input Mux */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + + SND_SOC_DAPM_PGA("Line input PGA", ES8316_ADC_PDN_LINSEL, + 7, 1, NULL, 0), + SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL, 6, 1), + SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0, + &es8316_dmic_src_controls), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 1, + ES8316_SERDATA_ADC, 6, 1), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC SRC Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + + SND_SOC_DAPM_DAC("Right DAC", NULL, SND_SOC_NOPM, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, SND_SOC_NOPM, 4, 1), + + /* Headphone Output Side */ + SND_SOC_DAPM_MUX("Left Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + SND_SOC_DAPM_MIXER("Left Hp mixer", SND_SOC_NOPM, + 4, 1, &es8316_out_left_mix[0], + ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", SND_SOC_NOPM, + 0, 1, &es8316_out_right_mix[0], + ARRAY_SIZE(es8316_out_right_mix)), + + /* Output charge pump */ + SND_SOC_DAPM_PGA("HPCP L", SND_SOC_NOPM, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPCP R", SND_SOC_NOPM, 2, 0, NULL, 0), + + /* Output Driver */ + SND_SOC_DAPM_PGA("HPVOL L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPVOL R", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8316_dapm_routes[] = { + /* Recording */ + {"MIC1", NULL, "micbias"}, + {"MIC2", NULL, "micbias"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "Line input PGA"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + + /* Playback */ + {"DAC SRC Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC SRC Mux"}, + {"Right DAC", NULL, "DAC SRC Mux"}, + + {"Left Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + {"Right Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Hp mixer", "LLIN Switch", "Left Hp mux"}, + {"Left Hp mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Hp mixer", "RLIN Switch", "Right Hp mux"}, + {"Right Hp mixer", "Right DAC Switch", "Right DAC"}, + + {"HPCP L", NULL, "Left Hp mixer"}, + {"HPCP R", NULL, "Right Hp mixer"}, + + {"HPVOL L", NULL, "HPCP L"}, + {"HPVOL R", NULL, "HPCP R"}, + + {"HPOL", NULL, "HPVOL L"}, + {"HPOR", NULL, "HPVOL R"}, +}; + +static int es8316_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 es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + int i; + + if (freq == 0) + return -EINVAL; + + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + es8316->allowed_rates[i] = freq / ratio; + } + + es8316->sysclk_constraints.list = es8316->allowed_rates; + es8316->sysclk_constraints.count = NR_SUPPORTED_MCLK_LRCK_RATIOS; + es8316->sysclk = freq; + + return 0; +} + +static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 serdata1 = 0; + u8 serdata2 = 0; + u8 mask; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Codec driver only supports slave mode\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Codec driver only supports I2S format\n"); + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + case SND_SOC_DAIFMT_IB_NF: + serdata1 |= ES8316_SERDATA1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + serdata2 |= ES8316_SERDATA2_ADCLRP; + break; + default: + return -EINVAL; + } + + mask = ES8316_SERDATA1_MASTER | ES8316_SERDATA1_BCLK_INV; + snd_soc_update_bits(codec, ES8316_SERDATA1, mask, serdata1); + + mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP; + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, mask, serdata2); + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, mask, serdata2); + + return 0; +} + +static int es8316_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC. + */ + if (es8316->sysclk) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &es8316->sysclk_constraints); + + return 0; +} + +static int es8316_pcm_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 snd_soc_codec *codec = rtd->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + u8 wordlen = 0; + + if (!es8316->sysclk) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wordlen = ES8316_SERDATA2_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wordlen = ES8316_SERDATA2_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wordlen = ES8316_SERDATA2_LEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wordlen = ES8316_SERDATA2_LEN_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ES8316_SERDATA_DAC, + ES8316_SERDATA2_LEN_MASK, wordlen); + snd_soc_update_bits(codec, ES8316_SERDATA_ADC, + ES8316_SERDATA2_LEN_MASK, wordlen); + return 0; +} + +static int es8316_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ES8316_DAC_SET1, 0x20, + mute ? 0x20 : 0); + return 0; +} + +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_write(codec, ES8316_RESET, 0xC0); + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x66); + snd_soc_write(codec, ES8316_DAC_PDN, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x10); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x03); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x00); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0x00); + snd_soc_write(codec, ES8316_SYS_PDN, 0x00); + snd_soc_write(codec, ES8316_SYS_LP1, 0x04); + snd_soc_write(codec, ES8316_SYS_LP2, 0x0c); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN, 0x11); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x06); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x33); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0xc0); + snd_soc_write(codec, ES8316_SYS_PDN, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP1, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP2, 0x1f); + snd_soc_write(codec, ES8316_RESET, 0x00); + break; + } + + return 0; +} + +#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops es8316_ops = { + .startup = es8316_pcm_startup, + .hw_params = es8316_pcm_hw_params, + .set_fmt = es8316_set_dai_fmt, + .set_sysclk = es8316_set_dai_sysclk, + .digital_mute = es8316_mute, +}; + +static struct snd_soc_dai_driver es8316_dai = { + .name = "ES8316 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ES8316_FORMATS, + }, + .ops = &es8316_ops, + .symmetric_rates = 1, +}; + +static int es8316_probe(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ES8316_RESET, 0x3f); + usleep_range(5000, 5500); + snd_soc_write(codec, ES8316_RESET, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xFF); + msleep(30); + + snd_soc_write(codec, ES8316_CLKMGR_CLKSEL, 0x08); + snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x20); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_CPDIV, 0x00); + snd_soc_write(codec, ES8316_SERDATA1, 0x04); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW, 0x7f); + snd_soc_write(codec, ES8316_CAL_TYPE, 0x0F); + snd_soc_write(codec, ES8316_CAL_HPLIV, 0x90); + snd_soc_write(codec, ES8316_CAL_HPRIV, 0x90); + snd_soc_write(codec, ES8316_ADC_VOLUME, 0x00); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x00); + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_DAC_SET2, 0x00); + snd_soc_write(codec, ES8316_DAC_SET3, 0x00); + snd_soc_write(codec, ES8316_DAC_VOLL, 0x02); + snd_soc_write(codec, ES8316_DAC_VOLR, 0x02); + snd_soc_write(codec, ES8316_SERDATA_ADC, 0x00); + snd_soc_write(codec, ES8316_SERDATA_DAC, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDLOW, 0x11); + snd_soc_write(codec, ES8316_SYS_VSEL, 0xfc); + snd_soc_write(codec, ES8316_SYS_REF, 0x28); + snd_soc_write(codec, ES8316_HPMIX_SEL, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH, 0x88); + snd_soc_write(codec, ES8316_HPMIX_VOL, 0x88); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL, 0x00); + snd_soc_write(codec, ES8316_GPIO_SEL, 0x00); + snd_soc_write(codec, ES8316_GPIO_DEBUNCE_INT, 0x02); + snd_soc_write(codec, ES8316_TESTMODE, 0xa0); + snd_soc_write(codec, ES8316_TEST1, 0x00); + snd_soc_write(codec, ES8316_TEST2, 0x04); + snd_soc_write(codec, ES8316_RESET, 0xc0); + msleep(50); + snd_soc_write(codec, ES8316_ADC_PGAGAIN, 0x60); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x01); + + /* ADC DS mode, HPF enable */ + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC1, 0xcd); + snd_soc_write(codec, ES8316_ADC_ALC2, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC3, 0xa0); + snd_soc_write(codec, ES8316_ADC_ALC4, 0x05); + snd_soc_write(codec, ES8316_ADC_ALC5, 0x06); + snd_soc_write(codec, ES8316_ADC_ALC6, 0x61); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + .set_bias_level = es8316_set_bias_level, + .idle_bias_off = true, + + .component_driver = { + .controls = es8316_snd_controls, + .num_controls = ARRAY_SIZE(es8316_snd_controls), + .dapm_widgets = es8316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets), + .dapm_routes = es8316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), + }, +}; + +static const struct regmap_config es8316_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x53, + .cache_type = REGCACHE_RBTREE, +}; + +static int es8316_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct es8316_priv *es8316; + struct regmap *regmap; + + es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv), + GFP_KERNEL); + if (es8316 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, es8316); + + regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316, + &es8316_dai, 1); +} + +static int es8316_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id es8316_i2c_id[] = { + {"ESSX8316", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); + +static const struct of_device_id es8316_of_match[] = { + { .compatible = "everest,es8316", }, + {}, +}; +MODULE_DEVICE_TABLE(of, es8316_of_match); + +static const struct acpi_device_id es8316_acpi_match[] = { + {"ESSX8316", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); + +static struct i2c_driver es8316_i2c_driver = { + .driver = { + .name = "es8316", + .acpi_match_table = ACPI_PTR(es8316_acpi_match), + .of_match_table = of_match_ptr(es8316_of_match), + }, + .probe = es8316_i2c_probe, + .remove = es8316_i2c_remove, + .id_table = es8316_i2c_id, +}; +module_i2c_driver(es8316_i2c_driver); + +MODULE_DESCRIPTION("Everest Semi ES8316 ALSA SoC Codec Driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 000000000000..702bebc69ca8 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,122 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.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 _ES8316_H +#define _ES8316_H + +/* + * ES8316 register space + */ + +/* Reset Control */ +#define ES8316_RESET 0x00 + +/* Clock Management */ +#define ES8316_CLKMGR_CLKSW 0x01 +#define ES8316_CLKMGR_CLKSEL 0x02 +#define ES8316_CLKMGR_ADCOSR 0x03 +#define ES8316_CLKMGR_ADCDIV1 0x04 +#define ES8316_CLKMGR_ADCDIV2 0x05 +#define ES8316_CLKMGR_DACDIV1 0x06 +#define ES8316_CLKMGR_DACDIV2 0x07 +#define ES8316_CLKMGR_CPDIV 0x08 + +/* Serial Data Port Control */ +#define ES8316_SERDATA1 0x09 +#define ES8316_SERDATA_ADC 0x0a +#define ES8316_SERDATA_DAC 0x0b + +/* System Control */ +#define ES8316_SYS_VMIDSEL 0x0c +#define ES8316_SYS_PDN 0x0d +#define ES8316_SYS_LP1 0x0e +#define ES8316_SYS_LP2 0x0f +#define ES8316_SYS_VMIDLOW 0x10 +#define ES8316_SYS_VSEL 0x11 +#define ES8316_SYS_REF 0x12 + +/* Headphone Mixer */ +#define ES8316_HPMIX_SEL 0x13 +#define ES8316_HPMIX_SWITCH 0x14 +#define ES8316_HPMIX_PDN 0x15 +#define ES8316_HPMIX_VOL 0x16 + +/* Charge Pump Headphone driver */ +#define ES8316_CPHP_OUTEN 0x17 +#define ES8316_CPHP_ICAL_VOL 0x18 +#define ES8316_CPHP_PDN1 0x19 +#define ES8316_CPHP_PDN2 0x1a +#define ES8316_CPHP_LDOCTL 0x1b + +/* Calibration */ +#define ES8316_CAL_TYPE 0x1c +#define ES8316_CAL_SET 0x1d +#define ES8316_CAL_HPLIV 0x1e +#define ES8316_CAL_HPRIV 0x1f +#define ES8316_CAL_HPLMV 0x20 +#define ES8316_CAL_HPRMV 0x21 + +/* ADC Control */ +#define ES8316_ADC_PDN_LINSEL 0x22 +#define ES8316_ADC_PGAGAIN 0x23 +#define ES8316_ADC_D2SEPGA 0x24 +#define ES8316_ADC_DMIC 0x25 +#define ES8316_ADC_MUTE 0x26 +#define ES8316_ADC_VOLUME 0x27 +#define ES8316_ADC_ALC1 0x29 +#define ES8316_ADC_ALC2 0x2a +#define ES8316_ADC_ALC3 0x2b +#define ES8316_ADC_ALC4 0x2c +#define ES8316_ADC_ALC5 0x2d +#define ES8316_ADC_ALC6 0x2e + +/* DAC Control */ +#define ES8316_DAC_PDN 0x2f +#define ES8316_DAC_SET1 0x30 +#define ES8316_DAC_SET2 0x31 +#define ES8316_DAC_SET3 0x32 +#define ES8316_DAC_VOLL 0x33 +#define ES8316_DAC_VOLR 0x34 + +/* GPIO */ +#define ES8316_GPIO_SEL 0x4d +#define ES8316_GPIO_DEBUNCE_INT 0x4e +#define ES8316_GPIO_FLAG 0x4f + +/* Test mode */ +#define ES8316_TESTMODE 0x50 +#define ES8316_TEST1 0x51 +#define ES8316_TEST2 0x52 +#define ES8316_TEST3 0x53 + +/* + * Field definitions + */ + +/* ES8316_SERDATA1 */ +#define ES8316_SERDATA1_MASTER 0x80 +#define ES8316_SERDATA1_BCLK_INV 0x20 + +/* ES8316_SERDATA_ADC and _DAC */ +#define ES8316_SERDATA2_FMT_MASK 0x3 +#define ES8316_SERDATA2_FMT_I2S 0x00 +#define ES8316_SERDATA2_FMT_LEFTJ 0x01 +#define ES8316_SERDATA2_FMT_RIGHTJ 0x02 +#define ES8316_SERDATA2_FMT_PCM 0x03 +#define ES8316_SERDATA2_ADCLRP 0x20 +#define ES8316_SERDATA2_LEN_MASK 0x1c +#define ES8316_SERDATA2_LEN_24 0x00 +#define ES8316_SERDATA2_LEN_20 0x04 +#define ES8316_SERDATA2_LEN_18 0x08 +#define ES8316_SERDATA2_LEN_16 0x0c +#define ES8316_SERDATA2_LEN_32 0x10 + +#endif -- 2.11.0

3 7

[alsa-devel] [PATCH 00/36] Convert DocBook documents to ReST
by Mauro Carvalho Chehab 15 May '17

15 May '17

This patch series convert the following books from DocBook to ReST: - filesystems - kernel-hacking - kernel-locking - kgdb - libata - networking - rapidio - s390-drivers - scsi - w1 - z8530book It also adjusts some Sphinx-pedantic errors/warnings on some kernel-doc markups. I also added some patches here to add PDF output for all existing ReST books. I did my best to check if what's there is not too outdated, but the best is if the subsystem maintainers could check it. After this series, there are only 4 DocBook remaining conversion: - librs - lsm - mtdnand - sh I'll likely convert those remaining ones during this weekend. - This patch series is based on docs tree (next branch). The full patch series is on this tree: https://git.linuxtv.org//mchehab/experimental.git/log/?h=docbook And the HTML output at: http://www.infradead.org/~mchehab/kernel_docs/ https://mchehab.fedorapeople.org/kernel_docs/ Mauro Carvalho Chehab (36): docs-rst: convert kernel-hacking to ReST kernel-hacking: update document docs-rst: convert kernel-locking to ReST mutex, futex: adjust kernel-doc markups to generate ReST locking.rst: reformat locking table locking.rst: add captions to two tables locking.rst: Update some ReST markups docs-rst: convert kgdb DocBook to ReST kgdb.rst: Adjust ReST markups conf.py: define a color for important markup on PDF output docs-rst: conf.py: sort LaTeX documents in alphabetical order docs-rst: conf.py: remove kernel-documentation from LaTeX docs-rst: add crypto API book to pdf output docs-rst: add dev-tools book to pdf output docs-rst: add sound book to pdf output docs-rst: add userspace API book to pdf output docs-rst: convert filesystems book to ReST docs-rst: filesystems: use c domain references where needed fs: jbd2: make jbd2_journal_start() kernel-doc parseable docs-rst: don't ignore internal functions for jbd2 docs fs: locks: Fix some troubles at kernel-doc comments fs: add a blank lines on some kernel-doc comments fs: eventfd: fix identation on kernel-doc fs: jbd2: escape a string with special chars on a kernel-doc docs-rst: convert libata book to ReST libata.rst: add c function and struct cross-references libata: fix identation on a kernel-doc markup docs-rst: convert s390-drivers DocBook to ReST docs-rst: convert networking book to ReST net: skbuff.h: properly escape a macro name on kernel-doc net: fix some identation issues at kernel-doc markups docs-rst: convert z8530book DocBook to ReST docs-rst: convert scsi DocBook to ReST scsi: fix some kernel-doc markups docs-rst: convert w1 book to ReST docs-rst: convert rapidio book to ReST Documentation/DocBook/Makefile | 11 +- Documentation/DocBook/filesystems.tmpl | 381 ----- Documentation/DocBook/kernel-hacking.tmpl | 1312 ------------------ Documentation/DocBook/kernel-locking.tmpl | 2151 ----------------------------- Documentation/DocBook/kgdb.tmpl | 918 ------------ Documentation/DocBook/libata.tmpl | 1625 ---------------------- Documentation/DocBook/networking.tmpl | 111 -- Documentation/DocBook/rapidio.tmpl | 155 --- Documentation/DocBook/s390-drivers.tmpl | 161 --- Documentation/DocBook/scsi.tmpl | 409 ------ Documentation/DocBook/w1.tmpl | 101 -- Documentation/DocBook/z8530book.tmpl | 371 ----- Documentation/conf.py | 35 +- Documentation/crypto/conf.py | 10 + Documentation/dev-tools/index.rst | 1 + Documentation/dev-tools/kgdb.rst | 907 ++++++++++++ Documentation/driver-api/index.rst | 5 + Documentation/driver-api/libata.rst | 1031 ++++++++++++++ Documentation/driver-api/rapidio.rst | 107 ++ Documentation/driver-api/s390-drivers.rst | 111 ++ Documentation/driver-api/scsi.rst | 344 +++++ Documentation/driver-api/w1.rst | 70 + Documentation/filesystems/conf.py | 10 + Documentation/filesystems/index.rst | 317 +++++ Documentation/index.rst | 3 + Documentation/kernel-hacking/conf.py | 10 + Documentation/kernel-hacking/hacking.rst | 811 +++++++++++ Documentation/kernel-hacking/index.rst | 5 + Documentation/kernel-hacking/locking.rst | 1446 +++++++++++++++++++ Documentation/networking/conf.py | 10 + Documentation/networking/index.rst | 18 + Documentation/networking/kapi.rst | 147 ++ Documentation/networking/z8530book.rst | 256 ++++ Documentation/sound/conf.py | 10 + drivers/ata/libata-scsi.c | 7 +- drivers/net/phy/phy.c | 1 + drivers/scsi/scsi_scan.c | 7 +- drivers/scsi/scsi_transport_fc.c | 18 +- drivers/scsi/scsicam.c | 4 +- fs/eventfd.c | 4 +- fs/fs-writeback.c | 12 +- fs/jbd2/transaction.c | 42 +- fs/locks.c | 18 +- fs/mpage.c | 1 + fs/namei.c | 1 + include/linux/mutex.h | 6 +- include/linux/netdevice.h | 9 +- include/linux/skbuff.h | 2 +- include/net/sock.h | 9 +- kernel/futex.c | 40 +- kernel/locking/mutex.c | 6 +- net/core/datagram.c | 2 +- net/core/sock.c | 7 +- 53 files changed, 5764 insertions(+), 7802 deletions(-) delete mode 100644 Documentation/DocBook/filesystems.tmpl delete mode 100644 Documentation/DocBook/kernel-hacking.tmpl delete mode 100644 Documentation/DocBook/kernel-locking.tmpl delete mode 100644 Documentation/DocBook/kgdb.tmpl delete mode 100644 Documentation/DocBook/libata.tmpl delete mode 100644 Documentation/DocBook/networking.tmpl delete mode 100644 Documentation/DocBook/rapidio.tmpl delete mode 100644 Documentation/DocBook/s390-drivers.tmpl delete mode 100644 Documentation/DocBook/scsi.tmpl delete mode 100644 Documentation/DocBook/w1.tmpl delete mode 100644 Documentation/DocBook/z8530book.tmpl create mode 100644 Documentation/crypto/conf.py create mode 100644 Documentation/dev-tools/kgdb.rst create mode 100644 Documentation/driver-api/libata.rst create mode 100644 Documentation/driver-api/rapidio.rst create mode 100644 Documentation/driver-api/s390-drivers.rst create mode 100644 Documentation/driver-api/scsi.rst create mode 100644 Documentation/driver-api/w1.rst create mode 100644 Documentation/filesystems/conf.py create mode 100644 Documentation/filesystems/index.rst create mode 100644 Documentation/kernel-hacking/conf.py create mode 100644 Documentation/kernel-hacking/hacking.rst create mode 100644 Documentation/kernel-hacking/index.rst create mode 100644 Documentation/kernel-hacking/locking.rst create mode 100644 Documentation/networking/conf.py create mode 100644 Documentation/networking/index.rst create mode 100644 Documentation/networking/kapi.rst create mode 100644 Documentation/networking/z8530book.rst create mode 100644 Documentation/sound/conf.py -- 2.9.3

2 3

[alsa-devel] [PATCH 1/2] ALSA: firewire-tascam: Fix infinite loop in snd_tscm_stream_get_rate()
by Geert Uytterhoeven 15 May '17

15 May '17

Obviously the intention was to put a limit on the maximum number of operations. However, for this to work, the check should be "&& trials++ < 5", not "|| trials++ < 5". Fixes: 35efa5c489de63a9 ("ALSA: firewire-tascam: add streaming functionality") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- Compile-tested only. Triggered by a false-positive warning from gcc-4.1.2: warning: ‘err’ may be used uninitialized in this function --- sound/firewire/tascam/tascam-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index f1657a4e0621ef49..e433b92ac6904db5 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -84,7 +84,7 @@ int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate) unsigned int trials = 0; int err; - while (data == 0x0 || trials++ < 5) { + while (data == 0x0 && trials++ < 5) { err = get_clock(tscm, &data); if (err < 0) return err; -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: codec: wm9860: avoid maybe-uninitialized warning
by Arnd Bergmann 15 May '17

15 May '17

The new PLL configuration code triggers a harmless warning: sound/soc/codecs/wm8960.c: In function 'wm8960_configure_clocking': sound/soc/codecs/wm8960.c:735:3: error: 'best_freq_out' may be used uninitialized in this function [-Werror=maybe-uninitialized] wm8960_set_pll(codec, freq_in, best_freq_out); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/wm8960.c:699:12: note: 'best_freq_out' was declared here I think the warning was introduced by Daniel's bugfix. I've come up with a way to simplify the code in a way that is more readable to both humans and to gcc, which gets us rid of the warning. Fixes: 84fdc00d519f ("ASoC: codec: wm9860: Refactor PLL out freq search") Fixes: 303e8954af8d ("ASoC: codec: wm8960: Stop when a matching PLL freq is found") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm8960.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ace69da97cb8..ba0f80504c6b 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -696,14 +696,12 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); int sysclk, bclk, lrclk, freq_out; - int diff, best_freq_out; + int diff; int i, j, k; bclk = wm8960->bclk; lrclk = wm8960->lrclk; - *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { if (sysclk_divs[i] == -1) continue; @@ -720,22 +718,14 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, *sysclk_idx = i; *dac_idx = j; *bclk_idx = k; - best_freq_out = freq_out; - break; + return freq_out; } } - if (k != ARRAY_SIZE(bclk_divs)) - break; } - if (j != ARRAY_SIZE(dac_divs)) - break; } - - if (*bclk_idx != -1) - wm8960_set_pll(codec, freq_in, best_freq_out); - - return *bclk_idx; + return -EINVAL; } + static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); @@ -783,11 +773,12 @@ static int wm8960_configure_clocking(struct snd_soc_codec *codec) } } - ret = wm8960_configure_pll(codec, freq_in, &i, &j, &k); - if (ret < 0) { + freq_out = wm8960_configure_pll(codec, freq_in, &i, &j, &k); + if (freq_out < 0) { dev_err(codec->dev, "failed to configure clock via PLL\n"); - return -EINVAL; + return freq_out; } + wm8960_set_pll(codec, freq_in, freq_out); configure_clock: /* configure sysclk clock */ -- 2.9.0

4 7

[alsa-devel] HDMI+SND_JACK_AVOUT
by Pierre-Louis Bossart 15 May '17

15 May '17

Hi, Does anyone know the reason why the Intel/HDaudio HDMI layers use SND_JACK_AVOUT as a jack type? This value is composite and ends up having the side effect of sending 2 independent input events to userspace (SW_LINEOUT_INSERT and SW_VIDEOOUT_INSERT) and confusing the Android logic which deals with wired accessories. Shouldn't we use just SND_JACK_VIDEOOUT instead? What is the reason or rationale behind the use of LINEOUT for HDMI/DP jack events? Thanks -Pierre http://elixir.free-electrons.com/linux/latest/ident/SND_JACK_AVOUT

2 1

[alsa-devel] [PATCH] ASoC: utils: Add DT support
by Vishal Thanki 15 May '17

15 May '17

Add device tree support for the dummy codec driver. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> --- sound/soc/soc-utils.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 644d9a9..c76b938 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -19,6 +19,8 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <linux/module.h> +#include <linux/of.h> int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) { @@ -359,9 +361,19 @@ static int snd_soc_dummy_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dummy_dt_ids[] = { + { .compatible = "linux,snd-soc-dummy", }, + { } +}; +MODULE_DEVICE_TABLE(of, dummy_dt_ids); +#endif + + static struct platform_driver soc_dummy_driver = { .driver = { .name = "snd-soc-dummy", + .of_match_table = of_match_ptr(dummy_dt_ids), }, .probe = snd_soc_dummy_probe, .remove = snd_soc_dummy_remove, -- 2.4.11

2 3

[alsa-devel] [PATCH 1/3] ALSA: opl4: Use IS_REACHABLE()
by Takashi Iwai 15 May '17

15 May '17

Rewrite the complex ifdef condition with IS_REACHABLE(). The ifdef in opl4_local.h was without defined(MODLE) check, but this is likely the oversight. Use IS_REACHABLE() here as well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/opl4/opl4_lib.c | 4 ++-- sound/drivers/opl4/opl4_local.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 89c7aa04b3bc..240656e54400 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -153,7 +153,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) return 0; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) static void snd_opl4_seq_dev_free(struct snd_seq_device *seq_dev) { struct snd_opl4 *opl4 = seq_dev->private_data; @@ -249,7 +249,7 @@ int snd_opl4_create(struct snd_card *card, snd_opl4_create_mixer(opl4); snd_opl4_create_proc(opl4); -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) opl4->seq_client = -1; if (opl4->hardware < OPL3_HW_OPL4_ML) snd_opl4_create_seq_dev(opl4, seq_device); diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h index 9a41bdebce6b..d5bac93f8245 100644 --- a/sound/drivers/opl4/opl4_local.h +++ b/sound/drivers/opl4/opl4_local.h @@ -184,7 +184,7 @@ struct snd_opl4 { #endif struct mutex access_mutex; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) int used; int seq_dev_num; -- 2.12.2

2 3

[alsa-devel] [PATCH v2] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-audio-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - simple-scu-card -> simple-scu-audio-card on git log .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: Use IS_ENABLED() in common headers
by Takashi Iwai 15 May '17

15 May '17

Simplify the ifdef conditions with IS_ENABLED() macro in the common sound headers. No functional changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 6 +++--- include/sound/mixer_oss.h | 2 +- include/sound/opl3.h | 4 ++-- include/sound/pcm.h | 10 +++++----- include/sound/rawmidi.h | 4 ++-- 5 files changed, 13 insertions(+), 13 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f7d8c10c4c45..55385588eefa 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -142,7 +142,7 @@ struct snd_card { wait_queue_head_t power_sleep; #endif -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) struct snd_mixer_oss *mixer_oss; int mixer_oss_change_count; #endif @@ -243,7 +243,7 @@ int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size extern struct snd_card *snd_cards[SNDRV_CARDS]; int snd_card_locked(int card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SND_MIXER_OSS_NOTIFY_REGISTER 0 #define SND_MIXER_OSS_NOTIFY_DISCONNECT 1 #define SND_MIXER_OSS_NOTIFY_FREE 2 @@ -394,7 +394,7 @@ static inline void snd_printdd(const char *format, ...) {} #define SNDRV_OSS_VERSION ((3<<16)|(8<<8)|(1<<4)|(0)) /* 3.8.1a */ /* for easier backward-porting */ -#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#if IS_ENABLED(CONFIG_GAMEPORT) #define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev)) #define gameport_set_port_data(gp,r) ((gp)->port_data = (r)) #define gameport_get_port_data(gp) (gp)->port_data diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h index 13cb0b430a1b..930da10fb65b 100644 --- a/include/sound/mixer_oss.h +++ b/include/sound/mixer_oss.h @@ -22,7 +22,7 @@ * */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SNDRV_OSS_MAX_MIXERS 32 diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 6ba670707831..5a87641c6ae4 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -321,7 +321,7 @@ struct snd_opl3 { unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ unsigned char max_voices; /* max number of voices */ -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define SNDRV_OPL3_MODE_SYNTH 0 /* OSS - voices allocated by application */ #define SNDRV_OPL3_MODE_SEQ 1 /* ALSA - driver handles voice allocation */ int synth_mode; /* synth mode */ @@ -374,7 +374,7 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file); void snd_opl3_reset(struct snd_opl3 * opl3); -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, loff_t *offset); int snd_opl3_load_patch(struct snd_opl3 *opl3, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index eb16912d6046..3f084486868f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -34,7 +34,7 @@ #define snd_pcm_substream_chip(substream) ((substream)->private_data) #define snd_pcm_chip(pcm) ((pcm)->private_data) -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <sound/pcm_oss.h> #endif @@ -418,7 +418,7 @@ struct snd_pcm_runtime { struct snd_pcm_audio_tstamp_report audio_tstamp_report; struct timespec driver_tstamp; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif @@ -464,7 +464,7 @@ struct snd_pcm_substream { unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_substream oss; #endif @@ -494,7 +494,7 @@ struct snd_pcm_str { unsigned int substream_count; unsigned int substream_opened; struct snd_pcm_substream *substream; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_stream oss; #endif @@ -526,7 +526,7 @@ struct snd_pcm { void (*private_free) (struct snd_pcm *pcm); bool internal; /* pcm is for internal use only */ bool nonatomic; /* whole PCM operations are in non-atomic context */ -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss oss; #endif }; diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 492a3ca7f17b..6665cb29e1a2 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -30,7 +30,7 @@ #include <linux/workqueue.h> #include <linux/device.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #include <sound/seq_device.h> #endif @@ -144,7 +144,7 @@ struct snd_rawmidi { struct snd_info_entry *proc_entry; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) struct snd_seq_device *seq_dev; #endif }; -- 2.12.2

1 1

[alsa-devel] [PATCH] ALSA: pcm: Build pcm notifier code conditionally
by Takashi Iwai 15 May '17

15 May '17

The PCM notifier code is used only by OSS emulation layer, so we can build it conditionally for reducing the size. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 15 ++++++++------- sound/core/pcm.c | 33 ++++++++++++++++++++------------- 2 files changed, 28 insertions(+), 20 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 361749e60799..eb16912d6046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -531,13 +531,6 @@ struct snd_pcm { #endif }; -struct snd_pcm_notify { - int (*n_register) (struct snd_pcm * pcm); - int (*n_disconnect) (struct snd_pcm * pcm); - int (*n_unregister) (struct snd_pcm * pcm); - struct list_head list; -}; - /* * Registering */ @@ -552,7 +545,15 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, struct snd_pcm **rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +struct snd_pcm_notify { + int (*n_register) (struct snd_pcm * pcm); + int (*n_disconnect) (struct snd_pcm * pcm); + int (*n_unregister) (struct snd_pcm * pcm); + struct list_head list; +}; int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); +#endif /* * Native I/O diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8e980aa678d0..24acbfb5c531 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -36,8 +36,10 @@ MODULE_DESCRIPTION("Midlevel PCM code for ALSA."); MODULE_LICENSE("GPL"); static LIST_HEAD(snd_pcm_devices); -static LIST_HEAD(snd_pcm_notify_list); static DEFINE_MUTEX(register_mutex); +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +static LIST_HEAD(snd_pcm_notify_list); +#endif static int snd_pcm_free(struct snd_pcm *pcm); static int snd_pcm_dev_free(struct snd_device *device); @@ -884,16 +886,23 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) put_device(&pstr->dev); } +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +#define pcm_call_notify(pcm, call) \ + do { \ + struct snd_pcm_notify *_notify; \ + list_for_each_entry(_notify, &snd_pcm_notify_list, list) \ + _notify->call(pcm); \ + } while (0) +#else +#define pcm_call_notify(pcm, call) /* NOP */ +#endif + static int snd_pcm_free(struct snd_pcm *pcm) { - struct snd_pcm_notify *notify; - if (!pcm) return 0; - if (!pcm->internal) { - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_unregister(pcm); - } + if (!pcm->internal) + pcm_call_notify(pcm, n_unregister); if (pcm->private_free) pcm->private_free(pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -1069,7 +1078,6 @@ static int snd_pcm_dev_register(struct snd_device *device) { int cidx, err; struct snd_pcm_substream *substream; - struct snd_pcm_notify *notify; struct snd_pcm *pcm; if (snd_BUG_ON(!device || !device->device_data)) @@ -1107,8 +1115,7 @@ static int snd_pcm_dev_register(struct snd_device *device) snd_pcm_timer_init(substream); } - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_register(pcm); + pcm_call_notify(pcm, n_register); unlock: mutex_unlock(&register_mutex); @@ -1118,7 +1125,6 @@ static int snd_pcm_dev_register(struct snd_device *device) static int snd_pcm_dev_disconnect(struct snd_device *device) { struct snd_pcm *pcm = device->device_data; - struct snd_pcm_notify *notify; struct snd_pcm_substream *substream; int cidx; @@ -1138,8 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) } } if (!pcm->internal) { - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_disconnect(pcm); + pcm_call_notify(pcm, n_disconnect); } for (cidx = 0; cidx < 2; cidx++) { if (!pcm->internal) @@ -1151,6 +1156,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) return 0; } +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /** * snd_pcm_notify - Add/remove the notify list * @notify: PCM notify list @@ -1183,6 +1189,7 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) return 0; } EXPORT_SYMBOL(snd_pcm_notify); +#endif /* CONFIG_SND_PCM_OSS */ #ifdef CONFIG_SND_PROC_FS /* -- 2.12.2

1 0

[alsa-devel] [PATCH 1/3] ALSA: vx222: Use container_of()
by Takashi Iwai 15 May '17

15 May '17

The vx222 driver is using the explicit cast from the parent class pointer, but it'll be broken when the structure field randomization is applied. Use container_of() in a modern manner, instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/vx222/vx222.c | 4 ++-- sound/pci/vx222/vx222.h | 2 ++ sound/pci/vx222/vx222_ops.c | 26 +++++++++++++------------- 3 files changed, 17 insertions(+), 15 deletions(-) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index ecbaf473fc1e..55861849d7df 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -116,7 +116,7 @@ static struct snd_vx_hardware vx222_mic_hw = { */ static int snd_vx222_free(struct vx_core *chip) { - struct snd_vx222 *vx = (struct snd_vx222 *)chip; + struct snd_vx222 *vx = to_vx222(chip); if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); @@ -158,7 +158,7 @@ static int snd_vx222_create(struct snd_card *card, struct pci_dev *pci, pci_disable_device(pci); return -ENOMEM; } - vx = (struct snd_vx222 *)chip; + vx = to_vx222(chip); vx->pci = pci; if ((err = pci_request_regions(pci, CARD_NAME)) < 0) { diff --git a/sound/pci/vx222/vx222.h b/sound/pci/vx222/vx222.h index 2f0d78f609a6..cae355c8ed28 100644 --- a/sound/pci/vx222/vx222.h +++ b/sound/pci/vx222/vx222.h @@ -39,6 +39,8 @@ struct snd_vx222 { int mic_level; /* mic level for vx222 mic */ }; +#define to_vx222(x) container_of(x, struct snd_vx222, core) + /* we use a lookup table with 148 values, see vx_mixer.c */ #define VX2_AKM_LEVEL_MAX 0x93 diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7df1663ea510..d4298af6d3ee 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -86,7 +86,7 @@ static int vx2_reg_index[VX_REG_MAX] = { static inline unsigned long vx2_reg_addr(struct vx_core *_chip, int reg) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); return chip->port[vx2_reg_index[reg]] + vx2_reg_offset[reg]; } @@ -159,7 +159,7 @@ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) static void vx2_reset_dsp(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* set the reset dsp bit to 0 */ vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_DSP_RESET_MASK); @@ -174,7 +174,7 @@ static void vx2_reset_dsp(struct vx_core *_chip) static int vx2_test_xilinx(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); unsigned int data; dev_dbg(_chip->card->dev, "testing xilinx...\n"); @@ -479,7 +479,7 @@ static int vx2_test_and_ack(struct vx_core *chip) */ static void vx2_validate_irq(struct vx_core *_chip, int enable) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* Set the interrupt enable bit to 1 in CDSP register */ if (enable) { @@ -730,7 +730,7 @@ static void vx2_old_write_codec_bit(struct vx_core *chip, int codec, unsigned in */ static void vx2_reset_codec(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* Set the reset CODEC bit to 0. */ vx_outl(chip, CDSP, chip->regCDSP &~ VX_CDSP_CODEC_RESET_MASK); @@ -772,7 +772,7 @@ static void vx2_reset_codec(struct vx_core *_chip) */ static void vx2_change_audio_source(struct vx_core *_chip, int src) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); switch (src) { case VX_AUDIO_SRC_DIGITAL: @@ -791,7 +791,7 @@ static void vx2_change_audio_source(struct vx_core *_chip, int src) */ static void vx2_set_clock_source(struct vx_core *_chip, int source) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (source == INTERNAL_QUARTZ) chip->regCFG &= ~VX_CFG_CLOCKIN_SEL_MASK; @@ -805,7 +805,7 @@ static void vx2_set_clock_source(struct vx_core *_chip, int source) */ static void vx2_reset_board(struct vx_core *_chip, int cold_reset) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* initialize the register values */ chip->regCDSP = VX_CDSP_CODEC_RESET_MASK | VX_CDSP_DSP_RESET_MASK ; @@ -878,7 +878,7 @@ static int vx_input_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); mutex_lock(&_chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->input_level[0]; ucontrol->value.integer.value[1] = chip->input_level[1]; @@ -889,7 +889,7 @@ static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (ucontrol->value.integer.value[0] < 0 || ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; @@ -922,7 +922,7 @@ static int vx_mic_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); ucontrol->value.integer.value[0] = chip->mic_level; return 0; } @@ -930,7 +930,7 @@ static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (ucontrol->value.integer.value[0] < 0 || ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; @@ -973,7 +973,7 @@ static const struct snd_kcontrol_new vx_control_mic_level = { static int vx2_add_mic_controls(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); int err; if (_chip->type != VX_TYPE_MIC) -- 2.12.2

1 2

[alsa-devel] Headphones not detected when using Dell Dock WD15
by Sergio Villar Senin 15 May '17

15 May '17

Hi, I reported this bug to pulseaudio[1] but after some investigation it looks like the alsa mixer is not working for the dock. I'll be happy to provide any additional info which might be required (attaching alsa-info.sh BTW). Please add me to the Cc as I am not subscribed to this list BR [1] https://bugs.freedesktop.org/show_bug.cgi?id=100711

3 11

[alsa-devel] [PATCH 0/4] ALSA/ASoC: fix after merging ASoC PR for v4.12-rc1
by Takashi Sakamoto 15 May '17

15 May '17

Hi, This patchset is for current 'for-linus' remote branch[1], after merging PR from ALSA SoC part for v4.12-rc1[2]. The first two patch includes bug fixes introduced at former kernel releases, the rest is to suppress sparse. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120308.html Takashi Sakamoto (4): ASoC: codecs: da7213: fix invalid usage of bitwise operation in loop condition ASoC: codecs: msm8916: fix invalid cast to bool type ASoC: hisilicon: localize functions without external linkage ASoC: intel: atom: localize variable without external linkage sound/soc/codecs/da7213.c | 14 +++++--------- sound/soc/codecs/msm8916-wcd-analog.c | 6 +++--- sound/soc/hisilicon/hi6210-i2s.c | 11 ++++++----- sound/soc/intel/atom/sst/sst.c | 2 +- 4 files changed, 15 insertions(+), 18 deletions(-) -- 2.9.3

4 12

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Support for one channel (mono) audio
by Peter Ujfalusi 15 May '17

15 May '17

Mono audio can be achieved by configuring McASP to transmit/receive only during one timeslot. McASP will still going to generate clocks for the other slot(s), but will only use the single slot to transmit/receive. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5e02c7a15a83..31b4eac87aa2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -650,7 +650,7 @@ static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream, if (mcasp->tdm_mask[stream]) slots = hweight32(mcasp->tdm_mask[stream]); - for (i = 2; i <= slots; i++) + for (i = 1; i <= slots; i++) list[count++] = i; for (i = 2; i <= serializers; i++) @@ -1318,7 +1318,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - 2, max_channels); + 0, max_channels); snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -1480,13 +1480,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .suspend = davinci_mcasp_suspend, .resume = davinci_mcasp_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, @@ -1992,12 +1992,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) */ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 2), + (32 + mcasp->num_serializer - 1), GFP_KERNEL); mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 2), + (32 + mcasp->num_serializer - 1), GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || -- 2.12.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5514: fix gcc-7 warning
by Arnd Bergmann 15 May '17

15 May '17

gcc-7 warns that there is a duplicate 'const' specifier in some variables that are declared using the SOC_ENUM_SINGLE_DECL macro: sound/soc/codecs/rt5514.c:398:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( sound/soc/codecs/rt5514.c:405:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( This removes one to fix the warning. Fixes: 4a6180ea7399 ("ASoC: rt5514: add rt5514 codec driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/rt5514.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b1ddf0..28ab9e2bb051 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -395,14 +395,14 @@ static const char * const rt5514_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); static const struct snd_kcontrol_new rt5514_sto1_dmic_mux = SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); -- 2.9.0

2 3

[alsa-devel] [PATCH v2] ASoC: ak4613: disable asymmetric audio interface format
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Asymmetric audio interface formats exist in ak4613 by same register settings. Capture Playback 24bit LEFT_J 16bit RIGHT_J 24bit LEFT_J 20bit RIGHT_J 24bit LEFT_J 24bit RIGHT_J 24bit LEFT_J 24bit LEFT_J 24bit I2S 24bit I2S These asymmetric formats makes driver / behavior difficult. It is not HW limitation, but SW limitation. To makes code reading easy, this patch removes asymmetric format support. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - just remove code instead of comment out sound/soc/codecs/ak4613.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index b2dfdde..557ac16 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -139,9 +139,7 @@ struct ak4613_priv { #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ - [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, - [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, - [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + /* [0] - [2] are not supported */ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, }; @@ -262,11 +260,9 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fmt &= SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; - break; default: return -EINVAL; @@ -286,13 +282,8 @@ static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface, if (fmts->fmt != fmt) return false; - if (fmt == SND_SOC_DAIFMT_RIGHT_J) { - if (fmts->width != width) - return false; - } else { - if (fmts->width < width) - return false; - } + if (fmts->width != width) + return false; return true; } @@ -420,8 +411,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) -#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { .name = "ak4613-hifi", -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Move i915 registration to worker thread
by Subhransu S. Prusty 15 May '17

15 May '17

From: Vinod Koul <vinod.koul(a)intel.com> The i915 component framework expects the caller to be invoking snd_hdac_i915_init() from a thread context. Otherwise it results in lockups on drm side. So move the registering of component interface and probing of codecs on this bus to a worker thread. init_failed in skl structure is not used currently, so renamed to init_done and used to track the initialization done in worker thread. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl.c | 162 +++++++++++++++++++++++------------------- sound/soc/intel/skylake/skl.h | 4 +- 2 files changed, 91 insertions(+), 75 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6df3b317a476..4c9b5781282b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -410,7 +410,7 @@ static int skl_free(struct hdac_ext_bus *ebus) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); - skl->init_failed = 1; /* to be sure */ + skl->init_done = 0; /* to be sure */ snd_hdac_ext_stop_streams(ebus); @@ -428,8 +428,10 @@ static int skl_free(struct hdac_ext_bus *ebus) snd_hdac_ext_bus_exit(ebus); + cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) snd_hdac_i915_exit(&ebus->bus); + return 0; } @@ -566,6 +568,84 @@ static int skl_codec_create(struct hdac_ext_bus *ebus) .get_response = snd_hdac_bus_get_response, }; +static int skl_i915_init(struct hdac_bus *bus) +{ + int err; + + /* + * The HDMI codec is in GPU so we need to ensure that it is powered + * up and ready for probe + */ + err = snd_hdac_i915_init(bus); + if (err < 0) + return err; + + err = snd_hdac_display_power(bus, true); + if (err < 0) + dev_err(bus->dev, "Cannot turn on display power on i915\n"); + + return err; +} + +static void skl_probe_work(struct work_struct *work) +{ + struct skl *skl = container_of(work, struct skl, probe_work); + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; + int err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = skl_i915_init(bus); + if (err < 0) + return; + } + + err = skl_init_chip(bus, true); + if (err < 0) { + dev_err(bus->dev, "Init chip failed with err: %d\n", err); + goto out_err; + } + + /* codec detection */ + if (!bus->codec_mask) + dev_info(bus->dev, "no hda codecs found!\n"); + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + return; + } + } + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + return; + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + + /* configure PM */ + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + skl->init_done = 1; + + return; + +out_err: + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + err = snd_hdac_display_power(bus, false); +} + /* * constructor */ @@ -593,6 +673,7 @@ static int skl_create(struct pci_dev *pci, snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); ebus->bus.use_posbuf = 1; skl->pci = pci; + INIT_WORK(&skl->probe_work, skl_probe_work); ebus->bus.bdl_pos_adj = 0; @@ -601,27 +682,6 @@ static int skl_create(struct pci_dev *pci, return 0; } -static int skl_i915_init(struct hdac_bus *bus) -{ - int err; - - /* - * The HDMI codec is in GPU so we need to ensure that it is powered - * up and ready for probe - */ - err = snd_hdac_i915_init(bus); - if (err < 0) - return err; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - - return err; -} - static int skl_first_init(struct hdac_ext_bus *ebus) { struct skl *skl = ebus_to_skl(ebus); @@ -684,20 +744,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) /* initialize chip */ skl_init_pci(skl); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = skl_i915_init(bus); - if (err < 0) - return err; - } - - skl_init_chip(bus, true); - - /* codec detection */ - if (!bus->codec_mask) { - dev_info(bus->dev, "no hda codecs found!\n"); - } - - return 0; + return skl_init_chip(bus, true); } static int skl_probe(struct pci_dev *pci, @@ -706,7 +753,6 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; - struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -729,7 +775,7 @@ static int skl_probe(struct pci_dev *pci, if (skl->nhlt == NULL) { err = -ENODEV; - goto out_display_power_off; + goto out_free; } err = skl_nhlt_create_sysfs(skl); @@ -760,56 +806,24 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_bus_stop_chip(bus); + /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) goto out_dsp_free; - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - goto out_dmic_free; - - /* create codec instances */ - err = skl_codec_create(ebus); - if (err < 0) - goto out_unregister; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - return err; - } - } - - /* - * we are done probling so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); + schedule_work(&skl->probe_work); return 0; -out_unregister: - skl_platform_unregister(bus->dev); -out_dmic_free: - skl_dmic_device_unregister(skl); out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); -out_display_power_off: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, false); out_free: - skl->init_failed = 1; skl_free(ebus); return err; @@ -828,7 +842,7 @@ static void skl_shutdown(struct pci_dev *pci) skl = ebus_to_skl(ebus); - if (skl->init_failed) + if (!skl->init_done) return; snd_hdac_ext_stop_streams(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 5b3fa4b91691..c1968a06174a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -67,7 +67,7 @@ struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; - unsigned int init_failed:1; /* delayed init failed */ + unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; struct snd_soc_platform *platform; @@ -85,6 +85,8 @@ struct skl { const struct firmware *tplg; int supend_active; + + struct work_struct probe_work; }; #define skl_to_ebus(s) (&(s)->ebus) -- 1.9.1

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: Intel: Skylake: Move i915 registration to worker thread
by Subhransu S. Prusty 15 May '17

15 May '17

From: Vinod Koul <vinod.koul(a)intel.com> The i915 component framework expects the caller to be invoking snd_hdac_i915_init() from a thread context. Otherwise it results in lockups on drm side. So move the registering of component interface and probing of codecs on this bus to a worker thread. init_failed in skl structure is not used currently, so renamed to init_done and used to track the initialization done in worker thread. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v1->v2 - no change sound/soc/intel/skylake/skl.c | 162 +++++++++++++++++++++++------------------- sound/soc/intel/skylake/skl.h | 4 +- 2 files changed, 91 insertions(+), 75 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6df3b317a476..4c9b5781282b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -410,7 +410,7 @@ static int skl_free(struct hdac_ext_bus *ebus) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); - skl->init_failed = 1; /* to be sure */ + skl->init_done = 0; /* to be sure */ snd_hdac_ext_stop_streams(ebus); @@ -428,8 +428,10 @@ static int skl_free(struct hdac_ext_bus *ebus) snd_hdac_ext_bus_exit(ebus); + cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) snd_hdac_i915_exit(&ebus->bus); + return 0; } @@ -566,6 +568,84 @@ static int skl_codec_create(struct hdac_ext_bus *ebus) .get_response = snd_hdac_bus_get_response, }; +static int skl_i915_init(struct hdac_bus *bus) +{ + int err; + + /* + * The HDMI codec is in GPU so we need to ensure that it is powered + * up and ready for probe + */ + err = snd_hdac_i915_init(bus); + if (err < 0) + return err; + + err = snd_hdac_display_power(bus, true); + if (err < 0) + dev_err(bus->dev, "Cannot turn on display power on i915\n"); + + return err; +} + +static void skl_probe_work(struct work_struct *work) +{ + struct skl *skl = container_of(work, struct skl, probe_work); + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; + int err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = skl_i915_init(bus); + if (err < 0) + return; + } + + err = skl_init_chip(bus, true); + if (err < 0) { + dev_err(bus->dev, "Init chip failed with err: %d\n", err); + goto out_err; + } + + /* codec detection */ + if (!bus->codec_mask) + dev_info(bus->dev, "no hda codecs found!\n"); + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + return; + } + } + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + return; + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + + /* configure PM */ + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + skl->init_done = 1; + + return; + +out_err: + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + err = snd_hdac_display_power(bus, false); +} + /* * constructor */ @@ -593,6 +673,7 @@ static int skl_create(struct pci_dev *pci, snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); ebus->bus.use_posbuf = 1; skl->pci = pci; + INIT_WORK(&skl->probe_work, skl_probe_work); ebus->bus.bdl_pos_adj = 0; @@ -601,27 +682,6 @@ static int skl_create(struct pci_dev *pci, return 0; } -static int skl_i915_init(struct hdac_bus *bus) -{ - int err; - - /* - * The HDMI codec is in GPU so we need to ensure that it is powered - * up and ready for probe - */ - err = snd_hdac_i915_init(bus); - if (err < 0) - return err; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - - return err; -} - static int skl_first_init(struct hdac_ext_bus *ebus) { struct skl *skl = ebus_to_skl(ebus); @@ -684,20 +744,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) /* initialize chip */ skl_init_pci(skl); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = skl_i915_init(bus); - if (err < 0) - return err; - } - - skl_init_chip(bus, true); - - /* codec detection */ - if (!bus->codec_mask) { - dev_info(bus->dev, "no hda codecs found!\n"); - } - - return 0; + return skl_init_chip(bus, true); } static int skl_probe(struct pci_dev *pci, @@ -706,7 +753,6 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; - struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -729,7 +775,7 @@ static int skl_probe(struct pci_dev *pci, if (skl->nhlt == NULL) { err = -ENODEV; - goto out_display_power_off; + goto out_free; } err = skl_nhlt_create_sysfs(skl); @@ -760,56 +806,24 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_bus_stop_chip(bus); + /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) goto out_dsp_free; - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - goto out_dmic_free; - - /* create codec instances */ - err = skl_codec_create(ebus); - if (err < 0) - goto out_unregister; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - return err; - } - } - - /* - * we are done probling so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); + schedule_work(&skl->probe_work); return 0; -out_unregister: - skl_platform_unregister(bus->dev); -out_dmic_free: - skl_dmic_device_unregister(skl); out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); -out_display_power_off: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, false); out_free: - skl->init_failed = 1; skl_free(ebus); return err; @@ -828,7 +842,7 @@ static void skl_shutdown(struct pci_dev *pci) skl = ebus_to_skl(ebus); - if (skl->init_failed) + if (!skl->init_done) return; snd_hdac_ext_stop_streams(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 5b3fa4b91691..c1968a06174a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -67,7 +67,7 @@ struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; - unsigned int init_failed:1; /* delayed init failed */ + unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; struct snd_soc_platform *platform; @@ -85,6 +85,8 @@ struct skl { const struct firmware *tplg; int supend_active; + + struct work_struct probe_work; }; #define skl_to_ebus(s) (&(s)->ebus) -- 1.9.1

3 3

[alsa-devel] [PATCH] rt286: add Thinkpad Helix 2 to force_combo_jack_table
by Yifeng Li 15 May '17

15 May '17

Thinkpad Helix 2 is a tablet PC, the audio is powered by Core M broadwell-audio and rt286 codec. For all versions of Linux kernel, the stereo output doesn't work properly when earphones are plugged in, the sound was coming out from both channels even if the audio contains only the left or right channel. Furthermore, if a music recorded in stereo is played, the two channels cancle out each other out, as a result, no voice but only distorted background music can be heard, like a sound card with builtin a Karaoke sount effect. Apparently this tablet uses a combo jack with polarity incorrectly set by rt286 driver. This patch adds DMI information of Thinkpad Helix 2 to force_combo_jack_table[] and the issue is resolved. The microphone input doesn't work regardless to the presence of this patch and still needs help from other developers to investigate. This is my first patch to LKML directly, sorry for CC-ing too many people here. Link: https://bugzilla.kernel.org/show_bug.cgi?id=93841 Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Keyon <yang.jie(a)linux.intel.com> Cc: Jinliang Wang <jinliangx.wang(a)intel.com> Cc: stable(a)vger.kernel.org Signed-off-by: Yifeng Li <tomli(a)tomli.me> --- sound/soc/codecs/rt286.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..7899a2cdeb42 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1108,6 +1108,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") } }, + { + .ident = "Thinkpad Helix 2nd", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad Helix 2nd") + } + }, { } }; -- 2.12.1

2 1

[alsa-devel] [PATCH] ASoC: da7213: Fix incorrect usage of bitwise '&' operator for SRM check
by Adam Thomson 15 May '17

15 May '17

In the SRM lock check section of code the '&' bitwise operator is used as part of checking lock status. Functionally the code works as intended, but the conditional statement is a boolean comparison so should really use '&&' logical operator instead. This commit rectifies this discrepancy. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7213.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 12da558..d114f33 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -771,7 +771,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(50); } - } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + } while ((i < DA7213_SRM_CHECK_RETRIES) && (!srm_lock)); if (!srm_lock) dev_warn(codec->dev, "SRM failed to lock\n"); -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec" to the asoc tree
by Mark Brown 15 May '17

15 May '17

The patch ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx 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 51827c41c9ce07293b094691673e6ec23dfdc5e8 Mon Sep 17 00:00:00 2001 From: Tomas Vilda <tomas.vilda(a)gmail.com> Date: Sat, 13 May 2017 00:29:37 +0300 Subject: [PATCH] ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec Signed-off-by: Tomas Vilda <tomas.vilda(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8a90ba8cd71..d7d03c92cb8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { static struct snd_soc_dai_driver dac31xx_dai_driver[] = { { - .name = "tlv32dac31xx-hifi", + .name = "tlv320dac31xx-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, -- 2.11.0

1 0

[alsa-devel] [PATCH][V2] ASoC: Intel: sst: fix spelling mistake: "allocationf" -> "allocation"
by Colin King 15 May '17

15 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Also replace "fail" with "failure". Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 21cac1c8dd4c..e74119113713 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -690,7 +690,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - dev_err(rtd->dev, "dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocation failure\n"); return retval; } } -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: nau8824: TDM support
by John Hsu 15 May '17

15 May '17

Support TDM format for NAU88L24. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 12 +++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cca974d..3a309b1 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1125,6 +1125,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /** + * nau8824_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: Bitmask representing active TX slots. Ex. + * 0xf for normal 4 channel TDM. + * 0xf0 for shifted 4 channel TDM + * @rx_mask: Bitmask [0:1] representing active DACR RX slots. + * Bitmask [2:3] representing active DACL RX slots. + * 00=CH0,01=CH1,10=CH2,11=CH3. Ex. + * 0xf for DACL/R selecting TDM CH3. + * 0xf0 for DACL/R selecting shifted TDM CH3. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Only support 4 slots TDM. + */ +static int nau8824_set_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; + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int tslot_l = 0, ctrl_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf0) && (rx_mask & 0xf)) || + ((rx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf) && (tx_mask & 0xf0))) + return -EINVAL; + + ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + tslot_l = 4 * slot_width; + ctrl_val |= (tx_mask >> 4); + } else { + ctrl_val |= tx_mask; + } + if (rx_mask & 0xf0) + ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT); + else + ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT); + + regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL, + NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN | + NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK | + NAU8824_TDM_TX_MASK, ctrl_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT, + NAU8824_TSLOT_L_MASK, tslot_l); + + return 0; +} + +/** * nau8824_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. * @fs: sampling rate. @@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = { static const struct snd_soc_dai_ops nau8824_dai_ops = { .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, + .set_tdm_slot = nau8824_set_tdm_slot, }; #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 87ac9a3..21eae24 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -258,6 +258,18 @@ #define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) #define NAU8824_I2S_BLK_DIV_MASK 0x7 +/* PORT0_LEFT_TIME_SLOT (0x1E) */ +#define NAU8824_TSLOT_L_MASK 0x3ff + +/* TDM_CTRL (0x20) */ +#define NAU8824_TDM_MODE (0x1 << 15) +#define NAU8824_TDM_OFFSET_EN (0x1 << 14) +#define NAU8824_TDM_DACL_RX_SFT 6 +#define NAU8824_TDM_DACL_RX_MASK (0x3 << NAU8824_TDM_DACL_RX_SFT) +#define NAU8824_TDM_DACR_RX_SFT 4 +#define NAU8824_TDM_DACR_RX_MASK (0x3 << NAU8824_TDM_DACR_RX_SFT) +#define NAU8824_TDM_TX_MASK 0xf + /* ADC_FILTER_CTRL (0x24) */ #define NAU8824_ADC_SYNC_DOWN_MASK 0x3 #define NAU8824_ADC_SYNC_DOWN_32 0 -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Add more width and channels support in the TDM mode
by Oder Chiou 15 May '17

15 May '17

This patch adds more width and channels support in the TDM mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 23 +++++++++++++++++++++-- sound/soc/codecs/rt5514.h | 6 ++++++ 2 files changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b..ff97360 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -906,9 +906,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; - if (slots == 4) + switch (slots) { + case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + break; + + case 6: + val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH; + break; + + case 8: + val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH; + break; + case 2: + default: + break; + } switch (slot_width) { case 20: @@ -919,6 +933,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; break; + case 25: + val |= RT5514_TDM_MODE2; + break; + case 32: val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; break; @@ -930,7 +948,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | - RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | + RT5514_TDM_MODE2, val); return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 5d343fb..02bc212 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -117,6 +117,8 @@ #define RT5514_POW_ADCFEDL_BIT 0 /* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE2 (0x1 << 30) +#define RT5514_TDM_MODE2_SFT 30 #define RT5514_TDM_MODE (0x1 << 28) #define RT5514_TDM_MODE_SFT 28 #define RT5514_I2S_LR_MASK (0x1 << 26) @@ -136,6 +138,8 @@ #define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) #define RT5514_TDMSLOT_SEL_RX_SFT 10 #define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_TDMSLOT_SEL_RX_6CH (0x2 << 10) +#define RT5514_TDMSLOT_SEL_RX_8CH (0x3 << 10) #define RT5514_CH_LEN_RX_MASK (0x3 << 8) #define RT5514_CH_LEN_RX_SFT 8 #define RT5514_CH_LEN_RX_16 (0x0 << 8) @@ -145,6 +149,8 @@ #define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) #define RT5514_TDMSLOT_SEL_TX_SFT 6 #define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_TDMSLOT_SEL_TX_6CH (0x2 << 6) +#define RT5514_TDMSLOT_SEL_TX_8CH (0x3 << 6) #define RT5514_CH_LEN_TX_MASK (0x3 << 4) #define RT5514_CH_LEN_TX_SFT 4 #define RT5514_CH_LEN_TX_16 (0x0 << 4) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5665: add ADC STO2 ASRC support
by Bard Liao 15 May '17

15 May '17

"ADC Stereo2 Filter" is with ASRC supported. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd2230..ece77b8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2684,6 +2684,8 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { RT5665_DAC_MONO_R_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5665_ASRC_1, RT5665_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5665_ASRC_1, + RT5665_ADC_STO2_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono L ASRC", 1, RT5665_ASRC_1, RT5665_ADC_MONO_L_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono R ASRC", 1, RT5665_ASRC_1, @@ -3227,6 +3229,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc}, {"ADC Mono Left Filter", NULL, "ADC Mono L ASRC", is_using_asrc}, {"ADC Mono Right Filter", NULL, "ADC Mono R ASRC", is_using_asrc}, {"DAC Mono Left Filter", NULL, "DAC Mono L ASRC", is_using_asrc}, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

2 2

[alsa-devel] [PATCH][RESEND] ASoC: rsnd: add missing clocks/clock-names on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 3332910..13cb21d5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -368,6 +368,10 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI +- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. +- clock-names : List of necessary clock names. + "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", + "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout -- 1.9.1

1 0

[alsa-devel] [PATCH v2][RESEND] ASoC: rsnd: control kctrl items acceptance anytime/runtime
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SRC/DVC/CTU adds kctrl for each device, and SRC can adjust its sampling rate during playback, thus, this feature should be enabled only *during* playback. This patch controls it more clearly Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 17 +++++++++++++++++ sound/soc/sh/rcar/ctu.c | 6 ++++++ sound/soc/sh/rcar/dvc.c | 5 +++++ sound/soc/sh/rcar/rsnd.h | 16 ++++++++++------ sound/soc/sh/rcar/src.c | 2 ++ 5 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1744015..d3c9078 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -982,6 +982,9 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); int i, change = 0; + if (!cfg->accept(cfg->io)) + return 0; + for (i = 0; i < cfg->size; i++) { if (cfg->texts) { change |= (uc->value.enumerated.item[i] != cfg->val[i]); @@ -998,6 +1001,18 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, return change; } +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) +{ + return 1; +} + +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + return !!runtime; +} + struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; @@ -1016,6 +1031,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -1050,6 +1066,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->texts = texts; cfg->max = max; cfg->size = size; + cfg->accept = accept; cfg->update = update; cfg->card = card; cfg->kctrl = kctrl; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 9dcc1f9..4ba8f2f 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -279,12 +279,14 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* CTU Pass */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + rsnd_kctrl_accept_anytime, NULL, &ctu->pass, RSND_MAX_CHANNELS, 0xC); /* ROW0 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv0, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -293,6 +295,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW1 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv1, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -301,6 +304,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW2 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv2, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -309,6 +313,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW3 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv3, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -317,6 +322,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* Reset */ ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_kctrl_accept_anytime, rsnd_ctu_value_reset, &ctu->reset, 1); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 463de83..75af6e7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -257,6 +257,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->volume, slots, 0x00800000 - 1); @@ -267,6 +268,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->mute, slots, 1); @@ -277,6 +279,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_s(mod, io, rtd, is_play ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) @@ -285,6 +288,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rup, dvc_ramp_rate); @@ -294,6 +298,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rdown, dvc_ramp_rate); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index dbf4163..f68aecb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -594,6 +594,7 @@ struct rsnd_kctrl_cfg { unsigned int size; u32 *val; const char * const *texts; + int (*accept)(struct rsnd_dai_stream *io); void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod); struct rsnd_dai_stream *io; struct snd_card *card; @@ -611,12 +612,15 @@ struct rsnd_kctrl_cfg_s { u32 val; }; +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -624,16 +628,16 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int size, u32 max); -#define rsnd_kctrl_new_m(mod, io, rtd, name, update, cfg, size, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_m(cfg), \ +#define rsnd_kctrl_new_m(mod, io, rtd, name, accept, update, cfg, size, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_m(cfg), \ NULL, size, max) -#define rsnd_kctrl_new_s(mod, io, rtd, name, update, cfg, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_s(mod, io, rtd, name, accept, update, cfg, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ NULL, 1, max) -#define rsnd_kctrl_new_e(mod, io, rtd, name, update, cfg, texts) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_e(mod, io, rtd, name, accept, update, cfg, texts) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ texts, 1, ARRAY_SIZE(texts)) /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e4bde0c..e395fcb 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -489,6 +489,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate Switch" : "SRC In Rate Switch", + rsnd_kctrl_accept_anytime, rsnd_src_set_convert_rate, &src->sen, 1); if (ret < 0) @@ -498,6 +499,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", + rsnd_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); -- 1.9.1

1 0

[alsa-devel] linux-4.12-rc1/sound/soc/codecs/cs35l35.c: false comparison ?
by David Binderman 14 May '17

14 May '17

hello there, [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==1' is always false. [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==3' is always false. Source code is errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; if (classh->classh_wk_fet_disable == 0x00 && (errata_chk == 0x01 || errata_chk == 0x03)) { but ./sound/soc/codecs/cs35l35.h:#define CS35L35_SP_RATE_MASK 0xC0 Regards David Binderman

1 0

[alsa-devel] [sound:topic/for-4.13 22/30] sound/core/pcm.c:905:3: error: implicit declaration of function 'pcm_call_notify'
by kbuild test robot 12 May '17

12 May '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/for-4.13 head: c56407e6d958efda0d2a587ed89e02a251975e06 commit: e9a0fa885872c92e6ada08fb1cf3433945d235a3 [22/30] ALSA: pcm: Build pcm notifier code conditionally config: x86_64-kexec (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout e9a0fa885872c92e6ada08fb1cf3433945d235a3 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/core/pcm.c: In function 'snd_pcm_free': >> sound/core/pcm.c:905:3: error: implicit declaration of function 'pcm_call_notify' [-Werror=implicit-function-declaration] pcm_call_notify(pcm, n_unregister); ^~~~~~~~~~~~~~~ >> sound/core/pcm.c:905:24: error: 'n_unregister' undeclared (first use in this function) pcm_call_notify(pcm, n_unregister); ^~~~~~~~~~~~ sound/core/pcm.c:905:24: note: each undeclared identifier is reported only once for each function it appears in sound/core/pcm.c: In function 'snd_pcm_dev_register': >> sound/core/pcm.c:1118:23: error: 'n_register' undeclared (first use in this function) pcm_call_notify(pcm, n_register); ^~~~~~~~~~ sound/core/pcm.c: In function 'snd_pcm_dev_disconnect': >> sound/core/pcm.c:1147:24: error: 'n_disconnect' undeclared (first use in this function) pcm_call_notify(pcm, n_disconnect); ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pcm_call_notify +905 sound/core/pcm.c 899 900 static int snd_pcm_free(struct snd_pcm *pcm) 901 { 902 if (!pcm) 903 return 0; 904 if (!pcm->internal) > 905 pcm_call_notify(pcm, n_unregister); 906 if (pcm->private_free) 907 pcm->private_free(pcm); 908 snd_pcm_lib_preallocate_free_for_all(pcm); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/3] ALSA: USB-descriptor endianess fixes
by Johan Hovold 12 May '17

12 May '17

These patches add missing endianness conversions when accessing the USB device-descriptor fields. In the process, clean up the us122l driver which had product-id conditionals sprinkled throughout. The final patch drops the Kconfig dependency on x86 since it's a USB driver that compiles just fine on non-x86. Perhaps this should be a (X86 || COMPILE_TEST) dependency instead in case there are external dependencies that warrants this limitation. Note that these patches have only been compile-tested. Johan Johan Hovold (3): ALSA: usb-audio: fix Amanero Combo384 quirk on big-endian hosts ALSA: us122l: clean up US144 handling ALSA: us122l: drop x86 dependency sound/usb/Kconfig | 1 - sound/usb/quirks.c | 2 +- sound/usb/usx2y/us122l.c | 36 ++++++++++++++++++------------------ sound/usb/usx2y/us122l.h | 2 ++ 4 files changed, 21 insertions(+), 20 deletions(-) -- 2.13.0

2 5

[alsa-devel] [PATCH v2] sound: Disable the build of OSS drivers
by Takashi Iwai 12 May '17

12 May '17

OSS drivers are left as badly unmaintained, and now we're facing a problem to clean up the hackish set_fs() usage in their codes. Since most of drivers have been covered by ALSA, and the others are dead old and inactive, let's leave them RIP. This patch is the first step: disable the build of OSS drivers. We'll eventually drop the whole codes and clean up later. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Smaller changes, as Linus and Randy suggested. The first dmasound Kconfig inclusion is still commented out since it's shorter/simpler in the end. sound/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/Kconfig b/sound/Kconfig index ee2e69a9ecd1..de23fc39cec8 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -54,7 +54,7 @@ config SOUND_OSS_CORE_PRECLAIM If unsure, say Y. -source "sound/oss/dmasound/Kconfig" +# source "sound/oss/dmasound/Kconfig" if !M68K && !UML @@ -115,6 +115,7 @@ endif # SND menuconfig SOUND_PRIME tristate "Open Sound System (DEPRECATED)" select SOUND_OSS_CORE + depends on BROKEN help Say 'Y' or 'M' to enable Open Sound System drivers. -- 2.12.2

3 2

[alsa-devel] [sound:topic/kill-set_fs 40/43] sound/soc/blackfin/bf5xx-ac97.c:62:30: error: 'SRC' undeclared
by kbuild test robot 11 May '17

11 May '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/kill-set_fs head: 79778abbaa9e2c939edeee9a9f6e56a519325110 commit: f96cf8a3d3ada486bdef6aca6d8a615429c03d0b [40/43] ASoC: blackfin: Convert to copy_silence ops config: blackfin-allmodconfig (attached as .config) compiler: bfin-uclinux-gcc (GCC) 6.2.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout f96cf8a3d3ada486bdef6aca6d8a615429c03d0b # save the attached .config to linux build tree make.cross ARCH=blackfin All errors (new ones prefixed by >>): sound/soc/blackfin/bf5xx-ac97.c: In function 'bf5xx_pcm_to_ac97': >> sound/soc/blackfin/bf5xx-ac97.c:62:30: error: 'SRC' undeclared (first use in this function) dst->ac97_pcm_l = GET_VAL(SRC); ^ sound/soc/blackfin/bf5xx-ac97.c:47:24: note: in definition of macro 'GET_VAL' #define GET_VAL(src) (*src++) /* copy only */ ^~~ sound/soc/blackfin/bf5xx-ac97.c:62:30: note: each undeclared identifier is reported only once for each function it appears in dst->ac97_pcm_l = GET_VAL(SRC); ^ sound/soc/blackfin/bf5xx-ac97.c:47:24: note: in definition of macro 'GET_VAL' #define GET_VAL(src) (*src++) /* copy only */ ^~~ vim +/SRC +62 sound/soc/blackfin/bf5xx-ac97.c 56 dst->ac97_tag = TAG_VALID; 57 if (chan_mask & SP_FL) { 58 dst->ac97_pcm_r = GET_VAL(src); 59 dst->ac97_tag |= TAG_PCM_RIGHT; 60 } 61 if (chan_mask & SP_FR) { > 62 dst->ac97_pcm_l = GET_VAL(SRC); 63 dst->ac97_tag |= TAG_PCM_LEFT; 64 65 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

Re: [alsa-devel] [PATCH v2 3/3] ASoC: stm32: Add full duplex support to i2s (fwd)
by Julia Lawall 11 May '17

11 May '17

Please check whether the lock taken on line 585 should be freed on line 598. The report also mentions line 652, although that is not shown. julia ---------- Forwarded message ---------- Date: Thu, 11 May 2017 09:49:21 +0800 From: kbuild test robot <fengguang.wu(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: Re: [alsa-devel] [PATCH v2 3/3] ASoC: stm32: Add full duplex support to i2s Hi olivier, [auto build test WARNING on asoc/for-next] [also build test WARNING on next-20170510] [cannot apply to v4.11] [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] url: https://github.com/0day-ci/linux/commits/olivier-moysan/Add-I2S-driver/2017… base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next :::::: branch date: 2 hours ago :::::: commit date: 2 hours ago >> sound/soc/stm/stm32_i2s.c:598:3-9: preceding lock on line 585 >> sound/soc/stm/stm32_i2s.c:598:3-9: preceding lock on line 585 sound/soc/stm/stm32_i2s.c:652:3-9: preceding lock on line 585 git remote add linux-review https://github.com/0day-ci/linux git remote update linux-review git checkout 9f5663c4aa8961bf2a547683af787e606503376e vim +598 sound/soc/stm/stm32_i2s.c 1ebd36ae olivier moysan 2017-05-10 579 { 1ebd36ae olivier moysan 2017-05-10 580 struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); 1ebd36ae olivier moysan 2017-05-10 581 bool playback_flg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); 9f5663c4 olivier moysan 2017-05-10 582 u32 cfg1_mask, ier; 1ebd36ae olivier moysan 2017-05-10 583 int ret; 1ebd36ae olivier moysan 2017-05-10 584 9f5663c4 olivier moysan 2017-05-10 @585 spin_lock(&i2s->lock_fd); 9f5663c4 olivier moysan 2017-05-10 586 1ebd36ae olivier moysan 2017-05-10 587 switch (cmd) { 1ebd36ae olivier moysan 2017-05-10 588 case SNDRV_PCM_TRIGGER_START: 1ebd36ae olivier moysan 2017-05-10 589 case SNDRV_PCM_TRIGGER_RESUME: 1ebd36ae olivier moysan 2017-05-10 590 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: 1ebd36ae olivier moysan 2017-05-10 591 /* Enable i2s */ 1ebd36ae olivier moysan 2017-05-10 592 dev_dbg(cpu_dai->dev, "start I2S\n"); 1ebd36ae olivier moysan 2017-05-10 593 1ebd36ae olivier moysan 2017-05-10 594 ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, 1ebd36ae olivier moysan 2017-05-10 595 I2S_CR1_SPE, I2S_CR1_SPE); 1ebd36ae olivier moysan 2017-05-10 596 if (ret < 0) { 1ebd36ae olivier moysan 2017-05-10 597 dev_err(cpu_dai->dev, "Error %d enabling I2S\n", ret); 1ebd36ae olivier moysan 2017-05-10 @598 return ret; 1ebd36ae olivier moysan 2017-05-10 599 } 1ebd36ae olivier moysan 2017-05-10 600 1ebd36ae olivier moysan 2017-05-10 601 ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, :::::: The code at line 598 was first introduced by commit :::::: 1ebd36ae636af5c130a9df4d9a921e4ed984a6e9 ASoC: stm32: Add I2S driver :::::: TO: olivier moysan <olivier.moysan(a)st.com> :::::: CC: 0day robot <fengguang.wu(a)intel.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: Intel: sst: fix spelling mistake: "allocationf" -> "allocation"
by Colin King 10 May '17

10 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 21cac1c8dd4c..e74119113713 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -690,7 +690,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - dev_err(rtd->dev, "dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocation fail\n"); return retval; } } -- 2.11.0

2 1

[alsa-devel] [PATCH] conf: topology: Fix index mismatch for skl and bxt
by Subhransu S. Prusty 10 May '17

10 May '17

The topology index in the conf is set to 1, where the driver expects index set to 0. Fix the inconsistency. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- src/conf/topology/bxtrt298/bxt_i2s.conf | 62 ++++++++++++++++----------------- src/conf/topology/sklrt286/skl_i2s.conf | 56 ++++++++++++++--------------- 2 files changed, 59 insertions(+), 59 deletions(-) diff --git a/src/conf/topology/bxtrt298/bxt_i2s.conf b/src/conf/topology/bxtrt298/bxt_i2s.conf index 3b7a54e3a323..6b38f4f3c5ae 100644 --- a/src/conf/topology/bxtrt298/bxt_i2s.conf +++ b/src/conf/topology/bxtrt298/bxt_i2s.conf @@ -2924,7 +2924,7 @@ SectionData."lib_data" { SectionControlMixer."media0_in mi Switch" { - index"1" + index"0" invert "false" max "1" min"0" @@ -2944,7 +2944,7 @@ SectionControlMixer."media0_in mi Switch" { } } SectionControlMixer."dmic01_hifi_in mi Switch" { - index"1" + index"0" invert "false" max "1" min"0" @@ -2964,7 +2964,7 @@ SectionControlMixer."dmic01_hifi_in mi Switch" { } } SectionControlMixer."codec0_in mi Switch" { - index"1" + index"0" invert "false" max "1" min"0" @@ -2986,7 +2986,7 @@ SectionControlMixer."codec0_in mi Switch" { SectionWidget."media0_in cpr 0" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2998,7 +2998,7 @@ SectionWidget."media0_in cpr 0" { ] } SectionWidget."media0_in mi" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3011,7 +3011,7 @@ SectionWidget."media0_in mi" { ] } SectionWidget."codec0_in cpr 1" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -3023,7 +3023,7 @@ SectionWidget."codec0_in cpr 1" { ] } SectionWidget."codec0_in mi" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3036,12 +3036,12 @@ SectionWidget."codec0_in mi" { ] } SectionWidget."codec0_in" { - index"1" + index"0" type"aif_in" no_pm "true" } SectionWidget."codec0_out mo" { - index"1" + index"0" type"mixer" no_pm "true" event_type "1" @@ -3059,7 +3059,7 @@ SectionWidget."codec0_out mo" { ] } SectionWidget."codec0_out cpr 2" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3070,12 +3070,12 @@ SectionWidget."codec0_out cpr 2" { ] } SectionWidget."codec0_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."media0_out mo" { - index"1" + index"0" type"mixer" no_pm "true" event_type "1" @@ -3093,7 +3093,7 @@ SectionWidget."media0_out mo" { ] } SectionWidget."media0_out cpr 3" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3104,7 +3104,7 @@ SectionWidget."media0_out cpr 3" { ] } SectionWidget."dmic01_hifi_in cpr 4" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -3116,7 +3116,7 @@ SectionWidget."dmic01_hifi_in cpr 4" { ] } SectionWidget."dmic01_hifi_in mi" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3129,12 +3129,12 @@ SectionWidget."dmic01_hifi_in mi" { ] } SectionWidget."dmic01_hifi" { - index"1" + index"0" type"aif_in" no_pm "true" } SectionWidget."codec1_out mo" { - index"1" + index"0" type"mixer" no_pm "true" event_type "1" @@ -3152,7 +3152,7 @@ SectionWidget."codec1_out mo" { ] } SectionWidget."codec1_out cpr 5" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3163,12 +3163,12 @@ SectionWidget."codec1_out cpr 5" { ] } SectionWidget."codec1_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."hdmi1_pt_out cpr 6" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -3180,7 +3180,7 @@ SectionWidget."hdmi1_pt_out cpr 6" { ] } SectionWidget."hdmi1_pt_out cpr 7" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3191,12 +3191,12 @@ SectionWidget."hdmi1_pt_out cpr 7" { ] } SectionWidget."iDisp1_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."hdmi2_pt_out cpr 8" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -3208,7 +3208,7 @@ SectionWidget."hdmi2_pt_out cpr 8" { ] } SectionWidget."hdmi2_pt_out cpr 9" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3219,12 +3219,12 @@ SectionWidget."hdmi2_pt_out cpr 9" { ] } SectionWidget."iDisp2_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."hdmi3_pt_out cpr 10" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -3236,7 +3236,7 @@ SectionWidget."hdmi3_pt_out cpr 10" { ] } SectionWidget."hdmi3_pt_out cpr 11" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3247,12 +3247,12 @@ SectionWidget."hdmi3_pt_out cpr 11" { ] } SectionWidget."iDisp3_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."mch_cap_in cpr 12" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -3264,7 +3264,7 @@ SectionWidget."mch_cap_in cpr 12" { ] } SectionWidget."mch_cap_in cpr 13" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -3283,7 +3283,7 @@ SectionManifest."lib_data" { } SectionGraph."Pipeline 1 Graph" { - index"1" + index"0" lines [ "media0_in mi, , media0_in cpr 0" "media0_in cpr 0, , System Playback" diff --git a/src/conf/topology/sklrt286/skl_i2s.conf b/src/conf/topology/sklrt286/skl_i2s.conf index bb1fe4f8435f..fbf041bfd0cf 100644 --- a/src/conf/topology/sklrt286/skl_i2s.conf +++ b/src/conf/topology/sklrt286/skl_i2s.conf @@ -2604,7 +2604,7 @@ SectionData."hdmi3_pt_out cpr 12" { SectionControlMixer."media0_in mi Switch" { - index"1" + index"0" invert "false" max "1" min"0" @@ -2624,7 +2624,7 @@ SectionControlMixer."media0_in mi Switch" { } } SectionControlMixer."codec0_in mi Switch" { - index"1" + index"0" invert "false" max "1" min"0" @@ -2644,7 +2644,7 @@ SectionControlMixer."codec0_in mi Switch" { } } SectionControlMixer."dmic01_hifi_in mi Switch" { - index"1" + index"0" invert "false" max "1" min"0" @@ -2666,7 +2666,7 @@ SectionControlMixer."dmic01_hifi_in mi Switch" { SectionWidget."media0_in cpr 0" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2678,7 +2678,7 @@ SectionWidget."media0_in cpr 0" { ] } SectionWidget."media0_in mi" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2691,7 +2691,7 @@ SectionWidget."media0_in mi" { ] } SectionWidget."media0_out mo" { - index"1" + index"0" type"mixer" no_pm "true" event_type "1" @@ -2709,7 +2709,7 @@ SectionWidget."media0_out mo" { ] } SectionWidget."media0_out cpr 6" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2720,7 +2720,7 @@ SectionWidget."media0_out cpr 6" { ] } SectionWidget."codec0_out mo" { - index"1" + index"0" type"mixer" no_pm "true" event_type "1" @@ -2738,7 +2738,7 @@ SectionWidget."codec0_out mo" { ] } SectionWidget."codec0_out cpr 4" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2749,12 +2749,12 @@ SectionWidget."codec0_out cpr 4" { ] } SectionWidget."codec0_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."codec1_out mo" { - index"1" + index"0" type"mixer" no_pm "true" event_type "1" @@ -2772,7 +2772,7 @@ SectionWidget."codec1_out mo" { ] } SectionWidget."codec1_out cpr 5" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2783,12 +2783,12 @@ SectionWidget."codec1_out cpr 5" { ] } SectionWidget."codec1_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."codec0_in cpr 1" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2800,7 +2800,7 @@ SectionWidget."codec0_in cpr 1" { ] } SectionWidget."codec0_in mi" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2813,12 +2813,12 @@ SectionWidget."codec0_in mi" { ] } SectionWidget."codec0_in" { - index"1" + index"0" type"aif_in" no_pm "true" } SectionWidget."dmic01_hifi_in cpr 3" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2830,7 +2830,7 @@ SectionWidget."dmic01_hifi_in cpr 3" { ] } SectionWidget."dmic01_hifi_in mi" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2843,12 +2843,12 @@ SectionWidget."dmic01_hifi_in mi" { ] } SectionWidget."dmic01_hifi" { - index"1" + index"0" type"aif_in" no_pm "true" } SectionWidget."hdmi1_pt_out cpr 7" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2860,7 +2860,7 @@ SectionWidget."hdmi1_pt_out cpr 7" { ] } SectionWidget."hdmi1_pt_out cpr 8" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2871,12 +2871,12 @@ SectionWidget."hdmi1_pt_out cpr 8" { ] } SectionWidget."iDisp1_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."hdmi2_pt_out cpr 9" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2888,7 +2888,7 @@ SectionWidget."hdmi2_pt_out cpr 9" { ] } SectionWidget."hdmi2_pt_out cpr 10" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2899,12 +2899,12 @@ SectionWidget."hdmi2_pt_out cpr 10" { ] } SectionWidget."iDisp2_out" { - index"1" + index"0" type"aif_out" no_pm "true" } SectionWidget."hdmi3_pt_out cpr 11" { - index"1" + index"0" type"mixer" no_pm "true" event_type "3" @@ -2916,7 +2916,7 @@ SectionWidget."hdmi3_pt_out cpr 11" { ] } SectionWidget."hdmi3_pt_out cpr 12" { - index"1" + index"0" type"pga" no_pm "true" event_type "4" @@ -2927,7 +2927,7 @@ SectionWidget."hdmi3_pt_out cpr 12" { ] } SectionGraph."Pipeline 1 Graph" { - index"1" + index"0" lines [ "media0_in mi, , media0_in cpr 0" "media0_in cpr 0, , System Playback" -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix cpu lockup when stopping the cmd dmas
by jeeja.kp＠intel.com 10 May '17

10 May '17

From: Jeeja KP <jeeja.kp(a)intel.com> Using jiffies in hdac_wait_for_cmd_dmas() to determine when to time out when interrupts are off (snd_hdac_bus_stop_cmd_io()/spin_lock_irq()) causes hard lockup so unlock while waiting using jiffies. ---<-snip->--- <0>[ 1211.603046] NMI watchdog: Watchdog detected hard LOCKUP on cpu 3 <4>[ 1211.603047] Modules linked in: snd_hda_intel i915 vgem <4>[ 1211.603053] irq event stamp: 13366 <4>[ 1211.603053] hardirqs last enabled at (13365): ... <4>[ 1211.603059] Call Trace: <4>[ 1211.603059] ? delay_tsc+0x3d/0xc0 <4>[ 1211.603059] __delay+0xa/0x10 <4>[ 1211.603060] __const_udelay+0x31/0x40 <4>[ 1211.603060] snd_hdac_bus_stop_cmd_io+0x96/0xe0 [snd_hda_core] <4>[ 1211.603060] ? azx_dev_disconnect+0x20/0x20 [snd_hda_intel] <4>[ 1211.603061] snd_hdac_bus_stop_chip+0xb1/0x100 [snd_hda_core] <4>[ 1211.603061] azx_stop_chip+0x9/0x10 [snd_hda_codec] <4>[ 1211.603061] azx_suspend+0x72/0x220 [snd_hda_intel] <4>[ 1211.603061] pci_pm_suspend+0x71/0x140 <4>[ 1211.603062] dpm_run_callback+0x6f/0x330 <4>[ 1211.603062] ? pci_pm_freeze+0xe0/0xe0 <4>[ 1211.603062] __device_suspend+0xf9/0x370 <4>[ 1211.603062] ? dpm_watchdog_set+0x60/0x60 <4>[ 1211.603063] async_suspend+0x1a/0x90 <4>[ 1211.603063] async_run_entry_fn+0x34/0x160 <4>[ 1211.603063] process_one_work+0x1f4/0x6d0 <4>[ 1211.603063] ? process_one_work+0x16e/0x6d0 <4>[ 1211.603064] worker_thread+0x49/0x4a0 <4>[ 1211.603064] kthread+0x107/0x140 <4>[ 1211.603064] ? process_one_work+0x6d0/0x6d0 <4>[ 1211.603065] ? kthread_create_on_node+0x40/0x40 <4>[ 1211.603065] ret_from_fork+0x2e/0x40 Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=100419 Reported-by: Marta Lofstedt <marta.lofstedt(a)intel.com> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/hda/hdac_controller.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 6f1e99c..0f703372 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -106,7 +106,11 @@ void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus) /* disable ringbuffer DMAs */ snd_hdac_chip_writeb(bus, RIRBCTL, 0); snd_hdac_chip_writeb(bus, CORBCTL, 0); + spin_unlock_irq(&bus->reg_lock); + hdac_wait_for_cmd_dmas(bus); + + spin_lock_irq(&bus->reg_lock); /* disable unsolicited responses */ snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, 0); spin_unlock_irq(&bus->reg_lock); -- 2.5.0

3 2

[alsa-devel] `alsatplg` miscompiles UUID in ASoC topology
by ojab 09 May '17

09 May '17

Oh hai! I'm trying to get sound from my HP Chromebook 13 with proper linux and right now messing with ASoC topology: ChromeOS contains topology file that can't be loaded on up-to-date upstream kernel, so I've wrote a parser that converts dfw_sst.bin back to topology.conf that can be processed by `alsatplg`. AFAICS alsatplg miscompiles at least UUID field in the private data: for example `alsatplg -c /usr/share/alsa/topology/sklrt286/skl_i2s.conf -o tplg.bin` produces the file with no UUID in `strings tplg.bin` output and my parser shows that there is nonsense (mostly zeroes) where UUID should be (reproducible with both alsa-{lib,utils}-1.13 and current git master). Is it a bug or I'm doing something wrong? -- //wbr ojab

1 0

[alsa-devel] BUG/Oops triggered by zita-a2j on snd-aloop with 4.10.13
by Jörn Nettingsmeier 08 May '17

08 May '17

Hi *, openSUSE Tumbleweed with kernel Linux hoppetosse.villakunterbunt.netz 4.10.13-1-default #1 SMP PREEMPT Thu Apr 27 12:23:31 UTC 2017 (e5d11ce) x86_64 x86_64 x86_64 GNU/Linux jackd running on a hdspm MADIface, I try to add a zita-a2j client with zita-a2j -v -d hw:jackbridge,1,1 ("jackbridge" is a snd-aloop device) nettings@hoppetosse:~> aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC269VC Analog [ALC269VC 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 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: ma2chboxXT [RME MADIface], device 0: RME MADIface [RME MADIface] Subdevices: 0/1 Subdevice #0: subdevice #0 card 4: jackbridge [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 4: jackbridge [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 [ 1416.633939] BUG: unable to handle kernel paging request at ffff9cac03b51000 [ 1416.634046] IP: snd_pcm_format_set_silence+0xea/0x100 [snd_pcm] [ 1416.634092] PGD 17fc9d067 [ 1416.634093] PUD 17fc9e067 [ 1416.634117] PMD 18314b067 [ 1416.634139] PTE 0 [ 1416.634194] Oops: 0002 [#1] PREEMPT SMP [ 1416.634226] Modules linked in: snd_hrtimer snd_seq_midi snd_seq_midi_event snd_seq_dummy snd_seq ccm fuse af_packet nf_log_ipv6 xt_comment nf_log_ipv4 nf_log_common xt_LOG xt_limit uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_v4l2 videobuf2_core videodev nfnetlink_cthelper nfnetlink ip6t_REJECT nf_reject_ipv6 nf_conntrack_ipv6 nf_defrag_ipv6 ipt_REJECT nf_reject_ipv4 xt_pkttype xt_tcpudp iptable_filter ip6table_mangle nf_conntrack_netbios_ns nf_conntrack_broadcast nf_conntrack_ipv4 nf_defrag_ipv4 ip_tables xt_conntrack nf_conntrack ip6table_filter ip6_tables x_tables msr snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic intel_rapl x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm irqbypass crct10dif_pclmul crc32_pclmul mei_wdt iTCO_wdt iTCO_vendor_support arc4 ghash_clmulni_intel [ 1416.634838] pcbc iwldvm mac80211 aesni_intel aes_x86_64 iwlwifi crypto_simd glue_helper cryptd snd_hda_intel e1000e snd_hdspm snd_hda_codec cfg80211 thinkpad_acpi ptp lpc_ich snd_rawmidi ac rfkill joydev mei_me snd_seq_device snd_hda_core pcspkr battery i2c_i801 mfd_core pps_core mei snd_hwdep shpchp fjes thermal wmi tpm_tis tpm_tis_core tpm snd_aloop snd_pcm snd_timer snd soundcore btrfs xor raid6_pq crc32c_intel i915 serio_raw i2c_algo_bit drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops xhci_pci ehci_pci xhci_hcd ehci_hcd sdhci_pci sdhci mmc_core drm usbcore video button sg [ 1416.635255] CPU: 0 PID: 3387 Comm: zita-a2j Not tainted 4.10.13-1-default #1 [ 1416.635312] Hardware name: LENOVO 2325UR7/2325UR7, BIOS G2ETA2WW (2.62 ) 09/12/2014 [ 1416.635378] task: ffff8d6757c160c0 task.stack: ffff9cac03b74000 [ 1416.635444] RIP: 0010:snd_pcm_format_set_silence+0xea/0x100 [snd_pcm] [ 1416.635494] RSP: 0018:ffff9cac03b77d50 EFLAGS: 00010282 [ 1416.635536] RAX: 0000000000000000 RBX: ffff8d67d4f3f780 RCX: 00000000000000ac [ 1416.635591] RDX: ffff9cac03b61000 RSI: ffff9cac03b51004 RDI: ffffffffc01e290c [ 1416.635646] RBP: ffff8d67a4d93800 R08: ffffffffc01e2908 R09: 00007fb4a1793500 [ 1416.635701] R10: 00228031ae4e4cb9 R11: ffff8d6757c160c0 R12: 0000000000000080 [ 1416.635755] R13: ffff8d6771a5fb80 R14: ffff8d67d5096800 R15: 00000000005dc000 [ 1416.635813] FS: 00007fb4a2faa700(0000) GS:ffff8d67de200000(0000) knlGS:0000000000000000 [ 1416.635875] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 1416.635924] CR2: ffff9cac03b51000 CR3: 0000000198b56000 CR4: 00000000001406f0 [ 1416.636001] Call Trace: [ 1416.636054] loopback_prepare+0x2d6/0x2f0 [snd_aloop] [ 1416.636110] snd_pcm_do_prepare+0x13/0x30 [snd_pcm] [ 1416.636158] snd_pcm_action_single+0x2a/0x70 [snd_pcm] [ 1416.636207] ? snd_pcm_action_single+0x5d/0x70 [snd_pcm] [ 1416.636255] snd_pcm_action_nonatomic+0x61/0x70 [snd_pcm] [ 1416.636308] snd_pcm_common_ioctl1+0x6d8/0xc60 [snd_pcm] [ 1416.636357] ? __wake_up+0x34/0x50 [ 1416.636388] ? tty_write+0x228/0x2b0 [ 1416.636424] snd_pcm_capture_ioctl1+0x117/0x280 [snd_pcm] [ 1416.636474] ? __vfs_write+0x23/0x140 [ 1416.636510] snd_pcm_capture_ioctl+0x2a/0x30 [snd_pcm] [ 1416.636555] do_vfs_ioctl+0x8f/0x5d0 [ 1416.636589] ? __fget+0x70/0xc0 [ 1416.636616] SyS_ioctl+0x74/0x80 [ 1416.636651] entry_SYSCALL_64_fastpath+0x1e/0xad [ 1416.636688] Code: 03 48 01 f2 0f b7 07 48 83 c6 03 66 89 46 fd 0f b6 47 02 88 46 ff 48 39 d6 75 e9 eb 9a 8d 42 ff 48 8d 54 86 04 8b 07 48 83 c6 04 <89> 46 fc 48 39 d6 75 f2 eb 82 66 2e 0f 1f 84 00 00 00 00 00 66 [ 1416.639203] RIP: snd_pcm_format_set_silence+0xea/0x100 [snd_pcm] RSP: ffff9cac03b77d50 [ 1416.641594] CR2: ffff9cac03b51000 [ 1416.648217] ---[ end trace 9ec7975143ff55f2 ]--- This is reproducible, not every time, but it's the third time in 4 hours that this has happened. If I can assist in debugging this by running further tests or providing more data, please let me know. Best, Jörn -- Jörn Nettingsmeier De Rijpgracht 8, 1055VR Amsterdam, Nederland Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio), Tonmeister VDT http://stackingdwarves.net

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Add mute led support for HP EliteBook 840 G3
by Florin Tudorache 08 May '17

08 May '17

Signed-off-by: Florin Tudorache <florin_tudorache(a)live.com> --- sound/pci/hda/patch_conexant.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e825373..63bc894 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -860,7 +860,9 @@ static const struct hda_fixup cxt_fixups[] = { { 0x16, 0x21011020 }, /* line-out */ { 0x18, 0x2181103f }, /* line-in */ { } - } + }, + .chained = true, + .chain_id = CXT_FIXUP_MUTE_LED_GPIO, }, [CXT_FIXUP_HP_SPECTRE] = { .type = HDA_FIXUP_PINS, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: ak4613: disable asymmetric audio interface format
by Kuninori Morimoto 08 May '17

08 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Asymmetric audio interface formats exist in ak4613 by same register settings. Capture Playback 24bit LEFT_J 16bit RIGHT_J 24bit LEFT_J 20bit RIGHT_J 24bit LEFT_J 24bit RIGHT_J 24bit LEFT_J 24bit LEFT_J 24bit I2S 24bit I2S These asymmetric formats makes driver / behavior difficult. It is not HW limitation, but SW limitation. To makes code reading easy, this patch doesn't remove code, but just comment out. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index b2dfdde..9d2be7c 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -139,9 +139,9 @@ struct ak4613_priv { #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ - [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, - [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, - [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, +/* [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, */ +/* [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, */ +/* [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, */ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, }; @@ -262,7 +262,7 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fmt &= SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { - case SND_SOC_DAIFMT_RIGHT_J: +/* case SND_SOC_DAIFMT_RIGHT_J:*/ case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; @@ -420,7 +420,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) -#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ +#define AK4613_PCM_FMTBIT (/*SNDRV_PCM_FMTBIT_S16_LE |*/ \ SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { -- 1.9.1

2 2

[alsa-devel] Novation - Focursrite Scarlet 6i6 Gen 2 Support request
by Joel Wirāmu Pauling 07 May '17

07 May '17

Hi all, I have been attempting to build up as much information as possible on the 2nd gen 6i6 in the hopes someone on the mailing list will be able to find a quick fix. The device is usable under linux only as a 2ch output device in it's current state using snd_usb_audio . And only if you have set Hardware mixer routing from the Proprietary Focus-rite app in Mac OSX or Windows first. The following link contains various USB Capture dumps doing different things with the DAC in various sates. There appears to be something odd with the way it enumerates sub devices on the USB port so attempts to pass-through to a Win10 VM via KVM fail. I think I can probably get this working if I pass the underlying PCI root device controlling the USB hub in it's entirety but haven't been able to do this yet. It also looks like there is a firmware load event happening somewhere. I am hoping the lsusb and the wireshark USB pcap dumps at the link might be of use to someone more familiar with alsa. Any help or requests for more information gladly accepted. I am away for the next week - but will try and get a successful init dump out of a windows laptop and add to the repository of dumps. https://drive.google.com/drive/folders/0Bw2fu74VAaO5bksyUEhMZDBtUE0?usp=sha… Kind regards -Joel New Zealand http://linkedin.com/in/aenertia

1 0

[alsa-devel] [PATCH v2] topology: Fix issue in parsing routes when generating topology binary
by fuweix.tang＠intel.com 05 May '17

05 May '17

From: Fuwei Tang <fuweix.tang(a)intel.com> We missed parsing the index value, which is used as a use case indicator, when processing the route objects. This patch fixes the bug. Signed-off-by: Fuwei Tang <fuweix.tang(a)intel.com> Tested-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- src/topology/dapm.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 13aa1c4..6af750b 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -381,7 +381,7 @@ done: } -static int tplg_parse_routes(snd_tplg_t *tplg, snd_config_t *cfg) +static int tplg_parse_routes(snd_tplg_t *tplg, snd_config_t *cfg, int index) { snd_config_iterator_t i, next; snd_config_t *n; @@ -399,7 +399,7 @@ static int tplg_parse_routes(snd_tplg_t *tplg, snd_config_t *cfg) elem = tplg_elem_new_route(tplg); if (!elem) return -ENOMEM; - + elem->index = index; line = elem->route; err = tplg_parse_line(val, line); @@ -419,7 +419,8 @@ int tplg_parse_dapm_graph(snd_tplg_t *tplg, snd_config_t *cfg, snd_config_iterator_t i, next; snd_config_t *n; int err; - const char *graph_id; + const char *graph_id, *val = NULL; + int index; if (snd_config_get_type(cfg) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("error: compound is expected for dapm graph definition\n"); @@ -436,8 +437,14 @@ int tplg_parse_dapm_graph(snd_tplg_t *tplg, snd_config_t *cfg, continue; } + if (strcmp(id, "index") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + index = atoi(val); + } + if (strcmp(id, "lines") == 0) { - err = tplg_parse_routes(tplg, n); + err = tplg_parse_routes(tplg, n, index); if (err < 0) { SNDERR("error: failed to parse dapm graph %s\n", graph_id); -- 2.7.4

2 1

[alsa-devel] [PATCH] topology: Fix issue in parsing routes when generating topology binary
by fuweix.tang＠intel.com 05 May '17

05 May '17

From: Fuwei Tang <fuweix.tang(a)intel.com> We missed parsing the index value, which is used as a use case indicator, when processing the route objects. This patch fixes the bug. Signed-off-by: Fuwei Tang <fuweix.tang(a)intel.com> Tested-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- src/topology/dapm.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 13aa1c4..2fa387f 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -381,7 +381,7 @@ done: } -static int tplg_parse_routes(snd_tplg_t *tplg, snd_config_t *cfg) +static int tplg_parse_routes(snd_tplg_t *tplg, snd_config_t *cfg, int index) { snd_config_iterator_t i, next; snd_config_t *n; @@ -399,7 +399,7 @@ static int tplg_parse_routes(snd_tplg_t *tplg, snd_config_t *cfg) elem = tplg_elem_new_route(tplg); if (!elem) return -ENOMEM; - + elem->index = index; line = elem->route; err = tplg_parse_line(val, line); @@ -419,7 +419,8 @@ int tplg_parse_dapm_graph(snd_tplg_t *tplg, snd_config_t *cfg, snd_config_iterator_t i, next; snd_config_t *n; int err; - const char *graph_id; + const char *graph_id, *val = NULL; + int index; if (snd_config_get_type(cfg) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("error: compound is expected for dapm graph definition\n"); @@ -436,8 +437,14 @@ int tplg_parse_dapm_graph(snd_tplg_t *tplg, snd_config_t *cfg, continue; } + if (strcmp(id, "index") == 0) { + if (snd_config_get_string(n, &val) < 0) + return NULL; + index = atoi(val); + } + if (strcmp(id, "lines") == 0) { - err = tplg_parse_routes(tplg, n); + err = tplg_parse_routes(tplg, n, index); if (err < 0) { SNDERR("error: failed to parse dapm graph %s\n", graph_id); -- 2.7.4

3 2

[alsa-devel] [PATCH] ALSA: hda: Use loop counter for hdac_wait_for_cmd_dmas() timeout
by Chris Wilson 05 May '17

05 May '17

hdac_wait_for_cmd_dmas() uses a jiffie timeout to ensure that we do not wait forever for stuck hardware. However, it is called from an irq-disabled context which prevents jiffie from advancing and so the loop doesn't terminate if the hardware fails. This can then cause NMI watchdog warnings, such as: NMI watchdog: Watchdog detected hard LOCKUP on cpu 3 Modules linked in: snd_hda_intel i915 vgem snd_hda_codec_hdmi x86_pkg_temp_thermal intel_powerclamp coretemp crct10dif_pclmul crc32_pclmul snd_hda_codec_realtek snd_hda_codec_generic ghash_clmulni_intel e1000e snd_hda_codec snd_hwdep snd_hda_core snd_pcm ptp mei_me prime_numbers pps_core mei lpc_ich i2c_hid i2c_designware_platform i2c_designware_core [last unloaded: i915] irq event stamp: 13366 hardirqs last enabled at (13365): [<ffffffff81891a87>] _raw_spin_unlock_irq+0x27/0x50 hardirqs last disabled at (13366): [<ffffffff818918d2>] _raw_spin_lock_irq+0x12/0x50 softirqs last enabled at (12744): [<ffffffff81085c79>] __do_softirq+0x1d9/0x4c0 softirqs last disabled at (12721): [<ffffffff810860d9>] irq_exit+0xa9/0xc0 CPU: 3 PID: 10443 Comm: kworker/u8:11 Tainted: G U 4.11.0-rc4-CI-CI_DRM_319+ #1 Hardware name: /NUC5i5RYB, BIOS RYBDWi35.86A.0362.2017.0118.0940 01/18/2017 Workqueue: events_unbound async_run_entry_fn task: ffff88024cd32740 task.stack: ffffc9000162c000 RIP: 0010:preempt_count_add+0xe/0xc0 RSP: 0018:ffffc9000162fbd8 EFLAGS: 00000082 RAX: 0000000080000001 RBX: 0000000704b96558 RCX: 0000000000000002 RDX: 0000000000000000 RSI: ffffffff81c74f2d RDI: 0000000000000001 RBP: ffffc9000162fc08 R08: 00000000bbcc90cc R09: 23c7b07100000000 R10: ffffffff827901a8 R11: ffff88024cd32740 R12: 0000000704b92baa R13: 0000000000003ea0 R14: 0000000000000003 R15: ffffffffa00061f0 FS: 0000000000000000(0000) GS:ffff880256d80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f90f84a5144 CR3: 0000000003e0f000 CR4: 00000000003406e0 Call Trace: ? delay_tsc+0x3d/0xc0 __delay+0xa/0x10 __const_udelay+0x31/0x40 snd_hdac_bus_stop_cmd_io+0x96/0xe0 [snd_hda_core] ? azx_dev_disconnect+0x20/0x20 [snd_hda_intel] snd_hdac_bus_stop_chip+0xb1/0x100 [snd_hda_core] azx_stop_chip+0x9/0x10 [snd_hda_codec] azx_suspend+0x72/0x220 [snd_hda_intel] pci_pm_suspend+0x71/0x140 dpm_run_callback+0x6f/0x330 ? pci_pm_freeze+0xe0/0xe0 __device_suspend+0xf9/0x370 ? dpm_watchdog_set+0x60/0x60 async_suspend+0x1a/0x90 async_run_entry_fn+0x34/0x160 process_one_work+0x1f4/0x6d0 ? process_one_work+0x16e/0x6d0 worker_thread+0x49/0x4a0 kthread+0x107/0x140 ? process_one_work+0x6d0/0x6d0 ? kthread_create_on_node+0x40/0x40 ret_from_fork+0x2e/0x40 Fixes: 38b19ed7f81e ("ALSA: hda: fix to wait for RIRB & CORB DMA to set") Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=100419 Signed-off-by: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: Jeeja KP <jeeja.kp(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: <stable(a)vger.kernel.org> # v4.7+ --- sound/hda/hdac_controller.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index ee08c389b4d6..7f8806b03982 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -85,14 +85,14 @@ static void hdac_wait_for_cmd_dmas(struct hdac_bus *bus) { unsigned long timeout; - timeout = jiffies + msecs_to_jiffies(100); + timeout = 100 * 100; /* 100ms */ while ((snd_hdac_chip_readb(bus, RIRBCTL) & AZX_RBCTL_DMA_EN) - && time_before(jiffies, timeout)) + && timeout--) udelay(10); - timeout = jiffies + msecs_to_jiffies(100); + timeout = 100 * 100; /* 100ms */ while ((snd_hdac_chip_readb(bus, CORBCTL) & AZX_CORBCTL_RUN) - && time_before(jiffies, timeout)) + && timeout--) udelay(10); } -- 2.11.0

4 5

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix spelling mistake: "plalform" -> "platform"
by Colin King 04 May '17

04 May '17

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..a8a81fda4da8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -779,7 +779,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) dev_dbg(dev, "%s()\n", __func__); if (!hcd) { - dev_err(dev, "%s: No plalform data\n", __func__); + dev_err(dev, "%s: No platform data\n", __func__); return -EINVAL; } -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: da7218: Fix incorrect usage of bitwise '&' operator for SRM check
by Adam Thomson 03 May '17

03 May '17

In the SRM lock check section of code the '&' bitwise operator is used as part of checking lock status. Functionally the code works as intended, but the conditional statement is a boolean comparison so should really use '&&' logical operator instead. This commit rectifies this discrepancy. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- 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 d256ebf..6e1940e 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1457,7 +1457,7 @@ static int da7218_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(DA7218_SRM_CHECK_DELAY); } - } while ((i < DA7218_SRM_CHECK_TRIES) & (!success)); + } while ((i < DA7218_SRM_CHECK_TRIES) && (!success)); if (!success) dev_warn(codec->dev, "SRM failed to lock\n"); -- 1.9.1

2 2

[alsa-devel] [PATCH v2 00/11] drm/i915: LPE audio runtime PM and multipipe (v2)
by ville.syrjala＠linux.intel.com 03 May '17

03 May '17

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Okay, here's the second attempt at getting multiple pipes playing back audio on the VLV/CHV HDMI LPE audio device. The main change from v1 is that now the PCM devices are associated with ports instead of pipes, so the audio from one device always gets output on the same display. I've also tacked on the alsa-lib conf update. No clue whether it's really correct or not (the config language isn't a close friend of mine). BTW I did notice that with LPE audio all the controls say iface=PCM, whereas on HDA a bunch of them say iface=MIXER. No idea if that's OK or not, just something I spotted when I was comparing the results with HDA. Entire series available here: git://github.com/vsyrjala/linux.git lpe_audio_multipipe_2 Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Ville Syrjälä (11): drm/i915: Fix runtime PM for LPE audio ALSA: x86: Clear the pdata.notify_lpe_audio pointer before teardown drm/i915: Stop pretending to mask/unmask LPE audio interrupts drm/i915: Remove the unused pending_notify from LPE platform data drm/i915: Replace tmds_clock_speed and link_rate with just ls_clock drm/i915: Remove hdmi_connected from LPE audio pdata drm/i915: Reorganize intel_lpe_audio_notify() arguments drm/i915: Clean up the LPE audio platform data ALSA: x86: Prepare LPE audio ctls for multiple PCMs ALSA: x86: Split snd_intelhad into card and PCM specific structures ALSA: x86: Register multiple PCM devices for the LPE audio card drivers/gpu/drm/i915/i915_drv.h | 4 +- drivers/gpu/drm/i915/i915_irq.c | 15 +- drivers/gpu/drm/i915/intel_audio.c | 19 +- drivers/gpu/drm/i915/intel_lpe_audio.c | 99 ++++------ include/drm/intel_lpe_audio.h | 22 +-- sound/x86/intel_hdmi_audio.c | 328 ++++++++++++++++++++------------- sound/x86/intel_hdmi_audio.h | 20 +- 7 files changed, 271 insertions(+), 236 deletions(-) -- 2.10.2

4 24

[alsa-devel] 1.1.4 release?
by Takashi Iwai 03 May '17

03 May '17

Hi Jaroslav, how about the 1.1.4 release? It's been a while since 1.1.3, and we've gathered already enough changes. thanks, Takashi

1 0

[alsa-devel] [PATCH 1/2] conf: Remove dmix.direct_memory_access setup
by Takashi Iwai 03 May '17

03 May '17

The commit [22eca6468b4a: pcm: dmix: Allow disabling x86 optimizations] introduced the new flag for dmix & co, direct_memory_access. However, it turned out that such an addition of the new flag in the default pcm definition causes an error when it's used with old alsa-lib codes. Although the code added here is correct, per se, and it works as expected, it's not wise to break the configuration with old stuff -- even if the usage is somehow incorrect and should be avoided. Since the usage of the new flag is only for HDMI LPE audio, and the usage of dmix itself should be limited with that hardware, this patch removes the setup so that it works with the old alsa-lib again. We may introduce the dmix behavior change in a smarter way, e.g. passing some flag from the hardware driver so that it works more generically without the manual fiddling of config files. Bugzilla: http://bugzilla.opensuse.org/show_bug.cgi?id=1037021 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf/cards/HdmiLpeAudio.conf | 23 ----------------------- src/conf/pcm/dmix.conf | 15 --------------- 2 files changed, 38 deletions(-) diff --git a/src/conf/cards/HdmiLpeAudio.conf b/src/conf/cards/HdmiLpeAudio.conf index 61bdfeae2917..dae71facac6d 100644 --- a/src/conf/cards/HdmiLpeAudio.conf +++ b/src/conf/cards/HdmiLpeAudio.conf @@ -20,29 +20,6 @@ HdmiLpeAudio.pcm.front.0 { } } -# uncached memory reads have a high penalty -HdmiLpeAudio.dmix.direct_memory_access false - -# default with dmix+softvol -HdmiLpeAudio.pcm.default { - @args [ CARD ] - @args.CARD { - type string - } - type plug - slave.pcm { - type softvol - slave.pcm { - @func concat - strings [ "dmix:" $CARD ] - } - control { - name "PCM Playback Volume" - card $CARD - } - } -} - <confdir:pcm/surround40.conf> <confdir:pcm/surround21.conf> <confdir:pcm/surround41.conf> diff --git a/src/conf/pcm/dmix.conf b/src/conf/pcm/dmix.conf index 2d3b329e533a..7d0aa0158c42 100644 --- a/src/conf/pcm/dmix.conf +++ b/src/conf/pcm/dmix.conf @@ -49,21 +49,6 @@ pcm.!dmix { @func refer name defaults.pcm.ipc_perm } - direct_memory_access { - @func refer - name { - @func concat - strings [ - "cards." - { - @func card_driver - card $CARD - } - ".dmix.direct_memory_access" - ] - } - default true - } slave { pcm { type hw -- 2.12.2

1 1

[alsa-devel] [PATCH] sound: core: move exports next to corresponding symbols
by Seraphime Kirkovski 03 May '17

03 May '17

With this patch all files in sound/core export symbols next to their definition, as the kernel coding style mandates. Signed-off-by: Seraphime Kirkovski <kirkseraph(a)gmail.com> --- sound/core/memalloc.c | 14 ++++---------- sound/core/seq/seq_midi_emul.c | 9 ++++----- sound/core/seq/seq_midi_event.c | 21 ++++++++------------- sound/core/timer.c | 27 +++++++++++++-------------- 4 files changed, 29 insertions(+), 42 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f05cb6a8cbe0..207de142ee3d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -54,6 +54,7 @@ void *snd_malloc_pages(size_t size, gfp_t gfp_flags) pg = get_order(size); return (void *) __get_free_pages(gfp_flags, pg); } +EXPORT_SYMBOL(snd_malloc_pages); /** * snd_free_pages - release the pages @@ -71,6 +72,7 @@ void snd_free_pages(void *ptr, size_t size) pg = get_order(size); free_pages((unsigned long) ptr, pg); } +EXPORT_SYMBOL(snd_free_pages); /* * @@ -217,6 +219,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->bytes = size; return 0; } +EXPORT_SYMBOL(snd_dma_alloc_pages); /** * snd_dma_alloc_pages_fallback - allocate the buffer area according to the given type with fallback @@ -254,7 +257,7 @@ int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size, return -ENOMEM; return 0; } - +EXPORT_SYMBOL(snd_dma_alloc_pages_fallback); /** * snd_dma_free_pages - release the allocated buffer @@ -287,13 +290,4 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) pr_err("snd-malloc: invalid device type %d\n", dmab->dev.type); } } - -/* - * exports - */ -EXPORT_SYMBOL(snd_dma_alloc_pages); -EXPORT_SYMBOL(snd_dma_alloc_pages_fallback); EXPORT_SYMBOL(snd_dma_free_pages); - -EXPORT_SYMBOL(snd_malloc_pages); -EXPORT_SYMBOL(snd_free_pages); diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 7ba937399ac7..9e2912e3e80f 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -236,6 +236,7 @@ snd_midi_process_event(struct snd_midi_op *ops, break; } } +EXPORT_SYMBOL(snd_midi_process_event); /* @@ -409,6 +410,7 @@ snd_midi_channel_set_clear(struct snd_midi_channel_set *chset) chan->drum_channel = 0; } } +EXPORT_SYMBOL(snd_midi_channel_set_clear); /* * Process a rpn message. @@ -701,6 +703,7 @@ struct snd_midi_channel_set *snd_midi_channel_alloc_set(int n) } return chset; } +EXPORT_SYMBOL(snd_midi_channel_alloc_set); /* * Reset the midi controllers on a particular channel to default values. @@ -724,6 +727,7 @@ void snd_midi_channel_free_set(struct snd_midi_channel_set *chset) kfree(chset->channels); kfree(chset); } +EXPORT_SYMBOL(snd_midi_channel_free_set); static int __init alsa_seq_midi_emul_init(void) { @@ -736,8 +740,3 @@ static void __exit alsa_seq_midi_emul_exit(void) module_init(alsa_seq_midi_emul_init) module_exit(alsa_seq_midi_emul_exit) - -EXPORT_SYMBOL(snd_midi_process_event); -EXPORT_SYMBOL(snd_midi_channel_set_clear); -EXPORT_SYMBOL(snd_midi_channel_alloc_set); -EXPORT_SYMBOL(snd_midi_channel_free_set); diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 37db7ba492a6..90bbbdbeba03 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -134,6 +134,7 @@ int snd_midi_event_new(int bufsize, struct snd_midi_event **rdev) *rdev = dev; return 0; } +EXPORT_SYMBOL(snd_midi_event_new); void snd_midi_event_free(struct snd_midi_event *dev) { @@ -142,6 +143,7 @@ void snd_midi_event_free(struct snd_midi_event *dev) kfree(dev); } } +EXPORT_SYMBOL(snd_midi_event_free); /* * initialize record @@ -161,6 +163,7 @@ void snd_midi_event_reset_encode(struct snd_midi_event *dev) reset_encode(dev); spin_unlock_irqrestore(&dev->lock, flags); } +EXPORT_SYMBOL(snd_midi_event_reset_encode); void snd_midi_event_reset_decode(struct snd_midi_event *dev) { @@ -170,6 +173,7 @@ void snd_midi_event_reset_decode(struct snd_midi_event *dev) dev->lastcmd = 0xff; spin_unlock_irqrestore(&dev->lock, flags); } +EXPORT_SYMBOL(snd_midi_event_reset_decode); #if 0 void snd_midi_event_init(struct snd_midi_event *dev) @@ -183,6 +187,7 @@ void snd_midi_event_no_status(struct snd_midi_event *dev, int on) { dev->nostat = on ? 1 : 0; } +EXPORT_SYMBOL(snd_midi_event_no_status); /* * resize buffer @@ -232,6 +237,7 @@ long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long return result; } +EXPORT_SYMBOL(snd_midi_event_encode); /* * read one byte and encode to sequencer event: @@ -307,6 +313,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, spin_unlock_irqrestore(&dev->lock, flags); return rc; } +EXPORT_SYMBOL(snd_midi_event_encode_byte); /* encode note event */ static void note_event(struct snd_midi_event *dev, struct snd_seq_event *ev) @@ -408,6 +415,7 @@ long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long return qlen; } } +EXPORT_SYMBOL(snd_midi_event_decode); /* decode note event */ @@ -524,19 +532,6 @@ static int extra_decode_xrpn(struct snd_midi_event *dev, unsigned char *buf, return idx; } -/* - * exports - */ - -EXPORT_SYMBOL(snd_midi_event_new); -EXPORT_SYMBOL(snd_midi_event_free); -EXPORT_SYMBOL(snd_midi_event_reset_encode); -EXPORT_SYMBOL(snd_midi_event_reset_decode); -EXPORT_SYMBOL(snd_midi_event_no_status); -EXPORT_SYMBOL(snd_midi_event_encode); -EXPORT_SYMBOL(snd_midi_event_encode_byte); -EXPORT_SYMBOL(snd_midi_event_decode); - static int __init alsa_seq_midi_event_init(void) { return 0; diff --git a/sound/core/timer.c b/sound/core/timer.c index 6d4fbc439246..f7fb15790d0a 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -319,6 +319,7 @@ int snd_timer_open(struct snd_timer_instance **ti, *ti = timeri; return 0; } +EXPORT_SYMBOL(snd_timer_open); /* * close a timer instance @@ -384,6 +385,7 @@ int snd_timer_close(struct snd_timer_instance *timeri) mutex_unlock(&register_mutex); return 0; } +EXPORT_SYMBOL(snd_timer_close); unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) { @@ -398,6 +400,7 @@ unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) } return 0; } +EXPORT_SYMBOL(snd_timer_resolution); static void snd_timer_notify1(struct snd_timer_instance *ti, int event) { @@ -589,6 +592,7 @@ int snd_timer_start(struct snd_timer_instance *timeri, unsigned int ticks) else return snd_timer_start1(timeri, true, ticks); } +EXPORT_SYMBOL(snd_timer_start); /* * stop the timer instance. @@ -602,6 +606,7 @@ int snd_timer_stop(struct snd_timer_instance *timeri) else return snd_timer_stop1(timeri, true); } +EXPORT_SYMBOL(snd_timer_stop); /* * start again.. the tick is kept. @@ -617,6 +622,7 @@ int snd_timer_continue(struct snd_timer_instance *timeri) else return snd_timer_start1(timeri, false, 0); } +EXPORT_SYMBOL(snd_timer_continue); /* * pause.. remember the ticks left @@ -628,6 +634,7 @@ int snd_timer_pause(struct snd_timer_instance * timeri) else return snd_timer_stop1(timeri, false); } +EXPORT_SYMBOL(snd_timer_pause); /* * reschedule the timer @@ -809,6 +816,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) if (use_tasklet) tasklet_schedule(&timer->task_queue); } +EXPORT_SYMBOL(snd_timer_interrupt); /* @@ -859,6 +867,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, *rtimer = timer; return 0; } +EXPORT_SYMBOL(snd_timer_new); static int snd_timer_free(struct snd_timer *timer) { @@ -978,6 +987,7 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstam } spin_unlock_irqrestore(&timer->lock, flags); } +EXPORT_SYMBOL(snd_timer_notify); /* * exported functions for global timers @@ -993,11 +1003,13 @@ int snd_timer_global_new(char *id, int device, struct snd_timer **rtimer) tid.subdevice = 0; return snd_timer_new(NULL, id, &tid, rtimer); } +EXPORT_SYMBOL(snd_timer_global_new); int snd_timer_global_free(struct snd_timer *timer) { return snd_timer_free(timer); } +EXPORT_SYMBOL(snd_timer_global_free); int snd_timer_global_register(struct snd_timer *timer) { @@ -1007,6 +1019,7 @@ int snd_timer_global_register(struct snd_timer *timer) dev.device_data = timer; return snd_timer_dev_register(&dev); } +EXPORT_SYMBOL(snd_timer_global_register); /* * System timer @@ -2119,17 +2132,3 @@ static void __exit alsa_timer_exit(void) module_init(alsa_timer_init) module_exit(alsa_timer_exit) - -EXPORT_SYMBOL(snd_timer_open); -EXPORT_SYMBOL(snd_timer_close); -EXPORT_SYMBOL(snd_timer_resolution); -EXPORT_SYMBOL(snd_timer_start); -EXPORT_SYMBOL(snd_timer_stop); -EXPORT_SYMBOL(snd_timer_continue); -EXPORT_SYMBOL(snd_timer_pause); -EXPORT_SYMBOL(snd_timer_new); -EXPORT_SYMBOL(snd_timer_notify); -EXPORT_SYMBOL(snd_timer_global_new); -EXPORT_SYMBOL(snd_timer_global_free); -EXPORT_SYMBOL(snd_timer_global_register); -EXPORT_SYMBOL(snd_timer_interrupt); -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: snd_soc_set_dmi_name will use DMI_SYS_VENDOR if available
by mengdong.lin＠linux.intel.com 02 May '17

02 May '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This fix is based on Daniel Drake's suggestion: most x86 machines have a meaningfull DMI_SYS_VENDOR rather than DMI_BOARD_VENDOR. So snd_soc_set_dmi_name will try to get vendor name from DMI_SYS_VENDOR at first, and if that's null, then use DMI_BOARD_VENDOR. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 525f2f3..45f0a34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1974,7 +1974,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) return 0; /* long name already set by driver or from DMI */ /* make up dmi long name as: vendor.product.version.board */ - vendor = dmi_get_system_info(DMI_BOARD_VENDOR); + vendor = dmi_get_system_info(DMI_SYS_VENDOR); + if (!vendor) + vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; -- 2.5.0

4 3

[alsa-devel] snd_soc_set_dmi_name - Shouldn't it use SYS_VENDOR?
by Daniel Drake 02 May '17

02 May '17

Hi, Nice to see DMI info being attached to SoC sound devices so that we can differentiate between differerent products in userspace, thanks for the effort there. However I just noticed that it using DMI_BOARD_VENDOR. Is this intentional? I have several examples of products here where DMI_BOARD_VENDOR is not set to a meaningful value, but DMI_SYS_VENDOR is just fine. DMI quirks in the kernel ordinarily tend to match products by combining DMI_SYS_VENDOR with DMI_PRODUCT_NAME. In the SMBIOS data format, Type 1 (System information) includes DMI_SYS_VENDOR and DMI_PRODUCT_NAME. Type 2 (Base board information) includes DMI_BOARD_VENDOR and DMI_BOARD_NAME. See dmi_decode() in drivers/firmware/dmi_scan.c It seems strange that ASoC is pulling the product name from Type 1 but the vendor from type 2. Can we make it instead just use Type 1 data, DMI_SYS_VENDOR and DMI_PRODUCT_NAME? Thanks Daniel

6 10

[alsa-devel] [PATCH v3 0/2] Add STM32 SAI support
by olivier moysan 02 May '17

02 May '17

This patch-set handles the Serial Audio Interface (SAI) IP on STM32 platforms. The SAI IP is composed of two Sub-block A and B. (see the figure below) Each sub-block makes use of individual and shared resources. Shared resources: - Reset line. Reset common and sub-block registers. - Bus interface clocks. This clock is not exposed in device as it it managed by clock driver, according to kernel clocks gating. - Common register. SAI IP exhibits a common configuration register to manage synchronization modes. NB: These modes are not yet implemented in this version of the SAI driver. - Interrupt. Sub-blocks have their own interrupt status registers but they share the same interrupt line. Individual resources: - Register set - DMA request line. - Communication interface. Each sub-block has its own GPIOs and associated bus lines. - Kernel clock. Each sub-block has its own dedicated clock for its communication interface. To reflect this architecture Sub-block A and B are handled by 2 child devices. Sub-block A and B devices can be configured independently either as transmitter or receiver. A PCM device is associated to each sub-block. A sub-block has to select the appropriated parent clock at runtime, depending on the audio stream sampling rate to be played or captured. Two parent clocks must be provided to support sampling rates multiples of 8 kHz or 11.025kHz. Interface clock +--------------------------------+ +---------------> | SAI IP | Reset | +-----------------+ | +---------------> | |common registers | | | +-----------------+ | | | | +----------------------+ | | | Sub-block A | | Kernel clock A | | | | Bus A +--------------------> | +----------------+ +----------------> | | |A registers | | | | | +----------------+ | | | +----------------------+ | | | | +----------------------+ | | | Sub-block B | | Kernel clock B | | | | Bus B +--------------------> | +----------------+ +-----------------> | | |B registers | | | | | +----------------+ | | | +----------------------+ | | | +--------------------------------+ v3: correct DT bindings - remove aliases - use audio graph card olivier moysan (2): ASoC: stm32: add bindings for SAI ASoC: stm32: add SAI driver .../devicetree/bindings/sound/st,stm32-sai.txt | 89 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 6 + sound/soc/stm/stm32_sai.c | 115 +++ sound/soc/stm/stm32_sai.h | 200 +++++ sound/soc/stm/stm32_sai_sub.c | 884 +++++++++++++++++++++ 8 files changed, 1304 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_sai.c create mode 100644 sound/soc/stm/stm32_sai.h create mode 100644 sound/soc/stm/stm32_sai_sub.c -- 1.9.1

4 7

[alsa-devel] new codec tda1543
by George Makdessi 02 May '17

02 May '17

Commit 9f79c0289fa4b02544c6a51c789c84edd6a479fc ("audio: add support for philips tda1543 dac chip") new simple driver for the old well known philips tda1543 dac chip. Hello, We would like to contribute with an audio codec file for the well known Philips TDA1543 dac chip. We have already made a PR for Raspbian with Raspberry Pi specific driver that is well suited together with this codec. We have already got OK confirmation for our PR request and we just need to upstream this codec to finalize the system for our new commersial dac module for Raspberyy PI. https://github.com/McDaisy/linux-1/commit/9f79c0289fa4b02544c6a51c789c84edd… Thanks! BR George The code: /* * Driver for the tda1543 codec * * Author: Georgios F <georgios(a)gdis.se> * Copyright 2017 * based on code by Florian Meier <florian.meier(a)koalo.de> * * 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/init.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/soc.h> static struct snd_soc_dai_driver tda1543_dai = { .name = "tda1543-hifi", .playback = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE }, }; static struct snd_soc_codec_driver soc_codec_dev_tda1543; static int tda1543_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_tda1543, &tda1543_dai, 1); } static int tda1543_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } static const struct of_device_id tda1543_of_match[] = { { .compatible = "ti,tda1543", }, { } }; MODULE_DEVICE_TABLE(of, tda1543_of_match); static struct platform_driver tda1543_codec_driver = { .probe = tda1543_probe, .remove = tda1543_remove, .driver = { .name = "tda1543-codec", .owner = THIS_MODULE, .of_match_table = of_match_ptr(tda1543_of_match), }, }; module_platform_driver(tda1543_codec_driver); MODULE_AUTHOR("Georgios F <georgios(a)gdis.se>"); MODULE_DESCRIPTION("ASoC tda1543 codec driver"); MODULE_LICENSE("GPL v2");

2 2

[alsa-devel] [GIT PULL] ASoC updates for v4.12
by Mark Brown 02 May '17

02 May '17

The following changes since commit 5a7ad1146caa895ad718a534399e38bd2ba721b7: Linux 4.11-rc8 (2017-04-23 16:53:00 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.12 for you to fetch changes up to 20d5c84bef067b7e804a163e2abca16c47125bad: Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8978' and 'asoc/topic/zte-tdm' into asoc-next (2017-04-30 22:16:48 +0900) ---------------------------------------------------------------- ASoC: Updates for v4.12 A quiet release for the core, but lots of new drivers this time around: - A new, generalized, API for hooking up jacks which makes it easier to write generic machine drivers for simple cases. - Continuing fixes for issues with the x86 CPU drivers. - New drivers for Cirrus CS35L35, DIO DIO2125, Everest ES7132, HiSilicon hi6210, Maxim MAX98927, MT2701 systems with WM8960, Nuvoton NAU8824, Odroid systems, ST STM32 SAI controllers and x86 systems with DA7213 ---------------------------------------------------------------- Adrian Dinu (1): ASoC: Add space around '=' Adriana Constantinescu (1): ASoC: omap: Remove unnecessary 'out of memory' message Alexander Sverdlin (1): ASoC: cs4271: configure reset GPIO as output Alin Grigorean (1): ASoC: fsl: Remove unneeded init of static variable Andy Green (1): ASoC: hisilicon: Add hi6210 i2s audio driver Andy Shevchenko (1): ASoC: Intel: bdw-rt5677: Use devm_gpiod_get() Arnaud Pouliquen (1): ASoC: STI: Fix null ptr deference in IRQ handler Arnd Bergmann (2): ASoC: mediatek: add I2C dependency for CS42XX8 ASoC: Intel: Skylake: fix uninitialized pointer use Axel Lin (1): ASoC: cs35l35: Fix display revision id B, Jayachandran (1): ALSA: hda: Fix LLCH register read Baoyou Xie (2): ASoC: zx-tdm: add bindings doc for zte's tdm controller ASoC: zx-tdm: add zte's tdm controller driver Bard Liao (9): ASoC: rt5665: increase button detection accuracy ASoC: rt5665: enable TDM if more than 2 channels ASoC: rt5665: enhance jack type detection function ASoC: rt5665: move set_pll to codec level ASoC: rt5665: move set_sysclk to codec level ASoC: rt5665: set i2s pin share configuration ASoC: rt5665: fix wrong pre div reg of IF2 and IF3 ASoC: jack: add snd_soc_codec_set_jack ASoC: rt5665: move rt5665_set_jack_detect to .set_jack Bhumika Goyal (5): ASoC: pxa: constify snd_soc_ops structures ASoC: fsl: constify snd_soc_ops structures ASoC: mediatek: constify snd_soc_ops structures ASoC: omap: constify snd_soc_ops structures ASoC: blackfin: constify snd_soc_ops structures Brian Austin (2): ASoC: Add support for Cirrus Logic CS35L35 Amplifier ASoC: cs35l35: Add bindings file for CS35L35 Bryce Ferguson (1): ASoC: Add AU1761 audio codec as selectable option Calin Cruceru (1): ASoC: samsung: Remove extra blank lines Charles Keepax (12): ASoC: cs35l35: Add missing return in probe ASoC: cs35l35: Add device tree binding for I2S drive configuration ASoC: cs35l35: Add for configuring drive mode in unused slots ASoC: cs35l35: Add local variable for dev in probe ASoC: cs35l35: Add IRQF_SHARED to IRQ flags ASoC: cs35l35: Stash dev pointer directly rather than CODEC pointer ASoC: cs35l35: Clear reset_gpio on the error path in probe ASoC: cs35l35: Improve power down time ASoC: cs35l35: Correct handling of PDN_DONE with external boost ASoC: cs35l35: Add DT binding to specify usage of an external boost supply ASoC: cs35l35: Correct some register defaults ASoC: cs35l35: Allow user to configure IMON SCALE Codrut Grosu (17): ASoC: tegra: Remove unnecessary 'out of memory' message ASoC: pxa-ssp: Added blank line after declarations ASoC: pxa: Remove space before semicolon ASoC: pxa: Remove unneeded else after return statement ASoC: pxa: The open brace is placed with the previous line ASoC: pxa: Add space around ':' and '(' ASoC: pxa: Remove spaces before tabs ASoC: pxa: Remove unneeded return statement in void function ASoC: pxa-ssp: Line up *s in block comments ASoC: tegra: Add blank line after declarations ASoC: txx9: Added blank line after declarations ASoC: ux500: Added */ to the next line ASoC: ux500: Remove unuseful break after return ASoC: ux500: Added blank line after declarations ASoC: sirf: Added blank line after declarations ASoC: pxa: Add space around '=' ASoC: txx9: Added requiered spaces. Colin Ian King (6): ASoC: Intel: Skylake: fix spelling mistake: "allocationf" -> "allocation" ASoC: cs35l35: trivial fix to indentation ASoC: intel: remove unused variable data and associated code ASoC: Intel: bxtn: fix spelling mistake: "Timout" -> "Timeout" ASoC: topology: use j for internal loop counter ASoC: wm5100: fix spelling mistake: "micropone" -> "microphone" Dan Carpenter (4): ASoC: cs35l35: returning uninitialized in probe() ASoC: Intel: Skylake: Uninitialized variable in probe_codec() ASoC: Intel: Skylake: Fix a couple user after free bugs ASoC: sti: Fix error handling if of_clk_get() fails Daniel Baluta (8): ASoC: codec: wm8960: Refactor sysclk freq search ASoC: codec: wm8960: Relax bit clock computation ASoC: imx-wm8962: Let codec driver enable/disable its MCLK ASoC: imx-wm8962: Fix codec_clk cleanup ASoC: codec: wm9860: Refactor PLL out freq search ASoC: codec: wm8960: Stop when a matching PLL freq is found ASoC: codec: wm9860: avoid maybe-uninitialized warning ASoC: codec: wm8960: Relax bit clock computation when using PLL Dharageswari R (1): ASoC: Intel: Skylake: Fix parameter overwrite for KPB Module Dmitry Torokhov (1): ASoC: jack - check status of GPIO-based pins on resume Douglas Anderson (3): ASoC: rt5514: Mark rt5514_i2c_driver as static ASoC: rt5514: Avoid relying on uninitialized "val" value ASoC: rt5514: Unconfuse the rt5514 at probe / resume time Fabio Estevam (11): ASoC: tas2552: Improve DT binding document example ASoC: fsl_ssi: Remove FSLSSI_I2S_RATES definition ASoC: fsl_ssi: Use the tolower() function ASoC: sgtl5000: Allow LRCLK pad drive strength to be changed ASoC: tas2552: Return the real error code ASoC: tas2552: Describe the possible I2C addresses ASoC: tas2552: Fix typos in the binding doc ASoC: fsl ssi doc: Move optional properties to the correct section ASoC: fsl_esai: Remove unneeded definition ASoC: imx-wm8962: Remove global variables ASoC: tas2552: Propagate the error code in suspend/resume G Kranthi (5): ASoC: Intel: Skylake: Disable notifications at boot after DSP FW init ASoC: Intel: Skylake: Remove get dsp_ops in cleanup routine ASoC: Intel: Skylake: Add 16-bit constraint to FE bxt_rt298 machine ASoC: Intel: Skylake: Move sst common initialization to a helper function ASoC: Intel: Skylake: Modify arguments to reuse module transfer function Guneshwor Singh (2): ASoC: Intel: Skylake: Fix not to stop src pipe in pre pmd event handler ASoC: Intel: Skylake: Return negative error code Hardik T Shah (1): ASoC: Intel: Skylake: Fix DMA position reporting for capture stream Harsha Priya (1): ASoC: Intel: Update bxt_da7219_max98357a to add a new Hiroyuki Yokoyama (1): ASoC: rcar: enable PCM RATE untile 192000 Icenowy Zheng (2): ASoC: sun8i-codec-analog: split out mic2 ASoC: sun8i-codec-analog: split out line in Javier Martinez Canillas (8): ASoC: max9867: export OF device ID as module aliases ASoC: cs53l30: Set .of_match_table to OF device ID table ASoc: rt5645: Add OF device ID table ASoC: ssm4567: Add OF device ID table ASoC: sta529: Add OF device ID table ASoC: uda1380: Add OF device ID table ASoC: wm8978: Add OF device ID table ASoC: rt5677: Add OF device ID table Jeeja KP (13): ASoC: Intel: bxtn: Store the FW/Library context at boot ASoC: Intel: bxtn: optimize ROM init retries ASoC: Intel: Skylake: Fix to delete DSP pipe after stopping pipe ASoC: Intel: bxtn: Disable interrupt when DSP is in D3 ASoC: Intel: bxtn: Update DSP core state in D0 ASoC: Intel: bxtn: Reload the firmware in case of D3 failure ASoC: Intel: Skylake: Remove BE prepare ops ASoC: Intel: Skylake: Fix module load when module size > DMA buffer size ASoC: Intel: Skylake: Use the sig_bits to define dai bps capability ASoC: hdac_hdmi: Update sig_bits based on converter capability ASoC: Intel: Skylake: Rearrangement of code to cleanup SKL SST library ASoC: Intel: Skylake: Fix module state after unbind and delete ASoC: Intel: Skylake: Add support for deferred DSP module bind Jerome Brunet (5): ASoC: es7134: add es7134 DAC driver ASoC: es7134: add dt-bindings for the es7134 dac ASoC: dio2125: add dio2125 amp driver ASoC: dio2125: add dt-bindings ASoC: dio2125: use gpiod_set_value_cansleep Jianqun Xu (1): ASoC: rockchip: add bindings for rk3368 i2s John Hsu (4): ASoC: nau8540: fix tab conversion problem ASoC: nau8824: new driver ASoC: nau8824: rename controls to match DAPM controls ASoC: nau8824: leave Class D gain at chip default John Stultz (5): ASoC: add hi6210-i2s DT bindings ASoC: Improve hi6210-i2s DT bindings ASoC: hisilicon: Add error returns even for cases that shouldn't happen. ASoC: hisilicon: Address style nit to use break in final default of switch statement ASoC: hisilicon: Use devm_snd_soc_register_component Jose Abreu (1): ASoC: dwc: Disallow building designware_pcm as a module Kuninori Morimoto (15): ASoC: simple-card: use defined dev on probe() ASoC: simple-scu-card: use defined dev on probe() ASoC: simple-card: add new simple_priv_to_card() macro ASoC: simple-scu-card: add new simple_priv_to_card() macro ASoC: core: remove pointless auxiliary from snd_soc_component ASoC: rcar: ssi: don't set SSICR.CKDV = 000 with SSIWSR.CONT ASoC: rcar: remove rsnd_kctrl_remove() ASoC: rcar: fixup of_clk_add_provider() usage for multi clkout ASoC: rcar: call missing of_clk_del_provider() when remove ASoC: soc-core: verify Sound Card normality ASoC: rsnd: tidyup src->convert_rate reset timing ASoC: rsnd: merge rsnd_kctrl_new_m/s/e into rsnd_kctrl_new() ASoC: ak4613: use snd_soc_update_bits() to avoid Reserve bit on I/O CTRL ASoC: rsnd: rsnd_mod_make_sure() is not under DEBUG ASoC: rsnd: enable clock-frequency for both 44.1kHz/48kHz Linus Walleij (3): ASoC: ux500: drop platform DAI assignments ASoC: wm8903: add regulator handling ASoC: qcom: move clock names into LPASS variant struct Liviu Dudau (1): ASoC: TLV320AIC23: Unquote NULL from control name Lucas Stach (1): ASoC: imx-pcm-dma: simplify pcm_config Marek Vasut (1): ASoC: rsnd: Fix possible NULL pointer dereference Mark Brown (24): Merge branch 'fix/sun8i' of git://git.kernel.org/.../broonie/sound into asoc-sunxi Merge branch 'topic/hda' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'fix/rcar' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'topic/jack' of git://git.kernel.org/.../broonie/sound into asoc-rt5665 Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/topology' and 'asoc/fix/sti' into asoc-linus Merge tag 'asoc-fix-v4.11-rc7' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/cs4271' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5665' into asoc-next Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/ak4613', 'asoc/topic/au1761' and 'asoc/topic/blackfin' into asoc-next Merge remote-tracking branches 'asoc/topic/cs35l35', 'asoc/topic/cs53l30', 'asoc/topic/da7213', 'asoc/topic/dio2125' and 'asoc/topic/dwc' into asoc-next Merge remote-tracking branches 'asoc/topic/es7134', 'asoc/topic/es8328', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc' and 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-ssi', 'asoc/topic/hi6220' and 'asoc/topic/imx' into asoc-next Merge remote-tracking branches 'asoc/topic/max9878', 'asoc/topic/max98927', 'asoc/topic/mtk' and 'asoc/topic/nau8540' into asoc-next Merge remote-tracking branches 'asoc/topic/nau8824', 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next Merge remote-tracking branches 'asoc/topic/rockchip', 'asoc/topic/rt5514', 'asoc/topic/rt5645' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/simple', 'asoc/topic/sirf' and 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/sta529', 'asoc/topic/sti', 'asoc/topic/stm', 'asoc/topic/sunxi' and 'asoc/topic/tas2552' into asoc-next Merge remote-tracking branches 'asoc/topic/tegra', 'asoc/topic/tlv320aic23', 'asoc/topic/topology', 'asoc/topic/twl6040' and 'asoc/topic/txx9' into asoc-next Merge remote-tracking branches 'asoc/topic/uda1380', 'asoc/topic/unregister', 'asoc/topic/ux500', 'asoc/topic/wm5100' and 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8978' and 'asoc/topic/zte-tdm' into asoc-next Mayuresh Kulkarni (1): ASoC: wm_adsp: add support for DSP region lock Mousumi Jana (1): ASoC: topology: Fix to store enum text values Mylène Josserand (5): ASoC: sun8i-codec: Remove analog "HP" widget ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE ASoC: sun8i-codec: Convert to SOC_MIXER_ARRAY ASoC: sun8i-codec: Fix space on audio-routing widget ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN Nicole Faerber (1): ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk Peter Ujfalusi (1): ASoC: twl6040: Add control for HS and HF mono to stereo selection Pierre-Louis Bossart (7): ASoC: da7213: add ACPI support ASoC: Intel: add machine driver for BYT/CHT + DA7213 ASoC: Intel: Atom: enable BYT/CHT+DA7213 machine driver ASoC: Intel: boards: add card for MinnowBoardMax/Up I2S access ASoC: Intel: Enable bytcht_nocodec machine driver ASoC: Intel: Atom: update Thinkpad 10 quirk ASoC: bytcr_rt5640: log quirk configuration errors Richard Fitzgerald (1): ASoC: wm_adsp: Add support for ADSP2V2 Romain Perier (5): ASoC: es8328: Let device auto detect ratios in slave mode ASoC: es8328: Simplify rates definition ASoC: es8328: Enabling support for 192k ASoC: rockchip: Enable 192khz in hw_params operation ASoC: rockchip: Enable 192khz in hw_params operation Ryan Lee (1): ASoC: Add support for Maxim Integrated MAX98927 Amplifier Ryder Lee (2): ASoC: mediatek: Add mt2701-wm8960 machine driver ASoC: mediatek: add bindings for wm8960 codec machine driver Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add loadable module support on KBL platform Stefan Agner (1): ASoC: fsl_asrc_dma: use correct direction enum type Subhransu S. Prusty (3): ASoC: Intel: Skylake: Commonize library load ASoC: Intel: Skylake: Register dsp_fw_ops for kabylake ASoC: Intel: Skylake: Modify load_lib_ipc arguments for a nowait version Sylwester Nawrocki (2): ASoC: Add Odroid sound DT bindings documentation ASoC: samsung: Add Odroid ASoC machine driver Takashi Iwai (6): ASoC: intel: Don't print FW version repeatedly ASoC: Provide a dummy wrapper of snd_soc_set_dmi_name() ASoC: Call snd_soc_set_dmi_name() unconditionally ASoC: bytcr_rt5640: Allow quirk set via module option ASoC: intel: Fix PM and non-atomic crash in bytcr drivers ASoC: bytcr_rt5640: Fix a typo and quirk parameter type Takashi Sakamoto (2): ASoC: Intel: Skylake: use a helper macro to rounding-up calculation ASoC: Intel: Skylake: code cleanup for pin fixup limitation Vinod Koul (4): ASoC: Intel: Skylake: Don't unload module when in use ASoC: Intel: Skylake: Remove redundant vmixer handler ASoC: Intel: Skylake: remove hard coded ACPI path ASoC: Intel: Skylake: Fix unused variable warning Wolfram Sang (2): ASoC: rsnd: drop useles self-assignments ASoC: rsnd: check return value of init function kbuild test robot (2): ASoC: cs35l35: fix semicolon.cocci warnings ASoC: rsnd: fix semicolon.cocci warnings olivier moysan (2): ASoC: stm32: add bindings for SAI ASoC: stm32: add SAI driver .../devicetree/bindings/sound/cs35l35.txt | 180 ++ .../devicetree/bindings/sound/dioo,dio2125.txt | 12 + .../devicetree/bindings/sound/everest,es7134.txt | 10 + .../devicetree/bindings/sound/fsl,ssi.txt | 34 +- .../bindings/sound/hisilicon,hi6210-i2s.txt | 42 + .../devicetree/bindings/sound/max98925.txt | 22 - .../devicetree/bindings/sound/max98926.txt | 32 - .../devicetree/bindings/sound/max9892x.txt | 41 + .../devicetree/bindings/sound/mt2701-wm8960.txt | 24 + .../devicetree/bindings/sound/nau8824.txt | 88 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + .../devicetree/bindings/sound/samsung,odroid.txt | 57 + .../devicetree/bindings/sound/sgtl5000.txt | 9 + .../devicetree/bindings/sound/st,stm32-sai.txt | 89 + .../devicetree/bindings/sound/tas2552.txt | 10 +- Documentation/devicetree/bindings/sound/wm8903.txt | 13 + .../devicetree/bindings/sound/zte,tdm.txt | 30 + include/sound/cs35l35.h | 108 ++ include/sound/soc.h | 14 +- sound/hda/hdac_controller.c | 2 +- sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/blackfin/bfin-eval-adau1373.c | 2 +- sound/soc/blackfin/bfin-eval-adav80x.c | 2 +- sound/soc/codecs/Kconfig | 30 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ak4613.c | 10 +- sound/soc/codecs/cs35l35.c | 1580 +++++++++++++++++ sound/soc/codecs/cs35l35.h | 294 ++++ sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs53l30.c | 1 + sound/soc/codecs/da7213.c | 13 + sound/soc/codecs/dio2125.c | 120 ++ sound/soc/codecs/es7134.c | 116 ++ sound/soc/codecs/es8328.c | 51 +- sound/soc/codecs/hdac_hdmi.c | 4 +- sound/soc/codecs/max9867.c | 4 +- sound/soc/codecs/max98927.c | 841 +++++++++ sound/soc/codecs/max98927.h | 272 +++ sound/soc/codecs/nau8540.c | 1224 ++++++------- sound/soc/codecs/nau8540.h | 310 ++-- sound/soc/codecs/nau8824.c | 1831 ++++++++++++++++++++ sound/soc/codecs/nau8824.h | 466 +++++ sound/soc/codecs/rt5514.c | 36 +- sound/soc/codecs/rt5645.c | 10 + sound/soc/codecs/rt5665.c | 222 ++- sound/soc/codecs/rt5665.h | 2 - sound/soc/codecs/rt5670.c | 21 + sound/soc/codecs/rt5677.c | 7 + sound/soc/codecs/sgtl5000.c | 19 +- sound/soc/codecs/ssm4567.c | 9 + sound/soc/codecs/sta529.c | 7 + sound/soc/codecs/tas2552.c | 6 +- sound/soc/codecs/tlv320aic23.c | 7 +- sound/soc/codecs/twl6040.c | 8 + sound/soc/codecs/uda1380.c | 7 + sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8903.c | 31 + sound/soc/codecs/wm8960.c | 195 ++- sound/soc/codecs/wm8978.c | 7 + sound/soc/codecs/wm_adsp.c | 324 +++- sound/soc/codecs/wm_adsp.h | 24 + sound/soc/dwc/Kconfig | 4 +- sound/soc/dwc/Makefile | 6 +- sound/soc/dwc/{designware_i2s.c => dwc-i2s.c} | 0 sound/soc/dwc/{designware_pcm.c => dwc-pcm.c} | 3 - sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_esai.c | 5 +- sound/soc/fsl/fsl_ssi.c | 27 +- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 28 - sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/fsl/imx-wm8962.c | 72 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/generic/simple-card.c | 43 +- sound/soc/generic/simple-scu-card.c | 37 +- sound/soc/hisilicon/Kconfig | 5 + sound/soc/hisilicon/Makefile | 1 + sound/soc/hisilicon/hi6210-i2s.c | 618 +++++++ sound/soc/hisilicon/hi6210-i2s.h | 276 +++ sound/soc/intel/Kconfig | 24 + sound/soc/intel/atom/sst/sst_acpi.c | 41 +- sound/soc/intel/atom/sst/sst_ipc.c | 4 +- sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bdw-rt5677.c | 5 +- sound/soc/intel/boards/broadwell.c | 3 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 97 +- sound/soc/intel/boards/bxt_rt298.c | 3 + sound/soc/intel/boards/bytcht_da7213.c | 283 +++ sound/soc/intel/boards/bytcht_nocodec.c | 208 +++ sound/soc/intel/boards/bytcr_rt5640.c | 113 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 - sound/soc/intel/haswell/sst-haswell-ipc.c | 6 - sound/soc/intel/skylake/bxt-sst.c | 118 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-nhlt.c | 7 +- sound/soc/intel/skylake/skl-pcm.c | 118 +- sound/soc/intel/skylake/skl-sst-cldma.c | 26 +- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 6 +- sound/soc/intel/skylake/skl-sst-dsp.h | 40 +- sound/soc/intel/skylake/skl-sst-ipc.c | 76 +- sound/soc/intel/skylake/skl-sst-ipc.h | 17 +- sound/soc/intel/skylake/skl-sst-utils.c | 140 +- sound/soc/intel/skylake/skl-sst.c | 175 +- sound/soc/intel/skylake/skl-topology.c | 247 ++- sound/soc/intel/skylake/skl-topology.h | 17 +- sound/soc/intel/skylake/skl.c | 2 +- sound/soc/intel/skylake/skl.h | 1 + sound/soc/mediatek/Kconfig | 10 + sound/soc/mediatek/mt2701/Makefile | 1 + sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 16 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 176 ++ sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 2 +- sound/soc/omap/omap-twl4030.c | 2 +- sound/soc/omap/omap3pandora.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 7 +- sound/soc/pxa/brownstone.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/e750_wm9705.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/hx4700.c | 2 +- sound/soc/pxa/imote2.c | 2 +- sound/soc/pxa/magician.c | 4 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/pxa/mmp-pcm.c | 1 - sound/soc/pxa/mmp-sspa.c | 1 - sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/pxa-ssp.c | 15 +- sound/soc/pxa/pxa2xx-ac97.c | 5 +- sound/soc/pxa/pxa2xx-i2s.c | 8 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/raumfeld.c | 8 +- sound/soc/pxa/spitz.c | 6 +- sound/soc/pxa/tosa.c | 4 +- sound/soc/pxa/z2.c | 6 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 12 + sound/soc/qcom/lpass-cpu.c | 22 +- sound/soc/qcom/lpass-ipq806x.c | 6 + sound/soc/qcom/lpass.h | 2 + sound/soc/rockchip/rk3288_hdmi_analog.c | 3 + sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/bells.c | 1 - sound/soc/samsung/i2s-regs.h | 2 - sound/soc/samsung/i2s.c | 1 - sound/soc/samsung/odroid.c | 219 +++ sound/soc/samsung/s3c-i2s-v2.c | 1 - sound/soc/sh/rcar/adg.c | 75 +- sound/soc/sh/rcar/core.c | 111 +- sound/soc/sh/rcar/dvc.c | 24 +- sound/soc/sh/rcar/rsnd.h | 55 +- sound/soc/sh/rcar/src.c | 3 +- sound/soc/sh/rcar/ssi.c | 9 + sound/soc/sirf/sirf-audio-port.c | 1 + sound/soc/sirf/sirf-audio.c | 1 + sound/soc/sirf/sirf-usp.c | 3 + sound/soc/soc-core.c | 27 +- sound/soc/soc-jack.c | 48 + sound/soc/soc-topology.c | 6 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 37 +- sound/soc/sti/uniperif_reader.c | 24 +- sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 6 + sound/soc/stm/stm32_sai.c | 115 ++ sound/soc/stm/stm32_sai.h | 200 +++ sound/soc/stm/stm32_sai_sub.c | 884 ++++++++++ sound/soc/sunxi/sun8i-codec-analog.c | 168 +- sound/soc/sunxi/sun8i-codec.c | 10 +- sound/soc/tegra/tegra20_ac97.c | 1 - sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 1 - sound/soc/tegra/tegra20_spdif.c | 5 +- sound/soc/tegra/tegra30_ahub.c | 5 +- sound/soc/tegra/tegra30_i2s.c | 1 - sound/soc/tegra/tegra_alc5632.c | 4 +- sound/soc/tegra/tegra_max98090.c | 4 +- sound/soc/tegra/tegra_rt5640.c | 4 +- sound/soc/tegra/tegra_sgtl5000.c | 4 +- sound/soc/tegra/tegra_wm8753.c | 4 +- sound/soc/tegra/tegra_wm8903.c | 4 +- sound/soc/tegra/tegra_wm9712.c | 4 +- sound/soc/tegra/trimslice.c | 4 +- sound/soc/txx9/txx9aclc.c | 5 +- sound/soc/ux500/mop500.c | 4 - sound/soc/ux500/ux500_msp_dai.c | 4 +- sound/soc/ux500/ux500_msp_i2s.c | 1 - sound/soc/zte/Kconfig | 8 + sound/soc/zte/Makefile | 1 + sound/soc/zte/zx-tdm.c | 461 +++++ 207 files changed, 12860 insertions(+), 1973 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs35l35.txt create mode 100644 Documentation/devicetree/bindings/sound/dioo,dio2125.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es7134.txt create mode 100644 Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98925.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-wm8960.txt create mode 100644 Documentation/devicetree/bindings/sound/nau8824.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt create mode 100644 include/sound/cs35l35.h create mode 100644 sound/soc/codecs/cs35l35.c create mode 100644 sound/soc/codecs/cs35l35.h create mode 100644 sound/soc/codecs/dio2125.c create mode 100644 sound/soc/codecs/es7134.c create mode 100644 sound/soc/codecs/max98927.c create mode 100644 sound/soc/codecs/max98927.h create mode 100644 sound/soc/codecs/nau8824.c create mode 100644 sound/soc/codecs/nau8824.h rename sound/soc/dwc/{designware_i2s.c => dwc-i2s.c} (100%) rename sound/soc/dwc/{designware_pcm.c => dwc-pcm.c} (98%) create mode 100644 sound/soc/hisilicon/Kconfig create mode 100644 sound/soc/hisilicon/Makefile create mode 100644 sound/soc/hisilicon/hi6210-i2s.c create mode 100644 sound/soc/hisilicon/hi6210-i2s.h create mode 100644 sound/soc/intel/boards/bytcht_da7213.c create mode 100644 sound/soc/intel/boards/bytcht_nocodec.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-wm8960.c create mode 100644 sound/soc/samsung/odroid.c create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_sai.c create mode 100644 sound/soc/stm/stm32_sai.h create mode 100644 sound/soc/stm/stm32_sai_sub.c create mode 100644 sound/soc/zte/zx-tdm.c

2 1

[alsa-devel] [PATCH v2 0/2] Relax bitclk computation when using PLL
by Daniel Baluta 30 Apr '17

30 Apr '17

Using strict bitclk requirements we cannot support all promised rates and formats. For this reason we relax bitclk computation by choosing the best available bitclk. First patch in the series is based on Arnd's patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119899.html Second one does the actual bitclk relaxation. Daniel Baluta (2): ASoC: codec: wm9860: avoid maybe-uninitialized warning ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 36 +++++++++++++++++++++--------------- 1 file changed, 21 insertions(+), 15 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: dwc: Disallow building designware_pcm as a module
by Jose Abreu 30 Apr '17

30 Apr '17

Designware PCM is an extension to Designware I2S and they are dependent on each other. For this reason, make Designware PCM a boolean which will compile with Desigwnare I2S module. The name of the module is not changed but the name of the files need to be changed. Also, without this commit we get errors when probbing designware_i2s module because of unspecified license: designware_pcm: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint designware_pcm: Unknown symbol __rcu_read_lock (err 0) designware_pcm: Unknown symbol devm_snd_soc_register_platform (err 0) designware_pcm: Unknown symbol synchronize_rcu (err 0) designware_pcm: Unknown symbol __rcu_read_unlock (err 0) designware_pcm: Unknown symbol snd_soc_set_runtime_hwparams (err 0) So, this is really needed as a fix. Fixes: 79361b2b98b7 ("ASoC: dwc: Add PIO PCM extension") Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> Signed-off-by: Jose Abreu <joabreu(a)synopsys.com> Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Lubomir Rintel <lkundrak(a)v3.sk> --- Changes v1->v2: - Refrased commit message - Removed EXPORT_SYMBOL_GPL as this isn't needed anymore - Added my self (Jose Abreu) as commiter - Changed name of files to dwc-i2s and dwc-pcm sound/soc/dwc/Kconfig | 4 ++-- sound/soc/dwc/Makefile | 6 +++--- sound/soc/dwc/{designware_i2s.c => dwc-i2s.c} | 0 sound/soc/dwc/{designware_pcm.c => dwc-pcm.c} | 3 --- 4 files changed, 5 insertions(+), 8 deletions(-) rename sound/soc/dwc/{designware_i2s.c => dwc-i2s.c} (100%) rename sound/soc/dwc/{designware_pcm.c => dwc-pcm.c} (98%) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index c297efe..c6fd95f 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -8,10 +8,10 @@ config SND_DESIGNWARE_I2S maximum of 8 channels each for play and record. config SND_DESIGNWARE_PCM - tristate "PCM PIO extension for I2S driver" + bool "PCM PIO extension for I2S driver" depends on SND_DESIGNWARE_I2S help - Say Y, M or N if you want to add a custom ALSA extension that registers + Say Y or N if you want to add a custom ALSA extension that registers a PCM and uses PIO to transfer data. This functionality is specially suited for I2S devices that don't have diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile index 38f1ca3..3e24c0f 100644 --- a/sound/soc/dwc/Makefile +++ b/sound/soc/dwc/Makefile @@ -1,5 +1,5 @@ # SYNOPSYS Platform Support obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o -ifdef CONFIG_SND_DESIGNWARE_PCM -obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_pcm.o -endif + +designware_i2s-y := dwc-i2s.o +designware_i2s-$(CONFIG_SND_DESIGNWARE_PCM) += dwc-pcm.o diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/dwc-i2s.c similarity index 100% rename from sound/soc/dwc/designware_i2s.c rename to sound/soc/dwc/dwc-i2s.c diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/dwc-pcm.c similarity index 98% rename from sound/soc/dwc/designware_pcm.c rename to sound/soc/dwc/dwc-pcm.c index 459ec86..406fd867 100644 --- a/sound/soc/dwc/designware_pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -129,13 +129,11 @@ void dw_pcm_push_tx(struct dw_i2s_dev *dev) { dw_pcm_transfer(dev, true); } -EXPORT_SYMBOL_GPL(dw_pcm_push_tx); void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { dw_pcm_transfer(dev, false); } -EXPORT_SYMBOL_GPL(dw_pcm_pop_rx); static int dw_pcm_open(struct snd_pcm_substream *substream) { @@ -281,4 +279,3 @@ int dw_pcm_register(struct platform_device *pdev) { return devm_snd_soc_register_platform(&pdev->dev, &dw_pcm_platform); } -EXPORT_SYMBOL_GPL(dw_pcm_register); -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: cs4271: configure reset GPIO as output" to the asoc tree
by Mark Brown 30 Apr '17

30 Apr '17

The patch ASoC: cs4271: configure reset GPIO as output 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 49b2e27ab9f66b0a22c21980ad8118a4038324ae Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> Date: Sat, 29 Apr 2017 12:19:33 +0200 Subject: [PATCH] ASoC: cs4271: configure reset GPIO as output During reset "refactoring" the output configuration was lost. This commit repairs sound on EDB93XX boards. Fixes: 9a397f4 ("ASoC: cs4271: add regulator consumer support") Signed-off-by: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: <stable(a)vger.kernel.org> # 4.6+ --- sound/soc/codecs/cs4271.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 8c0f3b89b5bc..e78b5f055f25 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -498,7 +498,7 @@ static int cs4271_reset(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); if (gpio_is_valid(cs4271->gpio_nreset)) { - gpio_set_value(cs4271->gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); mdelay(1); gpio_set_value(cs4271->gpio_nreset, 1); mdelay(1); -- 2.11.0

1 0

[alsa-devel] [bug report] ASoC: sti: Add CPU DAI driver for playback
by Dan Carpenter 30 Apr '17

30 Apr '17

Hello Arnaud Pouliquen, The patch 76c2145ded6b: "ASoC: sti: Add CPU DAI driver for playback" from Jun 22, 2015, leads to the following static checker warning: sound/soc/sti/uniperif_player.c:1077 uni_player_init() warn: unused return: ret = PTR_ERR() sound/soc/sti/uniperif_player.c 1047 int uni_player_init(struct platform_device *pdev, 1048 struct uniperif *player) 1049 { 1050 int ret = 0; 1051 1052 player->dev = &pdev->dev; 1053 player->state = UNIPERIF_STATE_STOPPED; 1054 player->dai_ops = &uni_player_dai_ops; 1055 1056 /* Get PCM_CLK_SEL & PCMP_VALID_SEL from audio-glue-ctrl SoC reg */ 1057 ret = uni_player_parse_dt_audio_glue(pdev, player); 1058 1059 if (ret < 0) { 1060 dev_err(player->dev, "Failed to parse DeviceTree\n"); 1061 return ret; 1062 } 1063 1064 /* Underflow recovery is only supported on later ip revisions */ 1065 if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) 1066 player->underflow_enabled = 1; 1067 1068 if (UNIPERIF_TYPE_IS_TDM(player)) 1069 player->hw = &uni_tdm_hw; 1070 else 1071 player->hw = &uni_player_pcm_hw; 1072 1073 /* Get uniperif resource */ 1074 player->clk = of_clk_get(pdev->dev.of_node, 0); 1075 if (IS_ERR(player->clk)) { 1076 dev_err(player->dev, "Failed to get clock\n"); 1077 ret = PTR_ERR(player->clk); ^^^^^^^^^^^^^^^^^^^^^^^^^^ This should probably be "return PTR_ERR(player->clk);? Can we run without a clk? 1078 } 1079 1080 /* Select the frequency synthesizer clock */ 1081 if (player->clk_sel) { 1082 ret = regmap_field_write(player->clk_sel, 1); 1083 if (ret) { 1084 dev_err(player->dev, 1085 "%s: Failed to select freq synth clock\n", 1086 __func__); 1087 return ret; 1088 } 1089 } regards, dan carpenter

3 4

[alsa-devel] [PATCH 1/3] ASoC: Intel: Skylake: Fix uninitialized pointer usage
by Vinod Koul 30 Apr '17

30 Apr '17

Due to recent code move, one instance dev_err() was passed a wrong uninitialized pointer for the device, so fix it. sound/soc/intel/skylake/bxt-sst.c: In function 'bxt_sst_dsp_init': sound/soc/intel/skylake/bxt-sst.c:567:14: warning: 'skl' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(skl->dev, "%s: no device\n", __func__); ^ Fixes: 9fe9c7119283 ("ASoC: Intel: Skylake: Move sst common initialization to a helper function") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index fde4bc0f35b0..f5e7dbb1ba39 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -564,7 +564,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); if (ret < 0) { - dev_err(skl->dev, "%s: no device\n", __func__); + dev_err(dev, "%s: no device\n", __func__); return ret; } -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: nau8824: rename controls to match DAPM controls
by John Hsu 30 Apr '17

30 Apr '17

Rename the name of kcontrols to match up the DAPM widget controls. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 18552ed..cd358be0 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -362,27 +362,27 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("ADC Decimation Rate", nau8824_adc_decimation_enum), SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), - SOC_SINGLE_TLV("Speaker Right from DACR Volume", + SOC_SINGLE_TLV("Speaker Right DACR Volume", NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Speaker Left from DACL Volume", + SOC_SINGLE_TLV("Speaker Left DACL Volume", NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Speaker Left from DACR Volume", + SOC_SINGLE_TLV("Speaker Left DACR Volume", NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Speaker Right from DACL Volume", + SOC_SINGLE_TLV("Speaker Right DACL Volume", NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Headphone Right from DACR Volume", + SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Headphone Left from DACL Volume", + SOC_SINGLE_TLV("Headphone Left DACL Volume", NAU8824_REG_ATT_PORT0, 0, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Headphone Right from DACL Volume", + SOC_SINGLE_TLV("Headphone Right DACL Volume", NAU8824_REG_ATT_PORT1, 8, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Headphone Left from DACR Volume", + SOC_SINGLE_TLV("Headphone Left DACR Volume", NAU8824_REG_ATT_PORT1, 0, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Mic1 Volume", NAU8824_REG_FEPGA_II, + SOC_SINGLE_TLV("MIC1 Volume", NAU8824_REG_FEPGA_II, NAU8824_FEPGA_GAINL_SFT, 0x12, 0, mic_vol_tlv), - SOC_SINGLE_TLV("Mic2 Volume", NAU8824_REG_FEPGA_II, + SOC_SINGLE_TLV("MIC2 Volume", NAU8824_REG_FEPGA_II, NAU8824_FEPGA_GAINR_SFT, 0x12, 0, mic_vol_tlv), SOC_SINGLE_TLV("DMIC1 Volume", NAU8824_REG_ADC_CH0_DGAIN_CTRL, -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: nau8824: leave Class D gain at chip default
by John Hsu 30 Apr '17

30 Apr '17

Remove initial configuration of Class D gain for 1R and 2L. Leave them at the chip default. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cd358be0..cca974d 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1626,12 +1626,6 @@ static void nau8824_init_regs(struct nau8824 *nau8824) regmap_update_bits(regmap, NAU8824_REG_DAC_FILTER_CTRL_1, NAU8824_DAC_CICCLP_OFF | NAU8824_DAC_OVERSAMPLE_MASK, NAU8824_DAC_CICCLP_OFF | NAU8824_DAC_OVERSAMPLE_64); - /* Class D gain 9db for 1R and 2L */ - regmap_update_bits(regmap, NAU8824_REG_CLASSD_GAIN_1, - NAU8824_CLASSD_GAIN_1R_MASK, - (0xa << NAU8824_CLASSD_GAIN_1R_SFT)); - regmap_update_bits(regmap, NAU8824_REG_CLASSD_GAIN_2, - NAU8824_CLASSD_GAIN_2L_MASK, 0xa); /* DAC clock delay 2ns, VREF */ regmap_update_bits(regmap, NAU8824_REG_RDAC, NAU8824_RDAC_CLK_DELAY_MASK | NAU8824_RDAC_VREF_MASK, -- 2.6.4

2 1

[alsa-devel] [PATCH] ALSA: ali5451: fix spelling mistake in "ali_capture_preapre"
by Colin King 30 Apr '17

30 Apr '17

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_warn message, "ali_capture_preapre" should be "ali_capture_prepare" Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ali5451/ali5451.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 92b819e4f729..34bbc2e730a6 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1339,7 +1339,7 @@ static int snd_ali_prepare(struct snd_pcm_substream *substream) rate = snd_ali_get_spdif_in_rate(codec); if (rate == 0) { dev_warn(codec->card->dev, - "ali_capture_preapre: spdif rate detect err!\n"); + "ali_capture_prepare: spdif rate detect err!\n"); rate = 48000; } spin_lock_irq(&codec->reg_lock); -- 2.11.0

2 1

[alsa-devel] [PATCH v7 0/9][resend] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 28 Apr '17

28 Apr '17

Hi Mark, Rob These are resend of v7 of OF-graph sound card support. I posted because Japan will have long holiday from tomorrow 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): of-graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 73 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 600 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

2 11

[alsa-devel] [PATCH 0/3] ASoC: simple-card: Fine-tuning for three function implementations
by SF Markus Elfring 28 Apr '17

28 Apr '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 26 Apr 2017 14:54:32 +0200 Three update suggestions were taken into account from static source code analysis. Markus Elfring (3): Use devm_kcalloc() in asoc_simple_card_parse_aux_devs() Use devm_kcalloc() in asoc_simple_card_probe() SCU: Use devm_kcalloc() in asoc_simple_card_probe() sound/soc/generic/simple-card.c | 8 ++++---- sound/soc/generic/simple-scu-card.c | 5 ++--- 2 files changed, 6 insertions(+), 7 deletions(-) -- 2.12.2

2 5

[alsa-devel] [PATCH -next] ASoC: Intel: Skylake: Fix to use list_for_each_safe() when delete items
by Wei Yongjun 28 Apr '17

28 Apr '17

From: Wei Yongjun <weiyongjun1(a)huawei.com> Since we will remove items off the list using list_del() we need to use a safe version of the list_for_each() macro aptly named list_for_each_safe(). This is detected by Coccinelle semantic patch. Fixes: b8c722ddd548 ("ASoC: Intel: Skylake: Add support for deferred DSP module bind") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/intel/skylake/skl-pcm.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 4 ++-- 2 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 600faad..d43d197 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1323,10 +1323,10 @@ int skl_platform_unregister(struct device *dev) { struct hdac_ext_bus *ebus = dev_get_drvdata(dev); struct skl *skl = ebus_to_skl(ebus); - struct skl_module_deferred_bind *modules; + struct skl_module_deferred_bind *modules, *tmp; if (!list_empty(&skl->bind_list)) { - list_for_each_entry(modules, &skl->bind_list, node) { + list_for_each_entry_safe(modules, tmp, &skl->bind_list, node) { list_del(&modules->node); kfree(modules); } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 74b3acf..aea7917 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1091,7 +1091,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, struct skl_module_cfg *src_module = NULL, *dst_module; struct skl_sst *ctx = skl->skl_sst; struct skl_pipe *s_pipe = mconfig->pipe; - struct skl_module_deferred_bind *modules; + struct skl_module_deferred_bind *modules, *tmp; if (s_pipe->state == SKL_PIPE_INVALID) return -EINVAL; @@ -1105,7 +1105,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, src_module = w_module->w->priv; - list_for_each_entry(modules, &skl->bind_list, node) { + list_for_each_entry_safe(modules, tmp, &skl->bind_list, node) { /* * When the destination module is deleted, Unbind the * modules from deferred bind list.

3 5

[alsa-devel] [PATCH] ASoC: Intel: Skylake: use dev in dev_err rather than skl pointer
by Colin King 28 Apr '17

28 Apr '17

From: Colin Ian King <colin.king(a)canonical.com> The dev_err message is dereferencing an uininitialized skl pointer which should be avoided. Don't use skl, use dev instead. Detected by CoverityScan, CID#1432042 ("Uninitialized pointer read") Fixes: 9fe9c71192832 ("ASoC: Intel: Skylake: Move sst common initialization to a helper function") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/bxt-sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index fde4bc0f35b0..f5e7dbb1ba39 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -564,7 +564,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); if (ret < 0) { - dev_err(skl->dev, "%s: no device\n", __func__); + dev_err(dev, "%s: no device\n", __func__); return ret; } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: uninitialized variable in error message
by Dan Carpenter 28 Apr '17

28 Apr '17

"skl" hasn't been initialized yet, so we can't dereference it. Fixes: 9fe9c7119283 ("ASoC: Intel: Skylake: Move sst common initialization to a helper function") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index fde4bc0f35b0..f5e7dbb1ba39 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -564,7 +564,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); if (ret < 0) { - dev_err(skl->dev, "%s: no device\n", __func__); + dev_err(dev, "%s: no device\n", __func__); return ret; }

2 1

[alsa-devel] [PATCH] ASoC: dwc: disallow building designware_pcm as a module
by Lubomir Rintel 27 Apr '17

27 Apr '17

It makes not sense: the whether the PIO PCM extension is used is hardcoded to the designware_i2s driver and designware_pcm doesn't have any module metadata, causing a kernel taint: [ 44.287000] designware_pcm: module license 'unspecified' taints kernel. Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> --- sound/soc/dwc/Kconfig | 4 ++-- sound/soc/dwc/Makefile | 6 +++++- sound/soc/dwc/{designware_i2s.c => designware_i2s_main.c} | 2 +- sound/soc/dwc/{designware_pcm.c => designware_i2s_pcm.c} | 2 +- 4 files changed, 9 insertions(+), 5 deletions(-) rename sound/soc/dwc/{designware_i2s.c => designware_i2s_main.c} (99%) rename sound/soc/dwc/{designware_pcm.c => designware_i2s_pcm.c} (99%) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index c297efe..704b7b2 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -8,10 +8,10 @@ config SND_DESIGNWARE_I2S maximum of 8 channels each for play and record. config SND_DESIGNWARE_PCM - tristate "PCM PIO extension for I2S driver" + bool "PCM PIO extension for I2S driver" depends on SND_DESIGNWARE_I2S help - Say Y, M or N if you want to add a custom ALSA extension that registers + Say Y if you want to add a custom ALSA extension that registers a PCM and uses PIO to transfer data. This functionality is specially suited for I2S devices that don't have diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile index 38f1ca3..05c594f 100644 --- a/sound/soc/dwc/Makefile +++ b/sound/soc/dwc/Makefile @@ -1,5 +1,9 @@ # SYNOPSYS Platform Support + obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o + +designware_i2s-objs := designware_i2s_main.o + ifdef CONFIG_SND_DESIGNWARE_PCM -obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_pcm.o +designware_i2s-objs += designware_i2s_pcm.o endif diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s_main.c similarity index 99% rename from sound/soc/dwc/designware_i2s.c rename to sound/soc/dwc/designware_i2s_main.c index bdf8398..23c2212 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s_main.c @@ -1,7 +1,7 @@ /* * ALSA SoC Synopsys I2S Audio Layer * - * sound/soc/dwc/designware_i2s.c + * sound/soc/dwc/designware_i2s_main.c * * Copyright (C) 2010 ST Microelectronics * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/designware_i2s_pcm.c similarity index 99% rename from sound/soc/dwc/designware_pcm.c rename to sound/soc/dwc/designware_i2s_pcm.c index 4a83a22..27664e2 100644 --- a/sound/soc/dwc/designware_pcm.c +++ b/sound/soc/dwc/designware_i2s_pcm.c @@ -1,7 +1,7 @@ /* * ALSA SoC Synopsys PIO PCM for I2S driver * - * sound/soc/dwc/designware_pcm.c + * sound/soc/dwc/designware_i2s_pcm.c * * Copyright (C) 2016 Synopsys * Jose Abreu <joabreu(a)synopsys.com> -- 2.9.3

4 13

[alsa-devel] [PATCH 0/2] ASoC: stm32: Add I2S driver
by olivier moysan 27 Apr '17

27 Apr '17

This patch-set handles the SPI/I2S IP on STM32 platforms. It applies to STM32 platforms implementing version 2 of SPI/I2S IP like STM32H7 series. The SPI/I2S block is a serial peripheral interface (SPI), which can also be configured to work on I2S/PCM mode. ASoC STM32 I2S driver only supports this I2S/PCM mode. The I2S/PCM interface can either support full duplex or half-duplex communication (transmitter or receiver only). I2S driver exhibits 3 DAIs corresponding to a playback, capture or full duplex PCM device. When the interface is configured as full duplex, the related PCM device must be opened twice. The PCM streams are started when both playback and capture devices are opened. The relevant DAI can be selected through Device Tree using sound-dai-cells parameter. The I2S/PCM interface supports four audio standards: I2S Philips standard, MSB justified standard, LSB justified standard, PCM standard PCM standard is declined in two versions, PCM short and long, according frame synchronization duration. These standards can be mapped to ASoC standards as follows: - I2S: i2s - MSB justified standard: left_j - LSB justified standard: right_j - PCM short: dsp_a PCM long format is not exposed as it does not match supported ASoC standards. olivier moysan (2): dt-bindings: Document STM32 I2S bindings ASoC: stm32: Add I2S driver .../devicetree/bindings/sound/st,stm32h7-i2s.txt | 71 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 2 + sound/soc/stm/stm32_i2s.c | 1069 ++++++++++++++++++++ 6 files changed, 1152 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32h7-i2s.txt create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_i2s.c -- 1.9.1

3 15

[alsa-devel] [PATCH v1 1/8] lib/uuid: Introduce uuid_{be|le}_cmp_p{p}() helpers
by Andy Shevchenko 27 Apr '17

27 Apr '17

New helpers take pointers to uuid_{be|le} as parameters. When using them on a raw data we don't need to do an ugly dereference and, in some cases, a type casting. Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Srinivas Pandruvada <srinivas.pandruvada(a)linux.intel.com> Cc: Benjamin Tissoires <benjamin.tissoires(a)redhat.com> Cc: Kirti Wankhede <kwankhede(a)nvidia.com> Cc: Alex Williamson <alex.williamson(a)redhat.com> Cc: "K. Y. Srinivasan" <kys(a)microsoft.com> Cc: Haiyang Zhang <haiyangz(a)microsoft.com> Cc: Stephen Hemminger <sthemmin(a)microsoft.com> Cc: Tomas Winkler <tomas.winkler(a)intel.com> Cc: Matt Fleming <matt(a)codeblueprint.co.uk> Cc: Ard Biesheuvel <ard.biesheuvel(a)linaro.org> Cc: "Rafael J. Wysocki" <rjw(a)rjwysocki.net> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- include/linux/uuid.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/include/linux/uuid.h b/include/linux/uuid.h index 4dff73a89758..45312cb5ac65 100644 --- a/include/linux/uuid.h +++ b/include/linux/uuid.h @@ -58,6 +58,26 @@ static inline int uuid_be_cmp(const uuid_be u1, const uuid_be u2) return memcmp(&u1, &u2, sizeof(uuid_be)); } +static inline int uuid_le_cmp_p(const uuid_le *pu1, const uuid_le u2) +{ + return memcmp(pu1, &u2, sizeof(uuid_le)); +} + +static inline int uuid_be_cmp_p(const uuid_be *pu1, const uuid_be u2) +{ + return memcmp(pu1, &u2, sizeof(uuid_be)); +} + +static inline int uuid_le_cmp_pp(const uuid_le *pu1, const uuid_le *pu2) +{ + return memcmp(pu1, pu2, sizeof(uuid_le)); +} + +static inline int uuid_be_cmp_pp(const uuid_be *pu1, const uuid_be *pu2) +{ + return memcmp(pu1, pu2, sizeof(uuid_be)); +} + void generate_random_uuid(unsigned char uuid[16]); extern void uuid_le_gen(uuid_le *u); -- 2.11.0

7 18

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix uninitialized pointer use
by Arnd Bergmann 27 Apr '17

27 Apr '17

The error handling in bxt_sst_dsp_init() got changed in a way that it now derefences an uninitialized pointer when printing a warning about the device not being found: sound/soc/intel/skylake/bxt-sst.c: In function 'bxt_sst_dsp_init': sound/soc/intel/skylake/bxt-sst.c:567:14: error: 'skl' may be used uninitialized in this function [-Werror=maybe-uninitialized] As we do have a valid device pointer available at the call site, let's use that instead. Fixes: 9fe9c7119283 ("ASoC: Intel: Skylake: Move sst common initialization to a helper function") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/skylake/bxt-sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index fde4bc0f35b0..f5e7dbb1ba39 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -564,7 +564,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); if (ret < 0) { - dev_err(skl->dev, "%s: no device\n", __func__); + dev_err(dev, "%s: no device\n", __func__); return ret; } -- 2.9.0

3 2

[alsa-devel] FW: Further goals of thread-safe PCM API
by Wischer, Timo (ADITG/SW1) 27 Apr '17

27 Apr '17

Hi everyone, I am wondering about the implementation of the new thread-safety feature [1]. There are so many issues with deadlocks (e.g. [3]) which were already solved but also which are not yet solved. Why do you not using a recursive mutex to avoid most of this deadlocks? Using PTHREAD_MUTEX_RECURSIVE as the pthread attribute [2]. What is the preferred solution for external ALSA plugins which are using the external IO plugin SDK to prevent such deadlocks? For example I have one plugin which calls snd_pcm_avail() in the ext IO plug start callback. This ends up in a deadlock. #0 0x0000003f44e0feec in __lll_lock_wait () from /lib64/libpthread.so.0 #1 0x0000003f44e09b15 in pthread_mutex_lock () from /lib64/libpthread.so.0 #2 0x00007ffff7d3c8b4 in snd_pcm_lock (pcm=0x623150) at ../../../alsa-lib-1.1.2/src/pcm/pcm_local.h:1101 #3 snd_pcm_avail (pcm=0x623150) at ../../../alsa-lib-1.1.2/src/pcm/pcm.c:2762 #4 0x00007ffff7ae1e03 in ?? () from /usr/lib64/alsa-lib/libasound_module_pcm_loop.so #5 0x00007ffff7d7df85 in snd_pcm_ioplug_start (pcm=0x623150) at ../../../alsa-lib-1.1.2/src/pcm/pcm_ioplug.c:458 #6 0x00007ffff7d41065 in __snd_pcm_start (pcm=0x623150, pcm=0x623150) at ../../../alsa-lib-1.1.2/src/pcm/pcm_local.h:427 #7 snd1_pcm_write_areas (pcm=pcm@entry=0x623150, areas=areas@entry=0x7fffffffe750, offset=offset@entry=0, size=size@entry=64, func=func@entry=0x7ffff7d7de00 <ioplug_priv_transfer_areas>) at ../../../alsa-lib-1.1.2/src/pcm/pcm.c:7248 #8 0x00007ffff7d7eb65 in snd_pcm_ioplug_writei (pcm=0x623150, buffer=<optimized out>, size=64) at ../../../alsa-lib-1.1.2/src/pcm/pcm_ioplug.c:579 #9 0x000000000040601a in pcm_write ( data=0x623350 "\006\003\271\375N\004\063\376j\002n\374)\005\331\376I\005Z\376w\001\n\374\025\003\373\376{\003i\377T\001~\375}\375\244\372\267\002c\376%\004\352\377\312\b[\005\261\001\216\376\b\001,\377\352", count=count@entry=64) at ../../alsa-utils-1.1.2/aplay/aplay.c:2001 #10 0x00000000004062cb in playback_go (fd=6, loaded=<optimized out>, loaded@entry=0, count=147256, rtype=rtype@entry=2, name=name@entry=0x7fffffffed9a "/opt/platform/unit_tests/audio8k16S.wav") at ../../alsa-utils-1.1.2/aplay/aplay.c:2750 #11 0x0000000000407417 in playback (name=0x7fffffffed9a "/opt/platform/unit_tests/audio8k16S.wav") at ../../alsa-utils-1.1.2/aplay/aplay.c:2811 #12 0x0000000000409778 in main (argc=8, argv=0x7fffffffeb18) at ../../alsa-utils-1.1.2/aplay/aplay.c:859 Therefore do I have to replace all snd_pcm_* function calls in the external plugin with the non-thread-safe variants __snd_pcm_*? Or is there a better solution available? >From my point of view the usage of a recursive mutex would be the best and most stable solution. [1] http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=54931e5a5455ac681a32… [2] https://linux.die.net/man/3/pthread_mutexattr_settype [3] http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=24e63b75275e9c92… [4] http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___i_o_plug.html… Best regards Timo Wischer Advanced Driver Information Technology GmbH Software Group I (ADITG/SW1) Robert-Bosch-Str. 200 31139 Hildesheim Germany twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschaeftsfuehrung: Wilhelm Grabow, Ken Yaguchi

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[alsa-devel] [PATCH][resend] topology: Allow a data section to contain multiple tuples objects
by fuweix.tang＠intel.com 27 Apr '17

27 Apr '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> It's easy to use a vendor tuples object to define a C structure instance as vendor specific parameter for kernel drivers. And sometimes the kernel drivers may want a group of structures. So this patch will allow user to define multiple vendor tuples objects in a data section, to avoid defining multiple data sections and each data section only has 1 vendor tuples object. There is no ABI change in this patch. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Fuwei Tang <fuweix.tang(a)intel.com> --- include/topology.h | 14 ++++++++++ src/topology/data.c | 78 ++++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 74 insertions(+), 18 deletions(-) diff --git a/include/topology.h b/include/topology.h index 1bebe1e..ccb3a00 100644 --- a/include/topology.h +++ b/include/topology.h @@ -214,6 +214,20 @@ extern "C" { * The keyword tuples is to define vendor specific tuples. Please refer to * section Vendor Tokens and Vendor tuples. * + * It's easy to use a vendor tuples object to define a C structure instance. + * And a data section can include multiple vendor tuples objects: + * + * <pre> + * SectionData."data element name" { + * index "1" #Index number + * tuples [ + * "id of the 1st vendor tuples section" + * "id of the 2nd vendor tuples section" + * ... + * ] + * }; + * </pre> + * * <h5>How to define an element with private data</h5> * An element can refer to a single data section or multiple data * sections. diff --git a/src/topology/data.c b/src/topology/data.c index 00785ef..36060eb 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -408,7 +408,7 @@ static unsigned int get_tuple_size(int type) } } -/* fill a data element's private buffer with its tuples */ +/* Add a tuples object to the private buffer of its parent data element */ static int copy_tuples(struct tplg_elem *elem, struct tplg_vendor_tuples *tuples, struct tplg_vendor_tokens *tokens) { @@ -423,12 +423,9 @@ static int copy_tuples(struct tplg_elem *elem, unsigned int i, j; int token_val; - if (priv) { - SNDERR("error: %s has more data than tuples\n", elem->id); - return -EINVAL; - } + size = priv ? priv->size : 0; /* original private data size */ - size = 0; + /* scan each tuples set (one set per type) */ for (i = 0; i < tuples->num_sets ; i++) { tuple_set = tuples->set[i]; set_size = sizeof(struct snd_soc_tplg_vendor_array) @@ -448,7 +445,7 @@ static int copy_tuples(struct tplg_elem *elem, return -ENOMEM; off = priv->size; - priv->size = size; + priv->size = size; /* update private data size */ array = (struct snd_soc_tplg_vendor_array *)(priv->data + off); array->size = set_size; @@ -495,6 +492,7 @@ static int build_tuples(snd_tplg_t *tplg, struct tplg_elem *elem) struct tplg_ref *ref; struct list_head *base, *pos; struct tplg_elem *tuples, *tokens; + int err; base = &elem->ref_list; list_for_each(pos, base) { @@ -504,23 +502,27 @@ static int build_tuples(snd_tplg_t *tplg, struct tplg_elem *elem) if (!ref->id || ref->type != SND_TPLG_TYPE_TUPLE) continue; - tplg_dbg("look up tuples %s\n", ref->id); + tplg_dbg("tuples '%s' used by data '%s'\n", ref->id, elem->id); if (!ref->elem) ref->elem = tplg_elem_lookup(&tplg->tuple_list, ref->id, SND_TPLG_TYPE_TUPLE, elem->index); tuples = ref->elem; - if (!tuples) + if (!tuples) { + SNDERR("error: cannot find tuples %s\n", ref->id); return -EINVAL; + } - tplg_dbg("found tuples %s\n", tuples->id); tokens = get_tokens(tplg, tuples); - if (!tokens) + if (!tokens) { + SNDERR("error: cannot find token for %s\n", ref->id); return -EINVAL; + } - tplg_dbg("found tokens %s\n", tokens->id); - /* a data object can only have one tuples object */ - return copy_tuples(elem, tuples->tuples, tokens->tokens); + /* a data object can have multiple tuples objects */ + err = copy_tuples(elem, tuples->tuples, tokens->tokens); + if (err < 0) + return err; } return 0; @@ -689,6 +691,47 @@ static int parse_tuple_sets(snd_config_t *cfg, return 0; } +/* Parse tuples references for a data element, either a single tuples section + * or a list of tuples sections. + */ +static int parse_tuples_refs(snd_config_t *cfg, + struct tplg_elem *elem) +{ + snd_config_type_t type; + snd_config_iterator_t i, next; + snd_config_t *n; + const char *val = NULL; + + type = snd_config_get_type(cfg); + + /* refer to a single tuples section */ + if (type == SND_CONFIG_TYPE_STRING) { + if (snd_config_get_string(cfg, &val) < 0) + return -EINVAL; + tplg_dbg("\ttuples: %s\n", val); + return tplg_ref_add(elem, SND_TPLG_TYPE_TUPLE, val); + } + + if (type != SND_CONFIG_TYPE_COMPOUND) { + SNDERR("error: compound type expected for %s", elem->id); + return -EINVAL; + } + + /* refer to a list of data sections */ + snd_config_for_each(i, next, cfg) { + const char *val; + + n = snd_config_iterator_entry(i); + if (snd_config_get_string(n, &val) < 0) + continue; + + tplg_dbg("\ttuples: %s\n", val); + tplg_ref_add(elem, SND_TPLG_TYPE_TUPLE, val); + } + + return 0; +} + /* Parse private data references for the element, either a single data section * or a list of data sections. */ @@ -1002,10 +1045,9 @@ int tplg_parse_data(snd_tplg_t *tplg, snd_config_t *cfg, } if (strcmp(id, "tuples") == 0) { - if (snd_config_get_string(n, &val) < 0) - return -EINVAL; - tplg_dbg(" Data: %s\n", val); - tplg_ref_add(elem, SND_TPLG_TYPE_TUPLE, val); + err = parse_tuples_refs(n, elem); + if (err < 0) + return err; continue; } -- 2.7.4

2 1

[alsa-devel] [PATCH 00/11] drm/i915: LPE audio runtime PM and multipipe
by ville.syrjala＠linux.intel.com 26 Apr '17

26 Apr '17

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> I was wondering why my VLV no longer runtime suspended, and after some thinking I decided it had to be the LPE audio preventing it. Turns out I was right, so here's my attempt at fixing it. And while looking at the code I couldn't help but notice that it couldn't actually handle multiple pipes playing back audio at the same time. And even having multiple displays active even if only one was playing audio was probably a recipe for failure. So I tried to fix that by registering a separate PCM device for each pipe. Note that the patch subjects may not reflect the subsystem very well since most of these straddle the border between drm and alsa. I think I just slapped on drm/i915 to most where there was no clear winner. Entire series available here: git://github.com/vsyrjala/linux.git lpe_audio_multipipe Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Ville Syrjälä (11): drm/i915: Fix runtime PM for LPE audio ALSA: x86: Clear the pdata.notify_lpe_audio pointer before teardown drm/i915: Stop pretending to mask/unmask LPE audio interrupts drm/i915: Remove the unsued pending_notify from LPE platform data drm/i915: Replace tmds_clock_speed and link_rate with just ls_clock drm/i915: Remove hdmi_connected from LPE audio pdata drm/i915: Reorganize intel_lpe_audio_notify() arguments drm/i915: Clean up the LPE audio platform data ALSA: x86: Prepare LPE audio ctls for multiple PCMs ALSA: x86: Split snd_intelhad into card and PCM specific structures ALSA: x86: Register multiple PCM devices for the LPE audio card drivers/gpu/drm/i915/i915_drv.h | 4 +- drivers/gpu/drm/i915/i915_irq.c | 15 +- drivers/gpu/drm/i915/intel_audio.c | 19 +-- drivers/gpu/drm/i915/intel_lpe_audio.c | 95 ++++------- include/drm/intel_lpe_audio.h | 20 ++- sound/x86/intel_hdmi_audio.c | 288 +++++++++++++++++++-------------- sound/x86/intel_hdmi_audio.h | 17 +- 7 files changed, 230 insertions(+), 228 deletions(-) -- 2.10.2

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Re: [alsa-devel] Lenovo two codecs machine workaround
by Kailang 26 Apr '17

26 Apr '17

Attach again. C.c. to alsa-devel. This patch is dual codecs workarround code for Lenovo platform. > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Tuesday, April 25, 2017 4:40 PM > To: Kailang > Subject: Re: Lenovo two codecs machine workaround > > On Tue, 25 Apr 2017 10:35:24 +0200, > Kailang wrote: > > > > > > Attach is the support patch for Lenovo dual codecs. > > Looks good. Could you submit to alsa-devel ML? > > > thanks, > > Takashi > > ------Please consider the environment before printing this e-mail. >

2 1

[alsa-devel] [PATCH] ASoC: nau8824: new driver
by John Hsu 26 Apr '17

26 Apr '17

Add driver for NAU88L24. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- .../devicetree/bindings/sound/nau8824.txt | 88 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8824.c | 1837 ++++++++++++++++++++ sound/soc/codecs/nau8824.h | 466 +++++ 5 files changed, 2398 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8824.txt create mode 100644 sound/soc/codecs/nau8824.c create mode 100644 sound/soc/codecs/nau8824.h diff --git a/Documentation/devicetree/bindings/sound/nau8824.txt b/Documentation/devicetree/bindings/sound/nau8824.txt new file mode 100644 index 0000000..e0058b9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8824.txt @@ -0,0 +1,88 @@ +Nuvoton NAU8824 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8824" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-compare-time: SAR compare time + 0 - 500 ns + 1 - 1 us + 2 - 2 us + 3 - 4 us + + - nuvoton,sar-sampling-time: SAR sampling time + 0 - 2 us + 1 - 4 us + 2 - 8 us + 3 - 16 us + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30 ms + 1 - 50 ms + 2 - 100 ms + + - nuvoton,jack-eject-debounce: Jack ejection debounce time. + 0 - 0 ms + 1 - 1 ms + 2 - 10 ms + + +Example: + + headset: nau8824@1a { + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1957521..883ed4c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -101,6 +101,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1137,6 +1138,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8824 + tristate "Nuvoton Technology Corporation NAU88L24 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 966eb2e..28a63fd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -95,6 +95,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -328,6 +329,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c new file mode 100644 index 0000000..18552ed --- /dev/null +++ b/sound/soc/codecs/nau8824.c @@ -0,0 +1,1837 @@ +/* + * NAU88L24 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: John Hsu <KCHSU0(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/acpi.h> +#include <linux/math64.h> +#include <linux/semaphore.h> + +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8824.h" + + +static int nau8824_config_sysclk(struct nau8824 *nau8824, + int clk_id, unsigned int freq); +static bool nau8824_is_jack_inserted(struct nau8824 *nau8824); + +/* the ADC threshold of headset */ +#define DMIC_CLK 3072000 + +/* the ADC threshold of headset */ +#define HEADSET_SARADC_THD 0x80 + +/* the parameter threshold of FLL */ +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 124000000 +#define NAU_FVCO_MIN 90000000 + +/* scaling for mclk from sysclk_src output */ +static const struct nau8824_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, +}; + +/* ratio for input clk freq */ +static const struct nau8824_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau8824_fll_attr fll_pre_scalar[] = { + { 1, 0x0 }, + { 2, 0x1 }, + { 4, 0x2 }, + { 8, 0x3 }, +}; + +/* the maximum frequency of CLK_ADC and CLK_DAC */ +#define CLK_DA_AD_MAX 6144000 + +/* over sampling rate */ +static const struct nau8824_osr_attr osr_dac_sel[] = { + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 256, 0 }, /* OSR 256, SRC 1 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 0, 0 }, + { 32, 3 }, /* OSR 32, SRC 1/8 */ +}; + +static const struct nau8824_osr_attr osr_adc_sel[] = { + { 32, 3 }, /* OSR 32, SRC 1/8 */ + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 256, 0 }, /* OSR 256, SRC 1 */ +}; + +static const struct reg_default nau8824_reg_defaults[] = { + { NAU8824_REG_ENA_CTRL, 0x0000 }, + { NAU8824_REG_CLK_GATING_ENA, 0x0000 }, + { NAU8824_REG_CLK_DIVIDER, 0x0000 }, + { NAU8824_REG_FLL1, 0x0000 }, + { NAU8824_REG_FLL2, 0x3126 }, + { NAU8824_REG_FLL3, 0x0008 }, + { NAU8824_REG_FLL4, 0x0010 }, + { NAU8824_REG_FLL5, 0xC000 }, + { NAU8824_REG_FLL6, 0x6000 }, + { NAU8824_REG_FLL_VCO_RSV, 0xF13C }, + { NAU8824_REG_JACK_DET_CTRL, 0x0000 }, + { NAU8824_REG_INTERRUPT_SETTING_1, 0x0000 }, + { NAU8824_REG_IRQ, 0x0000 }, + { NAU8824_REG_CLEAR_INT_REG, 0x0000 }, + { NAU8824_REG_INTERRUPT_SETTING, 0x1000 }, + { NAU8824_REG_SAR_ADC, 0x0015 }, + { NAU8824_REG_VDET_COEFFICIENT, 0x0110 }, + { NAU8824_REG_VDET_THRESHOLD_1, 0x0000 }, + { NAU8824_REG_VDET_THRESHOLD_2, 0x0000 }, + { NAU8824_REG_VDET_THRESHOLD_3, 0x0000 }, + { NAU8824_REG_VDET_THRESHOLD_4, 0x0000 }, + { NAU8824_REG_GPIO_SEL, 0x0000 }, + { NAU8824_REG_PORT0_I2S_PCM_CTRL_1, 0x000B }, + { NAU8824_REG_PORT0_I2S_PCM_CTRL_2, 0x0010 }, + { NAU8824_REG_PORT0_LEFT_TIME_SLOT, 0x0000 }, + { NAU8824_REG_PORT0_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8824_REG_TDM_CTRL, 0x0000 }, + { NAU8824_REG_ADC_HPF_FILTER, 0x0000 }, + { NAU8824_REG_ADC_FILTER_CTRL, 0x0002 }, + { NAU8824_REG_DAC_FILTER_CTRL_1, 0x0000 }, + { NAU8824_REG_DAC_FILTER_CTRL_2, 0x0000 }, + { NAU8824_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8824_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8824_REG_EQ1_LOW, 0x112C }, + { NAU8824_REG_EQ2_EQ3, 0x2C2C }, + { NAU8824_REG_EQ4_EQ5, 0x2C2C }, + { NAU8824_REG_ADC_CH0_DGAIN_CTRL, 0x0100 }, + { NAU8824_REG_ADC_CH1_DGAIN_CTRL, 0x0100 }, + { NAU8824_REG_ADC_CH2_DGAIN_CTRL, 0x0100 }, + { NAU8824_REG_ADC_CH3_DGAIN_CTRL, 0x0100 }, + { NAU8824_REG_DAC_MUTE_CTRL, 0x0000 }, + { NAU8824_REG_DAC_CH0_DGAIN_CTRL, 0x0100 }, + { NAU8824_REG_DAC_CH1_DGAIN_CTRL, 0x0100 }, + { NAU8824_REG_ADC_TO_DAC_ST, 0x0000 }, + { NAU8824_REG_DRC_KNEE_IP12_ADC_CH01, 0x1486 }, + { NAU8824_REG_DRC_KNEE_IP34_ADC_CH01, 0x0F12 }, + { NAU8824_REG_DRC_SLOPE_ADC_CH01, 0x25FF }, + { NAU8824_REG_DRC_ATKDCY_ADC_CH01, 0x3457 }, + { NAU8824_REG_DRC_KNEE_IP12_ADC_CH23, 0x1486 }, + { NAU8824_REG_DRC_KNEE_IP34_ADC_CH23, 0x0F12 }, + { NAU8824_REG_DRC_SLOPE_ADC_CH23, 0x25FF }, + { NAU8824_REG_DRC_ATKDCY_ADC_CH23, 0x3457 }, + { NAU8824_REG_DRC_GAINL_ADC0, 0x0200 }, + { NAU8824_REG_DRC_GAINL_ADC1, 0x0200 }, + { NAU8824_REG_DRC_GAINL_ADC2, 0x0200 }, + { NAU8824_REG_DRC_GAINL_ADC3, 0x0200 }, + { NAU8824_REG_DRC_KNEE_IP12_DAC, 0x1486 }, + { NAU8824_REG_DRC_KNEE_IP34_DAC, 0x0F12 }, + { NAU8824_REG_DRC_SLOPE_DAC, 0x25F9 }, + { NAU8824_REG_DRC_ATKDCY_DAC, 0x3457 }, + { NAU8824_REG_DRC_GAIN_DAC_CH0, 0x0200 }, + { NAU8824_REG_DRC_GAIN_DAC_CH1, 0x0200 }, + { NAU8824_REG_MODE, 0x0000 }, + { NAU8824_REG_MODE1, 0x0000 }, + { NAU8824_REG_MODE2, 0x0000 }, + { NAU8824_REG_CLASSG, 0x0000 }, + { NAU8824_REG_OTP_EFUSE, 0x0000 }, + { NAU8824_REG_OTPDOUT_1, 0x0000 }, + { NAU8824_REG_OTPDOUT_2, 0x0000 }, + { NAU8824_REG_MISC_CTRL, 0x0000 }, + { NAU8824_REG_I2C_TIMEOUT, 0xEFFF }, + { NAU8824_REG_TEST_MODE, 0x0000 }, + { NAU8824_REG_I2C_DEVICE_ID, 0x1AF1 }, + { NAU8824_REG_SAR_ADC_DATA_OUT, 0x00FF }, + { NAU8824_REG_BIAS_ADJ, 0x0000 }, + { NAU8824_REG_PGA_GAIN, 0x0000 }, + { NAU8824_REG_TRIM_SETTINGS, 0x0000 }, + { NAU8824_REG_ANALOG_CONTROL_1, 0x0000 }, + { NAU8824_REG_ANALOG_CONTROL_2, 0x0000 }, + { NAU8824_REG_ENABLE_LO, 0x0000 }, + { NAU8824_REG_GAIN_LO, 0x0000 }, + { NAU8824_REG_CLASSD_GAIN_1, 0x0000 }, + { NAU8824_REG_CLASSD_GAIN_2, 0x0000 }, + { NAU8824_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8824_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8824_REG_RDAC, 0x0008 }, + { NAU8824_REG_MIC_BIAS, 0x0006 }, + { NAU8824_REG_HS_VOLUME_CONTROL, 0x0000 }, + { NAU8824_REG_BOOST, 0x0000 }, + { NAU8824_REG_FEPGA, 0x0000 }, + { NAU8824_REG_FEPGA_II, 0x0000 }, + { NAU8824_REG_FEPGA_SE, 0x0000 }, + { NAU8824_REG_FEPGA_ATTENUATION, 0x0000 }, + { NAU8824_REG_ATT_PORT0, 0x0000 }, + { NAU8824_REG_ATT_PORT1, 0x0000 }, + { NAU8824_REG_POWER_UP_CONTROL, 0x0000 }, + { NAU8824_REG_CHARGE_PUMP_CONTROL, 0x0300 }, + { NAU8824_REG_CHARGE_PUMP_INPUT, 0x0013 }, +}; + +static int nau8824_sema_acquire(struct nau8824 *nau8824, long timeout) +{ + int ret; + + if (timeout) { + ret = down_timeout(&nau8824->jd_sem, timeout); + if (ret < 0) + dev_warn(nau8824->dev, "Acquire semaphone timeout\n"); + } else { + ret = down_interruptible(&nau8824->jd_sem); + if (ret < 0) + dev_warn(nau8824->dev, "Acquire semaphone fail\n"); + } + + return ret; +} + +static inline void nau8824_sema_release(struct nau8824 *nau8824) +{ + up(&nau8824->jd_sem); +} + +static bool nau8824_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8824_REG_ENA_CTRL ... NAU8824_REG_FLL_VCO_RSV: + case NAU8824_REG_JACK_DET_CTRL: + case NAU8824_REG_INTERRUPT_SETTING_1: + case NAU8824_REG_IRQ: + case NAU8824_REG_CLEAR_INT_REG ... NAU8824_REG_VDET_THRESHOLD_4: + case NAU8824_REG_GPIO_SEL: + case NAU8824_REG_PORT0_I2S_PCM_CTRL_1 ... NAU8824_REG_TDM_CTRL: + case NAU8824_REG_ADC_HPF_FILTER ... NAU8824_REG_EQ4_EQ5: + case NAU8824_REG_ADC_CH0_DGAIN_CTRL ... NAU8824_REG_ADC_TO_DAC_ST: + case NAU8824_REG_DRC_KNEE_IP12_ADC_CH01 ... NAU8824_REG_DRC_GAINL_ADC3: + case NAU8824_REG_DRC_KNEE_IP12_DAC ... NAU8824_REG_DRC_GAIN_DAC_CH1: + case NAU8824_REG_CLASSG ... NAU8824_REG_OTP_EFUSE: + case NAU8824_REG_OTPDOUT_1 ... NAU8824_REG_OTPDOUT_2: + case NAU8824_REG_I2C_TIMEOUT: + case NAU8824_REG_I2C_DEVICE_ID ... NAU8824_REG_SAR_ADC_DATA_OUT: + case NAU8824_REG_BIAS_ADJ ... NAU8824_REG_CLASSD_GAIN_2: + case NAU8824_REG_ANALOG_ADC_1 ... NAU8824_REG_ATT_PORT1: + case NAU8824_REG_POWER_UP_CONTROL ... NAU8824_REG_CHARGE_PUMP_INPUT: + return true; + default: + return false; + } + +} + +static bool nau8824_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8824_REG_RESET ... NAU8824_REG_FLL_VCO_RSV: + case NAU8824_REG_JACK_DET_CTRL: + case NAU8824_REG_INTERRUPT_SETTING_1: + case NAU8824_REG_CLEAR_INT_REG ... NAU8824_REG_VDET_THRESHOLD_4: + case NAU8824_REG_GPIO_SEL: + case NAU8824_REG_PORT0_I2S_PCM_CTRL_1 ... NAU8824_REG_TDM_CTRL: + case NAU8824_REG_ADC_HPF_FILTER ... NAU8824_REG_EQ4_EQ5: + case NAU8824_REG_ADC_CH0_DGAIN_CTRL ... NAU8824_REG_ADC_TO_DAC_ST: + case NAU8824_REG_DRC_KNEE_IP12_ADC_CH01: + case NAU8824_REG_DRC_KNEE_IP34_ADC_CH01: + case NAU8824_REG_DRC_SLOPE_ADC_CH01: + case NAU8824_REG_DRC_ATKDCY_ADC_CH01: + case NAU8824_REG_DRC_KNEE_IP12_ADC_CH23: + case NAU8824_REG_DRC_KNEE_IP34_ADC_CH23: + case NAU8824_REG_DRC_SLOPE_ADC_CH23: + case NAU8824_REG_DRC_ATKDCY_ADC_CH23: + case NAU8824_REG_DRC_KNEE_IP12_DAC ... NAU8824_REG_DRC_ATKDCY_DAC: + case NAU8824_REG_CLASSG ... NAU8824_REG_OTP_EFUSE: + case NAU8824_REG_I2C_TIMEOUT: + case NAU8824_REG_BIAS_ADJ ... NAU8824_REG_CLASSD_GAIN_2: + case NAU8824_REG_ANALOG_ADC_1 ... NAU8824_REG_ATT_PORT1: + case NAU8824_REG_POWER_UP_CONTROL ... NAU8824_REG_CHARGE_PUMP_CONTROL: + return true; + default: + return false; + } +} + +static bool nau8824_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8824_REG_RESET: + case NAU8824_REG_IRQ ... NAU8824_REG_CLEAR_INT_REG: + case NAU8824_REG_DRC_GAINL_ADC0 ... NAU8824_REG_DRC_GAINL_ADC3: + case NAU8824_REG_DRC_GAIN_DAC_CH0 ... NAU8824_REG_DRC_GAIN_DAC_CH1: + case NAU8824_REG_OTPDOUT_1 ... NAU8824_REG_OTPDOUT_2: + case NAU8824_REG_I2C_DEVICE_ID ... NAU8824_REG_SAR_ADC_DATA_OUT: + case NAU8824_REG_CHARGE_PUMP_INPUT: + return true; + default: + return false; + } +} + +static const char * const nau8824_companding[] = { + "Off", "NC", "u-law", "A-law" }; + +static const struct soc_enum nau8824_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8824_REG_PORT0_I2S_PCM_CTRL_1, 12, + ARRAY_SIZE(nau8824_companding), nau8824_companding); + +static const struct soc_enum nau8824_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8824_REG_PORT0_I2S_PCM_CTRL_1, 14, + ARRAY_SIZE(nau8824_companding), nau8824_companding); + +static const char * const nau8824_adc_decimation[] = { + "32", "64", "128", "256" }; + +static const struct soc_enum nau8824_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8824_REG_ADC_FILTER_CTRL, 0, + ARRAY_SIZE(nau8824_adc_decimation), nau8824_adc_decimation); + +static const char * const nau8824_dac_oversampl[] = { + "64", "256", "128", "", "32" }; + +static const struct soc_enum nau8824_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8824_REG_DAC_FILTER_CTRL_1, 0, + ARRAY_SIZE(nau8824_dac_oversampl), nau8824_dac_oversampl); + +static const char * const nau8824_input_channel[] = { + "Input CH0", "Input CH1", "Input CH2", "Input CH3" }; + +static const struct soc_enum nau8824_adc_ch0_enum = + SOC_ENUM_SINGLE(NAU8824_REG_ADC_CH0_DGAIN_CTRL, 9, + ARRAY_SIZE(nau8824_input_channel), nau8824_input_channel); + +static const struct soc_enum nau8824_adc_ch1_enum = + SOC_ENUM_SINGLE(NAU8824_REG_ADC_CH1_DGAIN_CTRL, 9, + ARRAY_SIZE(nau8824_input_channel), nau8824_input_channel); + +static const struct soc_enum nau8824_adc_ch2_enum = + SOC_ENUM_SINGLE(NAU8824_REG_ADC_CH2_DGAIN_CTRL, 9, + ARRAY_SIZE(nau8824_input_channel), nau8824_input_channel); + +static const struct soc_enum nau8824_adc_ch3_enum = + SOC_ENUM_SINGLE(NAU8824_REG_ADC_CH3_DGAIN_CTRL, 9, + ARRAY_SIZE(nau8824_input_channel), nau8824_input_channel); + +static const char * const nau8824_tdm_slot[] = { + "Slot 0", "Slot 1", "Slot 2", "Slot 3" }; + +static const struct soc_enum nau8824_dac_left_sel_enum = + SOC_ENUM_SINGLE(NAU8824_REG_TDM_CTRL, 6, + ARRAY_SIZE(nau8824_tdm_slot), nau8824_tdm_slot); + +static const struct soc_enum nau8824_dac_right_sel_enum = + SOC_ENUM_SINGLE(NAU8824_REG_TDM_CTRL, 4, + ARRAY_SIZE(nau8824_tdm_slot), nau8824_tdm_slot); + +static const DECLARE_TLV_DB_MINMAX_MUTE(spk_vol_tlv, 0, 2400); +static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -3000, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 200, 0); +static const DECLARE_TLV_DB_SCALE(dmic_vol_tlv, -12800, 50, 0); + +static const struct snd_kcontrol_new nau8824_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8824_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8824_companding_dac_enum), + + SOC_ENUM("ADC Decimation Rate", nau8824_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), + + SOC_SINGLE_TLV("Speaker Right from DACR Volume", + NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + SOC_SINGLE_TLV("Speaker Left from DACL Volume", + NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + SOC_SINGLE_TLV("Speaker Left from DACR Volume", + NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + SOC_SINGLE_TLV("Speaker Right from DACL Volume", + NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + + SOC_SINGLE_TLV("Headphone Right from DACR Volume", + NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Headphone Left from DACL Volume", + NAU8824_REG_ATT_PORT0, 0, 0x1f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Headphone Right from DACL Volume", + NAU8824_REG_ATT_PORT1, 8, 0x1f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Headphone Left from DACR Volume", + NAU8824_REG_ATT_PORT1, 0, 0x1f, 0, hp_vol_tlv), + + SOC_SINGLE_TLV("Mic1 Volume", NAU8824_REG_FEPGA_II, + NAU8824_FEPGA_GAINL_SFT, 0x12, 0, mic_vol_tlv), + SOC_SINGLE_TLV("Mic2 Volume", NAU8824_REG_FEPGA_II, + NAU8824_FEPGA_GAINR_SFT, 0x12, 0, mic_vol_tlv), + + SOC_SINGLE_TLV("DMIC1 Volume", NAU8824_REG_ADC_CH0_DGAIN_CTRL, + 0, 0x164, 0, dmic_vol_tlv), + SOC_SINGLE_TLV("DMIC2 Volume", NAU8824_REG_ADC_CH1_DGAIN_CTRL, + 0, 0x164, 0, dmic_vol_tlv), + SOC_SINGLE_TLV("DMIC3 Volume", NAU8824_REG_ADC_CH2_DGAIN_CTRL, + 0, 0x164, 0, dmic_vol_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", NAU8824_REG_ADC_CH3_DGAIN_CTRL, + 0, 0x164, 0, dmic_vol_tlv), + + SOC_ENUM("ADC CH0 Select", nau8824_adc_ch0_enum), + SOC_ENUM("ADC CH1 Select", nau8824_adc_ch1_enum), + SOC_ENUM("ADC CH2 Select", nau8824_adc_ch2_enum), + SOC_ENUM("ADC CH3 Select", nau8824_adc_ch3_enum), + + SOC_SINGLE("ADC CH0 TX Switch", NAU8824_REG_TDM_CTRL, 0, 1, 0), + SOC_SINGLE("ADC CH1 TX Switch", NAU8824_REG_TDM_CTRL, 1, 1, 0), + SOC_SINGLE("ADC CH2 TX Switch", NAU8824_REG_TDM_CTRL, 2, 1, 0), + SOC_SINGLE("ADC CH3 TX Switch", NAU8824_REG_TDM_CTRL, 3, 1, 0), + + SOC_ENUM("DACL Channel Source", nau8824_dac_left_sel_enum), + SOC_ENUM("DACR Channel Source", nau8824_dac_right_sel_enum), + + SOC_SINGLE("DACL LR Mix", NAU8824_REG_DAC_MUTE_CTRL, 0, 1, 0), + SOC_SINGLE("DACR LR Mix", NAU8824_REG_DAC_MUTE_CTRL, 1, 1, 0), +}; + +static int nau8824_output_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8824->regmap, NAU8824_REG_ENABLE_LO, + NAU8824_TEST_DAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8824->regmap, NAU8824_REG_ENABLE_LO, + NAU8824_TEST_DAC_EN, NAU8824_TEST_DAC_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8824_spk_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 nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(nau8824->regmap, + NAU8824_REG_ANALOG_CONTROL_2, + NAU8824_CLASSD_CLAMP_DIS, NAU8824_CLASSD_CLAMP_DIS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8824->regmap, + NAU8824_REG_ANALOG_CONTROL_2, + NAU8824_CLASSD_CLAMP_DIS, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8824_pump_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 nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(10); + regmap_update_bits(nau8824->regmap, + NAU8824_REG_CHARGE_PUMP_CONTROL, + NAU8824_JAMNODCLOW, NAU8824_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8824->regmap, + NAU8824_REG_CHARGE_PUMP_CONTROL, + NAU8824_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int system_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + /* Set clock source to disable or internal clock before the + * playback or capture end. Codec needs clock for Jack + * detection and button press if jack inserted; otherwise, + * the clock should be closed. + */ + if (nau8824_is_jack_inserted(nau8824)) { + nau8824_config_sysclk(nau8824, + NAU8824_CLK_INTERNAL, 0); + } else { + nau8824_config_sysclk(nau8824, NAU8824_CLK_DIS, 0); + } + } + return 0; +} + +static int dmic_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + int src; + + /* The DMIC clock is gotten from system clock (256fs) divided by + * DMIC_SRC (1, 2, 4, 8, 16, 32). The clock has to be equal or + * less than 3.072 MHz. + */ + for (src = 0; src < 5; src++) { + if ((0x1 << (8 - src)) * nau8824->fs <= DMIC_CLK) + break; + } + dev_dbg(nau8824->dev, "dmic src %d for mclk %d\n", src, nau8824->fs * 256); + regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_DMIC_SRC_MASK, (src << NAU8824_CLK_DMIC_SRC_SFT)); + + return 0; +} + +static const struct snd_kcontrol_new nau8824_adc_ch0_dmic = + SOC_DAPM_SINGLE("Switch", NAU8824_REG_ENA_CTRL, + NAU8824_ADC_CH0_DMIC_SFT, 1, 0); + +static const struct snd_kcontrol_new nau8824_adc_ch1_dmic = + SOC_DAPM_SINGLE("Switch", NAU8824_REG_ENA_CTRL, + NAU8824_ADC_CH1_DMIC_SFT, 1, 0); + +static const struct snd_kcontrol_new nau8824_adc_ch2_dmic = + SOC_DAPM_SINGLE("Switch", NAU8824_REG_ENA_CTRL, + NAU8824_ADC_CH2_DMIC_SFT, 1, 0); + +static const struct snd_kcontrol_new nau8824_adc_ch3_dmic = + SOC_DAPM_SINGLE("Switch", NAU8824_REG_ENA_CTRL, + NAU8824_ADC_CH3_DMIC_SFT, 1, 0); + +static const struct snd_kcontrol_new nau8824_adc_left_mixer[] = { + SOC_DAPM_SINGLE("MIC Switch", NAU8824_REG_FEPGA, + NAU8824_FEPGA_MODEL_MIC1_SFT, 1, 0), + SOC_DAPM_SINGLE("HSMIC Switch", NAU8824_REG_FEPGA, + NAU8824_FEPGA_MODEL_HSMIC_SFT, 1, 0), +}; + +static const struct snd_kcontrol_new nau8824_adc_right_mixer[] = { + SOC_DAPM_SINGLE("MIC Switch", NAU8824_REG_FEPGA, + NAU8824_FEPGA_MODER_MIC2_SFT, 1, 0), + SOC_DAPM_SINGLE("HSMIC Switch", NAU8824_REG_FEPGA, + NAU8824_FEPGA_MODER_HSMIC_SFT, 1, 0), +}; + +static const struct snd_kcontrol_new nau8824_hp_left_mixer[] = { + SOC_DAPM_SINGLE("DAC Right Switch", NAU8824_REG_ENABLE_LO, + NAU8824_DACR_HPL_EN_SFT, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", NAU8824_REG_ENABLE_LO, + NAU8824_DACL_HPL_EN_SFT, 1, 0), +}; + +static const struct snd_kcontrol_new nau8824_hp_right_mixer[] = { + SOC_DAPM_SINGLE("DAC Left Switch", NAU8824_REG_ENABLE_LO, + NAU8824_DACL_HPR_EN_SFT, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", NAU8824_REG_ENABLE_LO, + NAU8824_DACR_HPR_EN_SFT, 1, 0), +}; + +static const char * const nau8824_dac_src[] = { "DACL", "DACR" }; + +static SOC_ENUM_SINGLE_DECL( + nau8824_dacl_enum, NAU8824_REG_DAC_CH0_DGAIN_CTRL, + NAU8824_DAC_CH0_SEL_SFT, nau8824_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8824_dacr_enum, NAU8824_REG_DAC_CH1_DGAIN_CTRL, + NAU8824_DAC_CH1_SEL_SFT, nau8824_dac_src); + +static const struct snd_kcontrol_new nau8824_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8824_dacl_enum); + +static const struct snd_kcontrol_new nau8824_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8824_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("System Clock", SND_SOC_NOPM, 0, 0, + system_clock_control, SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("HSMIC1"), + SND_SOC_DAPM_INPUT("HSMIC2"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("SAR", NAU8824_REG_SAR_ADC, + NAU8824_SAR_ADC_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", NAU8824_REG_MIC_BIAS, + NAU8824_MICBIAS_POWERUP_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC12 Power", NAU8824_REG_BIAS_ADJ, + NAU8824_DMIC1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC34 Power", NAU8824_REG_BIAS_ADJ, + NAU8824_DMIC2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Clock", SND_SOC_NOPM, 0, 0, + dmic_clock_control, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SWITCH("DMIC1 Enable", SND_SOC_NOPM, + 0, 0, &nau8824_adc_ch0_dmic), + SND_SOC_DAPM_SWITCH("DMIC2 Enable", SND_SOC_NOPM, + 0, 0, &nau8824_adc_ch1_dmic), + SND_SOC_DAPM_SWITCH("DMIC3 Enable", SND_SOC_NOPM, + 0, 0, &nau8824_adc_ch2_dmic), + SND_SOC_DAPM_SWITCH("DMIC4 Enable", SND_SOC_NOPM, + 0, 0, &nau8824_adc_ch3_dmic), + + SND_SOC_DAPM_MIXER("Left ADC", NAU8824_REG_POWER_UP_CONTROL, + 12, 0, nau8824_adc_left_mixer, + ARRAY_SIZE(nau8824_adc_left_mixer)), + SND_SOC_DAPM_MIXER("Right ADC", NAU8824_REG_POWER_UP_CONTROL, + 13, 0, nau8824_adc_right_mixer, + ARRAY_SIZE(nau8824_adc_right_mixer)), + + SND_SOC_DAPM_ADC("ADCL", NULL, NAU8824_REG_ANALOG_ADC_2, + NAU8824_ADCL_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, NAU8824_REG_ANALOG_ADC_2, + NAU8824_ADCR_EN_SFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFTX", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFRX", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, NAU8824_REG_RDAC, + NAU8824_DACL_EN_SFT, 0), + SND_SOC_DAPM_SUPPLY("DACL Clock", NAU8824_REG_RDAC, + NAU8824_DACL_CLK_SFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DACR", NULL, NAU8824_REG_RDAC, + NAU8824_DACR_EN_SFT, 0), + SND_SOC_DAPM_SUPPLY("DACR Clock", NAU8824_REG_RDAC, + NAU8824_DACR_CLK_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8824_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8824_dacr_mux), + + SND_SOC_DAPM_PGA_S("Output DACL", 0, NAU8824_REG_CHARGE_PUMP_CONTROL, + 8, 1, nau8824_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 0, NAU8824_REG_CHARGE_PUMP_CONTROL, + 9, 1, nau8824_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_S("ClassD", 0, NAU8824_REG_CLASSD_GAIN_1, + NAU8824_CLASSD_EN_SFT, 0, nau8824_spk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("Left Headphone", NAU8824_REG_CLASSG, + NAU8824_CLASSG_LDAC_EN_SFT, 0, nau8824_hp_left_mixer, + ARRAY_SIZE(nau8824_hp_left_mixer)), + SND_SOC_DAPM_MIXER("Right Headphone", NAU8824_REG_CLASSG, + NAU8824_CLASSG_RDAC_EN_SFT, 0, nau8824_hp_right_mixer, + ARRAY_SIZE(nau8824_hp_right_mixer)), + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8824_REG_CHARGE_PUMP_CONTROL, + NAU8824_CHARGE_PUMP_EN_SFT, 0, nau8824_pump_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("Output Driver L", + NAU8824_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver R", + NAU8824_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Main Driver L", + NAU8824_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Main Driver R", + NAU8824_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Boost Driver", NAU8824_REG_BOOST, + NAU8824_HP_BOOST_DIS_SFT, 1, NULL, 0), + SND_SOC_DAPM_PGA("Class G", NAU8824_REG_CLASSG, + NAU8824_CLASSG_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("SPKOUTL"), + SND_SOC_DAPM_OUTPUT("SPKOUTR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8824_dapm_routes[] = { + {"DMIC1 Enable", "Switch", "DMIC1"}, + {"DMIC2 Enable", "Switch", "DMIC2"}, + {"DMIC3 Enable", "Switch", "DMIC3"}, + {"DMIC4 Enable", "Switch", "DMIC4"}, + + {"DMIC1", NULL, "DMIC12 Power"}, + {"DMIC2", NULL, "DMIC12 Power"}, + {"DMIC3", NULL, "DMIC34 Power"}, + {"DMIC4", NULL, "DMIC34 Power"}, + {"DMIC12 Power", NULL, "DMIC Clock"}, + {"DMIC34 Power", NULL, "DMIC Clock"}, + + {"Left ADC", "MIC Switch", "MIC1"}, + {"Left ADC", "HSMIC Switch", "HSMIC1"}, + {"Right ADC", "MIC Switch", "MIC2"}, + {"Right ADC", "HSMIC Switch", "HSMIC2"}, + + {"ADCL", NULL, "Left ADC"}, + {"ADCR", NULL, "Right ADC"}, + + {"AIFTX", NULL, "MICBIAS"}, + {"AIFTX", NULL, "ADCL"}, + {"AIFTX", NULL, "ADCR"}, + {"AIFTX", NULL, "DMIC1 Enable"}, + {"AIFTX", NULL, "DMIC2 Enable"}, + {"AIFTX", NULL, "DMIC3 Enable"}, + {"AIFTX", NULL, "DMIC4 Enable"}, + + {"AIFTX", NULL, "System Clock"}, + {"AIFRX", NULL, "System Clock"}, + + {"DACL", NULL, "AIFRX"}, + {"DACL", NULL, "DACL Clock"}, + {"DACR", NULL, "AIFRX"}, + {"DACR", NULL, "DACR Clock"}, + + {"DACL Mux", "DACL", "DACL"}, + {"DACL Mux", "DACR", "DACR"}, + {"DACR Mux", "DACL", "DACL"}, + {"DACR Mux", "DACR", "DACR"}, + + {"Output DACL", NULL, "DACL Mux"}, + {"Output DACR", NULL, "DACR Mux"}, + + {"ClassD", NULL, "Output DACL"}, + {"ClassD", NULL, "Output DACR"}, + + {"Left Headphone", "DAC Left Switch", "Output DACL"}, + {"Left Headphone", "DAC Right Switch", "Output DACR"}, + {"Right Headphone", "DAC Left Switch", "Output DACL"}, + {"Right Headphone", "DAC Right Switch", "Output DACR"}, + + {"Charge Pump", NULL, "Left Headphone"}, + {"Charge Pump", NULL, "Right Headphone"}, + {"Output Driver L", NULL, "Charge Pump"}, + {"Output Driver R", NULL, "Charge Pump"}, + {"Main Driver L", NULL, "Output Driver L"}, + {"Main Driver R", NULL, "Output Driver R"}, + {"Class G", NULL, "Main Driver L"}, + {"Class G", NULL, "Main Driver R"}, + {"HP Boost Driver", NULL, "Class G"}, + + {"SPKOUTL", NULL, "ClassD"}, + {"SPKOUTR", NULL, "ClassD"}, + {"HPOL", NULL, "HP Boost Driver"}, + {"HPOR", NULL, "HP Boost Driver"}, +}; + +static bool nau8824_is_jack_inserted(struct nau8824 *nau8824) +{ + struct snd_soc_jack *jack = nau8824->jack; + bool insert = FALSE; + + if (nau8824->irq && jack) + insert = jack->status & SND_JACK_HEADPHONE; + + return insert; +} + +static void nau8824_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8824_REG_IRQ, &active_irq); + for (i = 0; i < NAU8824_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8824_REG_CLEAR_INT_REG, clear_irq); + } +} + +static void nau8824_eject_jack(struct nau8824 *nau8824) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + struct regmap *regmap = nau8824->regmap; + + /* Clear all interruption status */ + nau8824_int_status_clear_all(regmap); + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + + /* Enable the insertion interruption, disable the ejection + * interruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING, + NAU8824_IRQ_KEY_RELEASE_DIS | NAU8824_IRQ_KEY_SHORT_PRESS_DIS | + NAU8824_IRQ_EJECT_DIS | NAU8824_IRQ_INSERT_DIS, + NAU8824_IRQ_KEY_RELEASE_DIS | NAU8824_IRQ_KEY_SHORT_PRESS_DIS | + NAU8824_IRQ_EJECT_DIS); + regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING_1, + NAU8824_IRQ_INSERT_EN | NAU8824_IRQ_EJECT_EN, + NAU8824_IRQ_INSERT_EN); + regmap_update_bits(regmap, NAU8824_REG_ENA_CTRL, + NAU8824_JD_SLEEP_MODE, NAU8824_JD_SLEEP_MODE); + + /* Close clock for jack type detection at manual mode */ + nau8824_config_sysclk(nau8824, NAU8824_CLK_DIS, 0); +} + +static void nau8824_jdet_work(struct work_struct *work) +{ + struct nau8824 *nau8824 = container_of( + work, struct nau8824, jdet_work); + struct snd_soc_dapm_context *dapm = nau8824->dapm; + struct regmap *regmap = nau8824->regmap; + int adc_value, event = 0, event_mask = 0; + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + + msleep(100); + + regmap_read(regmap, NAU8824_REG_SAR_ADC_DATA_OUT, &adc_value); + adc_value = adc_value & NAU8824_SAR_ADC_DATA_MASK; + dev_dbg(nau8824->dev, "SAR ADC data 0x%02x\n", adc_value); + if (adc_value < HEADSET_SARADC_THD) { + event |= SND_JACK_HEADPHONE; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + } else { + event |= SND_JACK_HEADSET; + } + event_mask |= SND_JACK_HEADSET; + snd_soc_jack_report(nau8824->jack, event, event_mask); + + nau8824_sema_release(nau8824); +} + +static void nau8824_setup_auto_irq(struct nau8824 *nau8824) +{ + struct regmap *regmap = nau8824->regmap; + + /* Enable jack ejection, short key press and release interruption. */ + regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING_1, + NAU8824_IRQ_INSERT_EN | NAU8824_IRQ_EJECT_EN, + NAU8824_IRQ_EJECT_EN); + regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING, + NAU8824_IRQ_EJECT_DIS | NAU8824_IRQ_KEY_RELEASE_DIS | + NAU8824_IRQ_KEY_SHORT_PRESS_DIS, 0); + /* Enable internal VCO needed for interruptions */ + nau8824_config_sysclk(nau8824, NAU8824_CLK_INTERNAL, 0); + regmap_update_bits(regmap, NAU8824_REG_ENA_CTRL, + NAU8824_JD_SLEEP_MODE, 0); +} + +static int nau8824_button_decode(int value) +{ + int buttons = 0; + + /* The chip supports up to 8 buttons, but ALSA defines + * only 6 buttons. + */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +#define NAU8824_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8824_interrupt(int irq, void *data) +{ + struct nau8824 *nau8824 = (struct nau8824 *)data; + struct regmap *regmap = nau8824->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + if (regmap_read(regmap, NAU8824_REG_IRQ, &active_irq)) { + dev_err(nau8824->dev, "failed to read irq status\n"); + return IRQ_NONE; + } + dev_dbg(nau8824->dev, "IRQ %x\n", active_irq); + + if (active_irq & NAU8824_JACK_EJECTION_DETECTED) { + nau8824_eject_jack(nau8824); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8824_JACK_EJECTION_DETECTED; + /* release semaphore held after resume, + * and cancel jack detection + */ + nau8824_sema_release(nau8824); + cancel_work_sync(&nau8824->jdet_work); + } else if (active_irq & NAU8824_KEY_SHORT_PRESS_IRQ) { + int key_status, button_pressed; + + regmap_read(regmap, NAU8824_REG_CLEAR_INT_REG, + &key_status); + + /* lower 8 bits of the register are for pressed keys */ + button_pressed = nau8824_button_decode(key_status); + + event |= button_pressed; + dev_dbg(nau8824->dev, "button %x pressed\n", event); + event_mask |= NAU8824_BUTTONS; + clear_irq = NAU8824_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8824_KEY_RELEASE_IRQ) { + event_mask = NAU8824_BUTTONS; + clear_irq = NAU8824_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8824_JACK_INSERTION_DETECTED) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8824_REG_INTERRUPT_SETTING, + NAU8824_IRQ_INSERT_DIS, + NAU8824_IRQ_INSERT_DIS); + regmap_update_bits(regmap, + NAU8824_REG_INTERRUPT_SETTING_1, + NAU8824_IRQ_INSERT_EN, 0); + /* detect microphone and jack type */ + cancel_work_sync(&nau8824->jdet_work); + schedule_work(&nau8824->jdet_work); + + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8824_setup_auto_irq(nau8824); + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8824_REG_CLEAR_INT_REG, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8824->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static int nau8824_clock_check(struct nau8824 *nau8824, + int stream, int rate, int osr) +{ + int osrate; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (osr >= ARRAY_SIZE(osr_dac_sel)) + return -EINVAL; + osrate = osr_dac_sel[osr].osr; + } else { + if (osr >= ARRAY_SIZE(osr_adc_sel)) + return -EINVAL; + osrate = osr_adc_sel[osr].osr; + } + + if (!osrate || rate * osr > CLK_DA_AD_MAX) { + dev_err(nau8824->dev, "exceed the maximum frequency of CLK_ADC or CLK_DAC\n"); + return -EINVAL; + } + + return 0; +} + +static int nau8824_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 nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + + nau8824_sema_acquire(nau8824, HZ); + + /* CLK_DAC or CLK_ADC = OSR * FS + * DAC or ADC clock frequency is defined as Over Sampling Rate (OSR) + * multiplied by the audio sample rate (Fs). Note that the OSR and Fs + * values must be selected such that the maximum frequency is less + * than 6.144 MHz. + */ + nau8824->fs = params_rate(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8824->regmap, + NAU8824_REG_DAC_FILTER_CTRL_1, &osr); + osr &= NAU8824_DAC_OVERSAMPLE_MASK; + if (nau8824_clock_check(nau8824, substream->stream, + nau8824->fs, osr)) + return -EINVAL; + regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_DAC_SRC_MASK, + osr_dac_sel[osr].clk_src << NAU8824_CLK_DAC_SRC_SFT); + } else { + regmap_read(nau8824->regmap, + NAU8824_REG_ADC_FILTER_CTRL, &osr); + osr &= NAU8824_ADC_SYNC_DOWN_MASK; + if (nau8824_clock_check(nau8824, substream->stream, + nau8824->fs, osr)) + return -EINVAL; + regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_ADC_SRC_MASK, + osr_adc_sel[osr].clk_src << NAU8824_CLK_ADC_SRC_SFT); + } + + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8824->regmap, + NAU8824_REG_PORT0_I2S_PCM_CTRL_2, &ctrl_val); + if (ctrl_val & NAU8824_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / nau8824->fs; + if (bclk_fs <= 32) + bclk_div = 0x3; + else if (bclk_fs <= 64) + bclk_div = 0x2; + else if (bclk_fs <= 128) + bclk_div = 0x1; + else if (bclk_fs <= 256) + bclk_div = 0; + else + return -EINVAL; + regmap_update_bits(nau8824->regmap, + NAU8824_REG_PORT0_I2S_PCM_CTRL_2, + NAU8824_I2S_LRC_DIV_MASK | NAU8824_I2S_BLK_DIV_MASK, + (bclk_div << NAU8824_I2S_LRC_DIV_SFT) | bclk_div); + } + + switch (params_width(params)) { + case 16: + val_len |= NAU8824_I2S_DL_16; + break; + case 20: + val_len |= NAU8824_I2S_DL_20; + break; + case 24: + val_len |= NAU8824_I2S_DL_24; + break; + case 32: + val_len |= NAU8824_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, + NAU8824_I2S_DL_MASK, val_len); + + nau8824_sema_release(nau8824); + + return 0; +} + +static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + nau8824_sema_acquire(nau8824, HZ); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8824_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8824_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8824_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8824_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8824_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8824_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8824_I2S_DF_PCM_AB; + ctrl1_val |= NAU8824_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, + NAU8824_I2S_DF_MASK | NAU8824_I2S_BP_MASK | + NAU8824_I2S_PCMB_EN, ctrl1_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_2, + NAU8824_I2S_MS_MASK, ctrl2_val); + + nau8824_sema_release(nau8824); + + return 0; +} + +/** + * nau8824_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau8824_calc_fll_param(unsigned int fll_in, + unsigned int fs, struct nau8824_fll *fll_param) +{ + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing + * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8824_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in / fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } + } + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; + + /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco_max << 16, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 16) & 0x3FF; + fll_param->fll_frac = fvco & 0xFFFF; + return 0; +} + +static void nau8824_fll_apply(struct regmap *regmap, + struct nau8824_fll *fll_param) +{ + regmap_update_bits(regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_SRC_MASK | NAU8824_CLK_MCLK_SRC_MASK, + NAU8824_CLK_SRC_MCLK | fll_param->mclk_src); + regmap_update_bits(regmap, NAU8824_REG_FLL1, + NAU8824_FLL_RATIO_MASK, fll_param->ratio); + /* FLL 16-bit fractional input */ + regmap_write(regmap, NAU8824_REG_FLL2, fll_param->fll_frac); + /* FLL 10-bit integer input */ + regmap_update_bits(regmap, NAU8824_REG_FLL3, + NAU8824_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(regmap, NAU8824_REG_FLL4, + NAU8824_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8824_FLL_REF_DIV_SFT); + /* select divided VCO input */ + regmap_update_bits(regmap, NAU8824_REG_FLL5, + NAU8824_FLL_CLK_SW_MASK, NAU8824_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(regmap, + NAU8824_REG_FLL6, NAU8824_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(regmap, NAU8824_REG_FLL5, + NAU8824_FLL_PDB_DAC_EN | NAU8824_FLL_LOOP_FTR_EN | + NAU8824_FLL_FTR_SW_MASK, + NAU8824_FLL_PDB_DAC_EN | NAU8824_FLL_LOOP_FTR_EN | + NAU8824_FLL_FTR_SW_FILTER); + regmap_update_bits(regmap, NAU8824_REG_FLL6, + NAU8824_SDM_EN, NAU8824_SDM_EN); + } else { + regmap_update_bits(regmap, NAU8824_REG_FLL5, + NAU8824_FLL_PDB_DAC_EN | NAU8824_FLL_LOOP_FTR_EN | + NAU8824_FLL_FTR_SW_MASK, NAU8824_FLL_FTR_SW_ACCU); + regmap_update_bits(regmap, + NAU8824_REG_FLL6, NAU8824_SDM_EN, 0); + } +} + +/* freq_out must be 256*Fs in order to achieve the best performance */ +static int nau8824_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + struct nau8824_fll fll_param; + int ret, fs; + + fs = freq_out / 256; + ret = nau8824_calc_fll_param(freq_in, fs, &fll_param); + if (ret < 0) { + dev_err(nau8824->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_dbg(nau8824->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, + fll_param.fll_int, fll_param.clk_ref_div); + + nau8824_fll_apply(nau8824->regmap, &fll_param); + mdelay(2); + regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_SRC_MASK, NAU8824_CLK_SRC_VCO); + + return 0; +} + +static int nau8824_config_sysclk(struct nau8824 *nau8824, + int clk_id, unsigned int freq) +{ + struct regmap *regmap = nau8824->regmap; + + switch (clk_id) { + case NAU8824_CLK_DIS: + regmap_update_bits(regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_SRC_MASK, NAU8824_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8824_REG_FLL6, + NAU8824_DCO_EN, 0); + break; + + case NAU8824_CLK_MCLK: + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_SRC_MASK, NAU8824_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8824_REG_FLL6, + NAU8824_DCO_EN, 0); + nau8824_sema_release(nau8824); + break; + + case NAU8824_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8824_REG_FLL6, + NAU8824_DCO_EN, NAU8824_DCO_EN); + regmap_update_bits(regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_SRC_MASK, NAU8824_CLK_SRC_VCO); + break; + + case NAU8824_CLK_FLL_MCLK: + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(regmap, NAU8824_REG_FLL3, + NAU8824_FLL_CLK_SRC_MASK, NAU8824_FLL_CLK_SRC_MCLK); + nau8824_sema_release(nau8824); + break; + + case NAU8824_CLK_FLL_BLK: + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(regmap, NAU8824_REG_FLL3, + NAU8824_FLL_CLK_SRC_MASK, NAU8824_FLL_CLK_SRC_BLK); + nau8824_sema_release(nau8824); + break; + + case NAU8824_CLK_FLL_FS: + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(regmap, NAU8824_REG_FLL3, + NAU8824_FLL_CLK_SRC_MASK, NAU8824_FLL_CLK_SRC_FS); + nau8824_sema_release(nau8824); + break; + + default: + dev_err(nau8824->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8824->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + + return 0; +} + +static int nau8824_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + return nau8824_config_sysclk(nau8824, clk_id, freq); +} + +static void nau8824_resume_setup(struct nau8824 *nau8824) +{ + nau8824_config_sysclk(nau8824, NAU8824_CLK_DIS, 0); + if (nau8824->irq) { + /* Clear all interruption status */ + nau8824_int_status_clear_all(nau8824->regmap); + /* Enable jack detection at sleep mode, insertion detection, + * and ejection detection. + */ + regmap_update_bits(nau8824->regmap, NAU8824_REG_ENA_CTRL, + NAU8824_JD_SLEEP_MODE, NAU8824_JD_SLEEP_MODE); + regmap_update_bits(nau8824->regmap, + NAU8824_REG_INTERRUPT_SETTING_1, + NAU8824_IRQ_EJECT_EN | NAU8824_IRQ_INSERT_EN, + NAU8824_IRQ_EJECT_EN | NAU8824_IRQ_INSERT_EN); + regmap_update_bits(nau8824->regmap, + NAU8824_REG_INTERRUPT_SETTING, + NAU8824_IRQ_EJECT_DIS | NAU8824_IRQ_INSERT_DIS, 0); + } +} + +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* Setup codec configuration after resume */ + nau8824_resume_setup(nau8824); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(nau8824->regmap, + NAU8824_REG_INTERRUPT_SETTING, 0x3ff, 0x3ff); + regmap_update_bits(nau8824->regmap, + NAU8824_REG_INTERRUPT_SETTING_1, + NAU8824_IRQ_EJECT_EN | NAU8824_IRQ_INSERT_EN, 0); + break; + } + + return 0; +} + +static int nau8824_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + nau8824->dapm = dapm; + + return 0; +} + +static int __maybe_unused nau8824_suspend(struct snd_soc_codec *codec) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + if (nau8824->irq) { + disable_irq(nau8824->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + } + regcache_cache_only(nau8824->regmap, true); + regcache_mark_dirty(nau8824->regmap); + + return 0; +} + +static int __maybe_unused nau8824_resume(struct snd_soc_codec *codec) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(nau8824->regmap, false); + regcache_sync(nau8824->regmap); + if (nau8824->irq) { + /* Hold semaphore to postpone playback happening + * until jack detection done. + */ + nau8824_sema_acquire(nau8824, 0); + enable_irq(nau8824->irq); + } + + return 0; +} + +static struct snd_soc_codec_driver nau8824_codec_driver = { + .probe = nau8824_codec_probe, + .set_sysclk = nau8824_set_sysclk, + .set_pll = nau8824_set_pll, + .set_bias_level = nau8824_set_bias_level, + .suspend = nau8824_suspend, + .resume = nau8824_resume, + .suspend_bias_off = true, + + .component_driver = { + .controls = nau8824_snd_controls, + .num_controls = ARRAY_SIZE(nau8824_snd_controls), + .dapm_widgets = nau8824_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8824_dapm_widgets), + .dapm_routes = nau8824_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8824_dapm_routes), + }, +}; + +static const struct snd_soc_dai_ops nau8824_dai_ops = { + .hw_params = nau8824_hw_params, + .set_fmt = nau8824_set_fmt, +}; + +#define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8824_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8824_dai = { + .name = NAU8824_CODEC_DAI, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .ops = &nau8824_dai_ops, +}; + +static const struct regmap_config nau8824_regmap_config = { + .val_bits = NAU8824_REG_ADDR_LEN, + .reg_bits = NAU8824_REG_DATA_LEN, + + .max_register = NAU8824_REG_MAX, + .readable_reg = nau8824_readable_reg, + .writeable_reg = nau8824_writeable_reg, + .volatile_reg = nau8824_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8824_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8824_reg_defaults), +}; + +/** + * nau8824_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8824_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + int ret; + + nau8824->jack = jack; + /* Initiate jack detection work queue */ + INIT_WORK(&nau8824->jdet_work, nau8824_jdet_work); + ret = devm_request_threaded_irq(nau8824->dev, nau8824->irq, NULL, + nau8824_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8824", nau8824); + if (ret) { + dev_err(nau8824->dev, "Cannot request irq %d (%d)\n", + nau8824->irq, ret); + } + + return ret; +} +EXPORT_SYMBOL_GPL(nau8824_enable_jack_detect); + +static void nau8824_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8824_REG_RESET, 0x00); + regmap_write(regmap, NAU8824_REG_RESET, 0x00); +} + +static void nau8824_setup_buttons(struct nau8824 *nau8824) +{ + struct regmap *regmap = nau8824->regmap; + + regmap_update_bits(regmap, NAU8824_REG_SAR_ADC, + NAU8824_SAR_TRACKING_GAIN_MASK, + nau8824->sar_voltage << NAU8824_SAR_TRACKING_GAIN_SFT); + regmap_update_bits(regmap, NAU8824_REG_SAR_ADC, + NAU8824_SAR_COMPARE_TIME_MASK, + nau8824->sar_compare_time << NAU8824_SAR_COMPARE_TIME_SFT); + regmap_update_bits(regmap, NAU8824_REG_SAR_ADC, + NAU8824_SAR_SAMPLING_TIME_MASK, + nau8824->sar_sampling_time << NAU8824_SAR_SAMPLING_TIME_SFT); + + regmap_update_bits(regmap, NAU8824_REG_VDET_COEFFICIENT, + NAU8824_LEVELS_NR_MASK, + (nau8824->sar_threshold_num - 1) << NAU8824_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8824_REG_VDET_COEFFICIENT, + NAU8824_HYSTERESIS_MASK, + nau8824->sar_hysteresis << NAU8824_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8824_REG_VDET_COEFFICIENT, + NAU8824_SHORTKEY_DEBOUNCE_MASK, + nau8824->key_debounce << NAU8824_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8824_REG_VDET_THRESHOLD_1, + (nau8824->sar_threshold[0] << 8) | nau8824->sar_threshold[1]); + regmap_write(regmap, NAU8824_REG_VDET_THRESHOLD_2, + (nau8824->sar_threshold[2] << 8) | nau8824->sar_threshold[3]); + regmap_write(regmap, NAU8824_REG_VDET_THRESHOLD_3, + (nau8824->sar_threshold[4] << 8) | nau8824->sar_threshold[5]); + regmap_write(regmap, NAU8824_REG_VDET_THRESHOLD_4, + (nau8824->sar_threshold[6] << 8) | nau8824->sar_threshold[7]); +} + +static void nau8824_init_regs(struct nau8824 *nau8824) +{ + struct regmap *regmap = nau8824->regmap; + + /* Enable Bias/VMID/VMID Tieoff */ + regmap_update_bits(regmap, NAU8824_REG_BIAS_ADJ, + NAU8824_VMID | NAU8824_VMID_SEL_MASK, NAU8824_VMID | + (nau8824->vref_impedance << NAU8824_VMID_SEL_SFT)); + regmap_update_bits(regmap, NAU8824_REG_BOOST, + NAU8824_GLOBAL_BIAS_EN, NAU8824_GLOBAL_BIAS_EN); + mdelay(2); + regmap_update_bits(regmap, NAU8824_REG_MIC_BIAS, + NAU8824_MICBIAS_VOLTAGE_MASK, nau8824->micbias_voltage); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8824_REG_BOOST, + NAU8824_PRECHARGE_DIS | NAU8824_HP_BOOST_DIS | + NAU8824_HP_BOOST_G_DIS | NAU8824_SHORT_SHUTDOWN_EN, + NAU8824_PRECHARGE_DIS | NAU8824_HP_BOOST_DIS | + NAU8824_HP_BOOST_G_DIS | NAU8824_SHORT_SHUTDOWN_EN); + /* Scaling for ADC and DAC clock */ + regmap_update_bits(regmap, NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_ADC_SRC_MASK | NAU8824_CLK_DAC_SRC_MASK, + (0x1 << NAU8824_CLK_ADC_SRC_SFT) | + (0x1 << NAU8824_CLK_DAC_SRC_SFT)); + regmap_update_bits(regmap, NAU8824_REG_DAC_MUTE_CTRL, + NAU8824_DAC_ZC_EN, NAU8824_DAC_ZC_EN); + regmap_update_bits(regmap, NAU8824_REG_ENA_CTRL, + NAU8824_DAC_CH1_EN | NAU8824_DAC_CH0_EN | + NAU8824_ADC_CH0_EN | NAU8824_ADC_CH1_EN | + NAU8824_ADC_CH2_EN | NAU8824_ADC_CH3_EN, + NAU8824_DAC_CH1_EN | NAU8824_DAC_CH0_EN | + NAU8824_ADC_CH0_EN | NAU8824_ADC_CH1_EN | + NAU8824_ADC_CH2_EN | NAU8824_ADC_CH3_EN); + regmap_update_bits(regmap, NAU8824_REG_CLK_GATING_ENA, + NAU8824_CLK_ADC_CH23_EN | NAU8824_CLK_ADC_CH01_EN | + NAU8824_CLK_DAC_CH1_EN | NAU8824_CLK_DAC_CH0_EN | + NAU8824_CLK_I2S_EN | NAU8824_CLK_GAIN_EN | + NAU8824_CLK_SAR_EN | NAU8824_CLK_DMIC_CH23_EN, + NAU8824_CLK_ADC_CH23_EN | NAU8824_CLK_ADC_CH01_EN | + NAU8824_CLK_DAC_CH1_EN | NAU8824_CLK_DAC_CH0_EN | + NAU8824_CLK_I2S_EN | NAU8824_CLK_GAIN_EN | + NAU8824_CLK_SAR_EN | NAU8824_CLK_DMIC_CH23_EN); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8824_REG_CLASSG, + NAU8824_CLASSG_TIMER_MASK, + 0x20 << NAU8824_CLASSG_TIMER_SFT); + regmap_update_bits(regmap, NAU8824_REG_TRIM_SETTINGS, + NAU8824_DRV_CURR_INC, NAU8824_DRV_CURR_INC); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8824_REG_CHARGE_PUMP_CONTROL, + NAU8824_SPKR_PULL_DOWN | NAU8824_SPKL_PULL_DOWN | + NAU8824_POWER_DOWN_DACR | NAU8824_POWER_DOWN_DACL, + NAU8824_SPKR_PULL_DOWN | NAU8824_SPKL_PULL_DOWN | + NAU8824_POWER_DOWN_DACR | NAU8824_POWER_DOWN_DACL); + /* Enable TESTDAC. This sets the analog DAC inputs to a '0' input + * signal to avoid any glitches due to power up transients in both + * the analog and digital DAC circuit. + */ + regmap_update_bits(regmap, NAU8824_REG_ENABLE_LO, + NAU8824_TEST_DAC_EN, NAU8824_TEST_DAC_EN); + /* Config L/R channel */ + regmap_update_bits(regmap, NAU8824_REG_DAC_CH0_DGAIN_CTRL, + NAU8824_DAC_CH0_SEL_MASK, NAU8824_DAC_CH0_SEL_I2S0); + regmap_update_bits(regmap, NAU8824_REG_DAC_CH1_DGAIN_CTRL, + NAU8824_DAC_CH1_SEL_MASK, NAU8824_DAC_CH1_SEL_I2S1); + regmap_update_bits(regmap, NAU8824_REG_ENABLE_LO, + NAU8824_DACR_HPR_EN | NAU8824_DACL_HPL_EN, + NAU8824_DACR_HPR_EN | NAU8824_DACL_HPL_EN); + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8824_REG_ADC_FILTER_CTRL, + NAU8824_ADC_SYNC_DOWN_MASK, NAU8824_ADC_SYNC_DOWN_64); + regmap_update_bits(regmap, NAU8824_REG_DAC_FILTER_CTRL_1, + NAU8824_DAC_CICCLP_OFF | NAU8824_DAC_OVERSAMPLE_MASK, + NAU8824_DAC_CICCLP_OFF | NAU8824_DAC_OVERSAMPLE_64); + /* Class D gain 9db for 1R and 2L */ + regmap_update_bits(regmap, NAU8824_REG_CLASSD_GAIN_1, + NAU8824_CLASSD_GAIN_1R_MASK, + (0xa << NAU8824_CLASSD_GAIN_1R_SFT)); + regmap_update_bits(regmap, NAU8824_REG_CLASSD_GAIN_2, + NAU8824_CLASSD_GAIN_2L_MASK, 0xa); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8824_REG_RDAC, + NAU8824_RDAC_CLK_DELAY_MASK | NAU8824_RDAC_VREF_MASK, + (0x2 << NAU8824_RDAC_CLK_DELAY_SFT) | + (0x3 << NAU8824_RDAC_VREF_SFT)); + /* PGA input mode selection */ + regmap_update_bits(regmap, NAU8824_REG_FEPGA, + NAU8824_FEPGA_MODEL_SHORT_EN | NAU8824_FEPGA_MODER_SHORT_EN, + NAU8824_FEPGA_MODEL_SHORT_EN | NAU8824_FEPGA_MODER_SHORT_EN); + /* Digital microphone control */ + regmap_update_bits(regmap, NAU8824_REG_ANALOG_CONTROL_1, + NAU8824_DMIC_CLK_DRV_STRG | NAU8824_DMIC_CLK_SLEW_FAST, + NAU8824_DMIC_CLK_DRV_STRG | NAU8824_DMIC_CLK_SLEW_FAST); + regmap_update_bits(regmap, NAU8824_REG_JACK_DET_CTRL, + NAU8824_JACK_LOGIC, + /* jkdet_polarity - 1 is for active-low */ + nau8824->jkdet_polarity ? 0 : NAU8824_JACK_LOGIC); + regmap_update_bits(regmap, + NAU8824_REG_JACK_DET_CTRL, NAU8824_JACK_EJECT_DT_MASK, + (nau8824->jack_eject_debounce << NAU8824_JACK_EJECT_DT_SFT)); + if (nau8824->sar_threshold_num) + nau8824_setup_buttons(nau8824); +} + +static int nau8824_setup_irq(struct nau8824 *nau8824) +{ + /* Disable interruption before codec initiation done */ + regmap_update_bits(nau8824->regmap, NAU8824_REG_ENA_CTRL, + NAU8824_JD_SLEEP_MODE, NAU8824_JD_SLEEP_MODE); + regmap_update_bits(nau8824->regmap, + NAU8824_REG_INTERRUPT_SETTING, 0x3ff, 0x3ff); + regmap_update_bits(nau8824->regmap, NAU8824_REG_INTERRUPT_SETTING_1, + NAU8824_IRQ_EJECT_EN | NAU8824_IRQ_INSERT_EN, 0); + + return 0; +} + +static void nau8824_print_device_properties(struct nau8824 *nau8824) +{ + struct device *dev = nau8824->dev; + int i; + + dev_dbg(dev, "jkdet-polarity: %d\n", nau8824->jkdet_polarity); + dev_dbg(dev, "micbias-voltage: %d\n", nau8824->micbias_voltage); + dev_dbg(dev, "vref-impedance: %d\n", nau8824->vref_impedance); + + dev_dbg(dev, "sar-threshold-num: %d\n", nau8824->sar_threshold_num); + for (i = 0; i < nau8824->sar_threshold_num; i++) + dev_dbg(dev, "sar-threshold[%d]=%x\n", i, + nau8824->sar_threshold[i]); + + dev_dbg(dev, "sar-hysteresis: %d\n", nau8824->sar_hysteresis); + dev_dbg(dev, "sar-voltage: %d\n", nau8824->sar_voltage); + dev_dbg(dev, "sar-compare-time: %d\n", nau8824->sar_compare_time); + dev_dbg(dev, "sar-sampling-time: %d\n", nau8824->sar_sampling_time); + dev_dbg(dev, "short-key-debounce: %d\n", nau8824->key_debounce); + dev_dbg(dev, "jack-eject-debounce: %d\n", + nau8824->jack_eject_debounce); +} + +static int nau8824_read_device_properties(struct device *dev, + struct nau8824 *nau8824) { + int ret; + + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8824->jkdet_polarity); + if (ret) + nau8824->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8824->micbias_voltage); + if (ret) + nau8824->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8824->vref_impedance); + if (ret) + nau8824->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8824->sar_threshold_num); + if (ret) + nau8824->sar_threshold_num = 4; + ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8824->sar_threshold, nau8824->sar_threshold_num); + if (ret) { + nau8824->sar_threshold[0] = 0x0a; + nau8824->sar_threshold[1] = 0x14; + nau8824->sar_threshold[2] = 0x26; + nau8824->sar_threshold[3] = 0x73; + } + ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8824->sar_hysteresis); + if (ret) + nau8824->sar_hysteresis = 0; + ret = device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8824->sar_voltage); + if (ret) + nau8824->sar_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,sar-compare-time", + &nau8824->sar_compare_time); + if (ret) + nau8824->sar_compare_time = 1; + ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time", + &nau8824->sar_sampling_time); + if (ret) + nau8824->sar_sampling_time = 1; + ret = device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8824->key_debounce); + if (ret) + nau8824->key_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8824->jack_eject_debounce); + if (ret) + nau8824->jack_eject_debounce = 1; + + return 0; +} + +static int nau8824_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8824 *nau8824 = dev_get_platdata(dev); + int ret, value; + + if (!nau8824) { + nau8824 = devm_kzalloc(dev, sizeof(*nau8824), GFP_KERNEL); + if (!nau8824) + return -ENOMEM; + ret = nau8824_read_device_properties(dev, nau8824); + if (ret) + return ret; + } + i2c_set_clientdata(i2c, nau8824); + + nau8824->regmap = devm_regmap_init_i2c(i2c, &nau8824_regmap_config); + if (IS_ERR(nau8824->regmap)) + return PTR_ERR(nau8824->regmap); + nau8824->dev = dev; + nau8824->irq = i2c->irq; + sema_init(&nau8824->jd_sem, 1); + + nau8824_print_device_properties(nau8824); + + ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8824: %d\n", + ret); + return ret; + } + nau8824_reset_chip(nau8824->regmap); + nau8824_init_regs(nau8824); + + if (i2c->irq) + nau8824_setup_irq(nau8824); + + return snd_soc_register_codec(dev, + &nau8824_codec_driver, &nau8824_dai, 1); +} + + +static int nau8824_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8824_i2c_ids[] = { + { "nau8824", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8824_i2c_ids); + +#ifdef CONFIG_OF +static const struct of_device_id nau8824_of_ids[] = { + { .compatible = "nuvoton,nau8824", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8824_of_ids); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8824_acpi_match[] = { + { "10508824", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8824_acpi_match); +#endif + +static struct i2c_driver nau8824_i2c_driver = { + .driver = { + .name = "nau8824", + .of_match_table = of_match_ptr(nau8824_of_ids), + .acpi_match_table = ACPI_PTR(nau8824_acpi_match), + }, + .probe = nau8824_i2c_probe, + .remove = nau8824_i2c_remove, + .id_table = nau8824_i2c_ids, +}; +module_i2c_driver(nau8824_i2c_driver); + + +MODULE_DESCRIPTION("ASoC NAU88L24 driver"); +MODULE_AUTHOR("John Hsu <KCHSU0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h new file mode 100644 index 0000000..87ac9a3 --- /dev/null +++ b/sound/soc/codecs/nau8824.h @@ -0,0 +1,466 @@ +/* + * NAU88L24 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: John Hsu <KCHSU0(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8824_H__ +#define __NAU8824_H__ + +#define NAU8824_REG_RESET 0x00 +#define NAU8824_REG_ENA_CTRL 0x01 +#define NAU8824_REG_CLK_GATING_ENA 0x02 +#define NAU8824_REG_CLK_DIVIDER 0x03 +#define NAU8824_REG_FLL1 0x04 +#define NAU8824_REG_FLL2 0x05 +#define NAU8824_REG_FLL3 0x06 +#define NAU8824_REG_FLL4 0x07 +#define NAU8824_REG_FLL5 0x08 +#define NAU8824_REG_FLL6 0x09 +#define NAU8824_REG_FLL_VCO_RSV 0x0A +#define NAU8824_REG_JACK_DET_CTRL 0x0D +#define NAU8824_REG_INTERRUPT_SETTING_1 0x0F +#define NAU8824_REG_IRQ 0x10 +#define NAU8824_REG_CLEAR_INT_REG 0x11 +#define NAU8824_REG_INTERRUPT_SETTING 0x12 +#define NAU8824_REG_SAR_ADC 0x13 +#define NAU8824_REG_VDET_COEFFICIENT 0x14 +#define NAU8824_REG_VDET_THRESHOLD_1 0x15 +#define NAU8824_REG_VDET_THRESHOLD_2 0x16 +#define NAU8824_REG_VDET_THRESHOLD_3 0x17 +#define NAU8824_REG_VDET_THRESHOLD_4 0x18 +#define NAU8824_REG_GPIO_SEL 0x1A +#define NAU8824_REG_PORT0_I2S_PCM_CTRL_1 0x1C +#define NAU8824_REG_PORT0_I2S_PCM_CTRL_2 0x1D +#define NAU8824_REG_PORT0_LEFT_TIME_SLOT 0x1E +#define NAU8824_REG_PORT0_RIGHT_TIME_SLOT 0x1F +#define NAU8824_REG_TDM_CTRL 0x20 +#define NAU8824_REG_ADC_HPF_FILTER 0x23 +#define NAU8824_REG_ADC_FILTER_CTRL 0x24 +#define NAU8824_REG_DAC_FILTER_CTRL_1 0x25 +#define NAU8824_REG_DAC_FILTER_CTRL_2 0x26 +#define NAU8824_REG_NOTCH_FILTER_1 0x27 +#define NAU8824_REG_NOTCH_FILTER_2 0x28 +#define NAU8824_REG_EQ1_LOW 0x29 +#define NAU8824_REG_EQ2_EQ3 0x2A +#define NAU8824_REG_EQ4_EQ5 0x2B +#define NAU8824_REG_ADC_CH0_DGAIN_CTRL 0x2D +#define NAU8824_REG_ADC_CH1_DGAIN_CTRL 0x2E +#define NAU8824_REG_ADC_CH2_DGAIN_CTRL 0x2F +#define NAU8824_REG_ADC_CH3_DGAIN_CTRL 0x30 +#define NAU8824_REG_DAC_MUTE_CTRL 0x31 +#define NAU8824_REG_DAC_CH0_DGAIN_CTRL 0x32 +#define NAU8824_REG_DAC_CH1_DGAIN_CTRL 0x33 +#define NAU8824_REG_ADC_TO_DAC_ST 0x34 +#define NAU8824_REG_DRC_KNEE_IP12_ADC_CH01 0x38 +#define NAU8824_REG_DRC_KNEE_IP34_ADC_CH01 0x39 +#define NAU8824_REG_DRC_SLOPE_ADC_CH01 0x3A +#define NAU8824_REG_DRC_ATKDCY_ADC_CH01 0x3B +#define NAU8824_REG_DRC_KNEE_IP12_ADC_CH23 0x3C +#define NAU8824_REG_DRC_KNEE_IP34_ADC_CH23 0x3D +#define NAU8824_REG_DRC_SLOPE_ADC_CH23 0x3E +#define NAU8824_REG_DRC_ATKDCY_ADC_CH23 0x3F +#define NAU8824_REG_DRC_GAINL_ADC0 0x40 +#define NAU8824_REG_DRC_GAINL_ADC1 0x41 +#define NAU8824_REG_DRC_GAINL_ADC2 0x42 +#define NAU8824_REG_DRC_GAINL_ADC3 0x43 +#define NAU8824_REG_DRC_KNEE_IP12_DAC 0x45 +#define NAU8824_REG_DRC_KNEE_IP34_DAC 0x46 +#define NAU8824_REG_DRC_SLOPE_DAC 0x47 +#define NAU8824_REG_DRC_ATKDCY_DAC 0x48 +#define NAU8824_REG_DRC_GAIN_DAC_CH0 0x49 +#define NAU8824_REG_DRC_GAIN_DAC_CH1 0x4A +#define NAU8824_REG_MODE 0x4C +#define NAU8824_REG_MODE1 0x4D +#define NAU8824_REG_MODE2 0x4E +#define NAU8824_REG_CLASSG 0x50 +#define NAU8824_REG_OTP_EFUSE 0x51 +#define NAU8824_REG_OTPDOUT_1 0x53 +#define NAU8824_REG_OTPDOUT_2 0x54 +#define NAU8824_REG_MISC_CTRL 0x55 +#define NAU8824_REG_I2C_TIMEOUT 0x56 +#define NAU8824_REG_TEST_MODE 0x57 +#define NAU8824_REG_I2C_DEVICE_ID 0x58 +#define NAU8824_REG_SAR_ADC_DATA_OUT 0x59 +#define NAU8824_REG_BIAS_ADJ 0x66 +#define NAU8824_REG_PGA_GAIN 0x67 +#define NAU8824_REG_TRIM_SETTINGS 0x68 +#define NAU8824_REG_ANALOG_CONTROL_1 0x69 +#define NAU8824_REG_ANALOG_CONTROL_2 0x6A +#define NAU8824_REG_ENABLE_LO 0x6B +#define NAU8824_REG_GAIN_LO 0x6C +#define NAU8824_REG_CLASSD_GAIN_1 0x6D +#define NAU8824_REG_CLASSD_GAIN_2 0x6E +#define NAU8824_REG_ANALOG_ADC_1 0x71 +#define NAU8824_REG_ANALOG_ADC_2 0x72 +#define NAU8824_REG_RDAC 0x73 +#define NAU8824_REG_MIC_BIAS 0x74 +#define NAU8824_REG_HS_VOLUME_CONTROL 0x75 +#define NAU8824_REG_BOOST 0x76 +#define NAU8824_REG_FEPGA 0x77 +#define NAU8824_REG_FEPGA_II 0x78 +#define NAU8824_REG_FEPGA_SE 0x79 +#define NAU8824_REG_FEPGA_ATTENUATION 0x7A +#define NAU8824_REG_ATT_PORT0 0x7B +#define NAU8824_REG_ATT_PORT1 0x7C +#define NAU8824_REG_POWER_UP_CONTROL 0x7F +#define NAU8824_REG_CHARGE_PUMP_CONTROL 0x80 +#define NAU8824_REG_CHARGE_PUMP_INPUT 0x81 +#define NAU8824_REG_MAX NAU8824_REG_CHARGE_PUMP_INPUT +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8824_REG_ADDR_LEN 16 +#define NAU8824_REG_DATA_LEN 16 + + +/* ENA_CTRL (0x1) */ +#define NAU8824_DMIC_LCH_EDGE_CH23 (0x1 << 12) +#define NAU8824_DMIC_LCH_EDGE_CH01 (0x1 << 11) +#define NAU8824_JD_SLEEP_MODE (0x1 << 10) +#define NAU8824_ADC_CH3_DMIC_SFT 9 +#define NAU8824_ADC_CH3_DMIC_EN (0x1 << NAU8824_ADC_CH3_DMIC_SFT) +#define NAU8824_ADC_CH2_DMIC_SFT 8 +#define NAU8824_ADC_CH2_DMIC_EN (0x1 << NAU8824_ADC_CH2_DMIC_SFT) +#define NAU8824_ADC_CH1_DMIC_SFT 7 +#define NAU8824_ADC_CH1_DMIC_EN (0x1 << NAU8824_ADC_CH1_DMIC_SFT) +#define NAU8824_ADC_CH0_DMIC_SFT 6 +#define NAU8824_ADC_CH0_DMIC_EN (0x1 << NAU8824_ADC_CH0_DMIC_SFT) +#define NAU8824_DAC_CH1_EN (0x1 << 5) +#define NAU8824_DAC_CH0_EN (0x1 << 4) +#define NAU8824_ADC_CH3_EN (0x1 << 3) +#define NAU8824_ADC_CH2_EN (0x1 << 2) +#define NAU8824_ADC_CH1_EN (0x1 << 1) +#define NAU8824_ADC_CH0_EN 0x1 + +/* CLK_GATING_ENA (0x02) */ +#define NAU8824_CLK_ADC_CH23_EN (0x1 << 15) +#define NAU8824_CLK_ADC_CH01_EN (0x1 << 14) +#define NAU8824_CLK_DAC_CH1_EN (0x1 << 13) +#define NAU8824_CLK_DAC_CH0_EN (0x1 << 12) +#define NAU8824_CLK_I2S_EN (0x1 << 7) +#define NAU8824_CLK_GAIN_EN (0x1 << 5) +#define NAU8824_CLK_SAR_EN (0x1 << 3) +#define NAU8824_CLK_DMIC_CH23_EN (0x1 << 1) + +/* CLK_DIVIDER (0x3) */ +#define NAU8824_CLK_SRC_SFT 15 +#define NAU8824_CLK_SRC_MASK (1 << NAU8824_CLK_SRC_SFT) +#define NAU8824_CLK_SRC_VCO (1 << NAU8824_CLK_SRC_SFT) +#define NAU8824_CLK_SRC_MCLK (0 << NAU8824_CLK_SRC_SFT) +#define NAU8824_CLK_MCLK_SRC_MASK (0xf << 0) +#define NAU8824_CLK_DMIC_SRC_SFT 10 +#define NAU8824_CLK_DMIC_SRC_MASK (0x7 << NAU8824_CLK_DMIC_SRC_SFT) +#define NAU8824_CLK_ADC_SRC_SFT 6 +#define NAU8824_CLK_ADC_SRC_MASK (0x3 << NAU8824_CLK_ADC_SRC_SFT) +#define NAU8824_CLK_DAC_SRC_SFT 4 +#define NAU8824_CLK_DAC_SRC_MASK (0x3 << NAU8824_CLK_DAC_SRC_SFT) + +/* FLL1 (0x04) */ +#define NAU8824_FLL_RATIO_MASK (0x7f << 0) + +/* FLL3 (0x06) */ +#define NAU8824_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8824_FLL_CLK_SRC_SFT 10 +#define NAU8824_FLL_CLK_SRC_MASK (0x3 << NAU8824_FLL_CLK_SRC_SFT) +#define NAU8824_FLL_CLK_SRC_MCLK (0 << NAU8824_FLL_CLK_SRC_SFT) +#define NAU8824_FLL_CLK_SRC_BLK (0x2 << NAU8824_FLL_CLK_SRC_SFT) +#define NAU8824_FLL_CLK_SRC_FS (0x3 << NAU8824_FLL_CLK_SRC_SFT) + +/* FLL4 (0x07) */ +#define NAU8824_FLL_REF_DIV_SFT 10 +#define NAU8824_FLL_REF_DIV_MASK (0x3 << NAU8824_FLL_REF_DIV_SFT) + +/* FLL5 (0x08) */ +#define NAU8824_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8824_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8824_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8824_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8824_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8824_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8824_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8824_FLL_FTR_SW_FILTER (0x0 << 12) + +/* FLL6 (0x9) */ +#define NAU8824_DCO_EN (0x1 << 15) +#define NAU8824_SDM_EN (0x1 << 14) + +/* IRQ (0x10) */ +#define NAU8824_SHORT_CIRCUIT_IRQ (0x1 << 7) +#define NAU8824_IMPEDANCE_MEAS_IRQ (0x1 << 6) +#define NAU8824_KEY_RELEASE_IRQ (0x1 << 5) +#define NAU8824_KEY_LONG_PRESS_IRQ (0x1 << 4) +#define NAU8824_KEY_SHORT_PRESS_IRQ (0x1 << 3) +#define NAU8824_JACK_EJECTION_DETECTED (0x1 << 1) +#define NAU8824_JACK_INSERTION_DETECTED 0x1 + +/* JACK_DET_CTRL (0x0D) */ +#define NAU8824_JACK_EJECT_DT_SFT 2 +#define NAU8824_JACK_EJECT_DT_MASK (0x3 << NAU8824_JACK_EJECT_DT_SFT) +#define NAU8824_JACK_LOGIC 0x1 + + +/* INTERRUPT_SETTING_1 (0x0F) */ +#define NAU8824_IRQ_EJECT_EN (0x1 << 9) +#define NAU8824_IRQ_INSERT_EN (0x1 << 8) + +/* INTERRUPT_SETTING (0x12) */ +#define NAU8824_IRQ_KEY_RELEASE_DIS (0x1 << 5) +#define NAU8824_IRQ_KEY_SHORT_PRESS_DIS (0x1 << 3) +#define NAU8824_IRQ_EJECT_DIS (0x1 << 1) +#define NAU8824_IRQ_INSERT_DIS 0x1 + +/* SAR_ADC (0x13) */ +#define NAU8824_SAR_ADC_EN_SFT 12 +#define NAU8824_SAR_TRACKING_GAIN_SFT 8 +#define NAU8824_SAR_TRACKING_GAIN_MASK (0x7 << NAU8824_SAR_TRACKING_GAIN_SFT) +#define NAU8824_SAR_COMPARE_TIME_SFT 2 +#define NAU8824_SAR_COMPARE_TIME_MASK (3 << 2) +#define NAU8824_SAR_SAMPLING_TIME_SFT 0 +#define NAU8824_SAR_SAMPLING_TIME_MASK (3 << 0) + +/* VDET_COEFFICIENT (0x14) */ +#define NAU8824_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8824_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8824_SHORTKEY_DEBOUNCE_SFT) +#define NAU8824_LEVELS_NR_SFT 8 +#define NAU8824_LEVELS_NR_MASK (0x7 << 8) +#define NAU8824_HYSTERESIS_SFT 0 +#define NAU8824_HYSTERESIS_MASK 0xf + +/* PORT0_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8824_I2S_BP_SFT 7 +#define NAU8824_I2S_BP_MASK (1 << NAU8824_I2S_BP_SFT) +#define NAU8824_I2S_BP_INV (1 << NAU8824_I2S_BP_SFT) +#define NAU8824_I2S_PCMB_SFT 6 +#define NAU8824_I2S_PCMB_EN (1 << NAU8824_I2S_PCMB_SFT) +#define NAU8824_I2S_DL_SFT 2 +#define NAU8824_I2S_DL_MASK (0x3 << NAU8824_I2S_DL_SFT) +#define NAU8824_I2S_DL_16 (0 << NAU8824_I2S_DL_SFT) +#define NAU8824_I2S_DL_20 (1 << NAU8824_I2S_DL_SFT) +#define NAU8824_I2S_DL_24 (2 << NAU8824_I2S_DL_SFT) +#define NAU8824_I2S_DL_32 (3 << NAU8824_I2S_DL_SFT) +#define NAU8824_I2S_DF_MASK 0x3 +#define NAU8824_I2S_DF_RIGTH 0 +#define NAU8824_I2S_DF_LEFT 1 +#define NAU8824_I2S_DF_I2S 2 +#define NAU8824_I2S_DF_PCM_AB 3 + + +/* PORT0_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8824_I2S_LRC_DIV_SFT 12 +#define NAU8824_I2S_LRC_DIV_MASK (0x3 << NAU8824_I2S_LRC_DIV_SFT) +#define NAU8824_I2S_MS_SFT 3 +#define NAU8824_I2S_MS_MASK (1 << NAU8824_I2S_MS_SFT) +#define NAU8824_I2S_MS_MASTER (1 << NAU8824_I2S_MS_SFT) +#define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) +#define NAU8824_I2S_BLK_DIV_MASK 0x7 + +/* ADC_FILTER_CTRL (0x24) */ +#define NAU8824_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8824_ADC_SYNC_DOWN_32 0 +#define NAU8824_ADC_SYNC_DOWN_64 1 +#define NAU8824_ADC_SYNC_DOWN_128 2 +#define NAU8824_ADC_SYNC_DOWN_256 3 + +/* DAC_FILTER_CTRL_1 (0x25) */ +#define NAU8824_DAC_CICCLP_OFF (0x1 << 7) +#define NAU8824_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8824_DAC_OVERSAMPLE_64 0 +#define NAU8824_DAC_OVERSAMPLE_256 1 +#define NAU8824_DAC_OVERSAMPLE_128 2 +#define NAU8824_DAC_OVERSAMPLE_32 4 + +/* DAC_MUTE_CTRL (0x31) */ +#define NAU8824_DAC_CH01_MIX 0x3 +#define NAU8824_DAC_ZC_EN (0x1 << 11) + +/* DAC_CH0_DGAIN_CTRL (0x32) */ +#define NAU8824_DAC_CH0_SEL_SFT 9 +#define NAU8824_DAC_CH0_SEL_MASK (0x1 << NAU8824_DAC_CH0_SEL_SFT) +#define NAU8824_DAC_CH0_SEL_I2S0 (0x0 << NAU8824_DAC_CH0_SEL_SFT) +#define NAU8824_DAC_CH0_SEL_I2S1 (0x1 << NAU8824_DAC_CH0_SEL_SFT) +#define NAU8824_DAC_CH0_VOL_MASK 0x1ff + +/* DAC_CH1_DGAIN_CTRL (0x33) */ +#define NAU8824_DAC_CH1_SEL_SFT 9 +#define NAU8824_DAC_CH1_SEL_MASK (0x1 << NAU8824_DAC_CH1_SEL_SFT) +#define NAU8824_DAC_CH1_SEL_I2S0 (0x0 << NAU8824_DAC_CH1_SEL_SFT) +#define NAU8824_DAC_CH1_SEL_I2S1 (0x1 << NAU8824_DAC_CH1_SEL_SFT) +#define NAU8824_DAC_CH1_VOL_MASK 0x1ff + +/* CLASSG (0x50) */ +#define NAU8824_CLASSG_TIMER_SFT 8 +#define NAU8824_CLASSG_TIMER_MASK (0x3f << NAU8824_CLASSG_TIMER_SFT) +#define NAU8824_CLASSG_LDAC_EN_SFT 2 +#define NAU8824_CLASSG_RDAC_EN_SFT 1 +#define NAU8824_CLASSG_EN_SFT 0 + +/* SAR_ADC_DATA_OUT (0x59) */ +#define NAU8824_SAR_ADC_DATA_MASK 0xff + +/* BIAS_ADJ (0x66) */ +#define NAU8824_VMID (1 << 6) +#define NAU8824_VMID_SEL_SFT 4 +#define NAU8824_VMID_SEL_MASK (3 << NAU8824_VMID_SEL_SFT) +#define NAU8824_DMIC2_EN_SFT 3 +#define NAU8824_DMIC1_EN_SFT 2 + +/* TRIM_SETTINGS (0x68) */ +#define NAU8824_DRV_CURR_INC (1 << 15) + +/* ANALOG_CONTROL_1 (0x69) */ +#define NAU8824_DMIC_CLK_DRV_STRG (1 << 3) +#define NAU8824_DMIC_CLK_SLEW_FAST (0x7) + +/* ANALOG_CONTROL_2 (0x6A) */ +#define NAU8824_CLASSD_CLAMP_DIS_SFT 3 +#define NAU8824_CLASSD_CLAMP_DIS (0x1 << NAU8824_CLASSD_CLAMP_DIS_SFT) + +/* ENABLE_LO (0x6B) */ +#define NAU8824_TEST_DAC_SFT 14 +#define NAU8824_TEST_DAC_EN (0x3 << NAU8824_TEST_DAC_SFT) +#define NAU8824_DACL_HPR_EN_SFT 3 +#define NAU8824_DACL_HPR_EN (0x1 << NAU8824_DACL_HPR_EN_SFT) +#define NAU8824_DACR_HPR_EN_SFT 2 +#define NAU8824_DACR_HPR_EN (0x1 << NAU8824_DACR_HPR_EN_SFT) +#define NAU8824_DACR_HPL_EN_SFT 1 +#define NAU8824_DACR_HPL_EN (0x1 << NAU8824_DACR_HPL_EN_SFT) +#define NAU8824_DACL_HPL_EN_SFT 0 +#define NAU8824_DACL_HPL_EN 0x1 + +/* CLASSD_GAIN_1 (0x6D) */ +#define NAU8824_CLASSD_GAIN_1R_SFT 8 +#define NAU8824_CLASSD_GAIN_1R_MASK (0x1f << NAU8824_CLASSD_GAIN_1R_SFT) +#define NAU8824_CLASSD_EN_SFT 7 +#define NAU8824_CLASSD_EN (0x1 << NAU8824_CLASSD_EN_SFT) +#define NAU8824_CLASSD_GAIN_1L_MASK 0x1f + +/* CLASSD_GAIN_2 (0x6E) */ +#define NAU8824_CLASSD_GAIN_2R_SFT 8 +#define NAU8824_CLASSD_GAIN_2R_MASK (0x1f << NAU8824_CLASSD_GAIN_1R_SFT) +#define NAU8824_CLASSD_EN_SFT 7 +#define NAU8824_CLASSD_EN (0x1 << NAU8824_CLASSD_EN_SFT) +#define NAU8824_CLASSD_GAIN_2L_MASK 0x1f + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8824_ADCR_EN_SFT 7 +#define NAU8824_ADCL_EN_SFT 6 + +/* RDAC (0x73) */ +#define NAU8824_DACR_EN_SFT 13 +#define NAU8824_DACL_EN_SFT 12 +#define NAU8824_DACR_CLK_SFT 9 +#define NAU8824_DACL_CLK_SFT 8 +#define NAU8824_RDAC_CLK_DELAY_SFT 4 +#define NAU8824_RDAC_CLK_DELAY_MASK (0x7 << NAU8824_RDAC_CLK_DELAY_SFT) +#define NAU8824_RDAC_VREF_SFT 2 +#define NAU8824_RDAC_VREF_MASK (0x3 << NAU8824_RDAC_VREF_SFT) + +/* MIC_BIAS (0x74) */ +#define NAU8824_MICBIAS_JKSLV (1 << 14) +#define NAU8824_MICBIAS_JKR2 (1 << 12) +#define NAU8824_MICBIAS_POWERUP_SFT 8 +#define NAU8824_MICBIAS_VOLTAGE_SFT 0 +#define NAU8824_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8824_PRECHARGE_DIS (0x1 << 13) +#define NAU8824_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8824_HP_BOOST_DIS_SFT 9 +#define NAU8824_HP_BOOST_DIS (0x1 << NAU8824_HP_BOOST_DIS_SFT) +#define NAU8824_HP_BOOST_G_DIS_SFT 8 +#define NAU8824_HP_BOOST_G_DIS (0x1 << NAU8824_HP_BOOST_G_DIS_SFT) +#define NAU8824_SHORT_SHUTDOWN_DIG_EN (1 << 7) +#define NAU8824_SHORT_SHUTDOWN_EN (1 << 6) + +/* FEPGA (0x77) */ +#define NAU8824_FEPGA_MODER_SHORT_SFT 7 +#define NAU8824_FEPGA_MODER_SHORT_EN (0x1 << NAU8824_FEPGA_MODER_SHORT_SFT) +#define NAU8824_FEPGA_MODER_MIC2_SFT 5 +#define NAU8824_FEPGA_MODER_MIC2_EN (0x1 << NAU8824_FEPGA_MODER_MIC2_SFT) +#define NAU8824_FEPGA_MODER_HSMIC_SFT 4 +#define NAU8824_FEPGA_MODER_HSMIC_EN (0x1 << NAU8824_FEPGA_MODER_HSMIC_SFT) +#define NAU8824_FEPGA_MODEL_SHORT_SFT 3 +#define NAU8824_FEPGA_MODEL_SHORT_EN (0x1 << NAU8824_FEPGA_MODEL_SHORT_SFT) +#define NAU8824_FEPGA_MODEL_MIC1_SFT 1 +#define NAU8824_FEPGA_MODEL_MIC1_EN (0x1 << NAU8824_FEPGA_MODEL_MIC1_SFT) +#define NAU8824_FEPGA_MODEL_HSMIC_SFT 0 +#define NAU8824_FEPGA_MODEL_HSMIC_EN (0x1 << NAU8824_FEPGA_MODEL_HSMIC_SFT) + +/* FEPGA_II (0x78) */ +#define NAU8824_FEPGA_GAINR_SFT 5 +#define NAU8824_FEPGA_GAINR_MASK (0x1f << NAU8824_FEPGA_GAINR_SFT) +#define NAU8824_FEPGA_GAINL_SFT 0 +#define NAU8824_FEPGA_GAINL_MASK 0x1f + +/* CHARGE_PUMP_CONTROL (0x80) */ +#define NAU8824_JAMNODCLOW (0x1 << 15) +#define NAU8824_SPKR_PULL_DOWN (0x1 << 13) +#define NAU8824_SPKL_PULL_DOWN (0x1 << 12) +#define NAU8824_POWER_DOWN_DACR (0x1 << 9) +#define NAU8824_POWER_DOWN_DACL (0x1 << 8) +#define NAU8824_CHARGE_PUMP_EN_SFT 5 +#define NAU8824_CHARGE_PUMP_EN (0x1 << NAU8824_CHARGE_PUMP_EN_SFT) + + +#define NAU8824_CODEC_DAI "nau8824-hifi" + +/* System Clock Source */ +enum { + NAU8824_CLK_DIS, + NAU8824_CLK_MCLK, + NAU8824_CLK_INTERNAL, + NAU8824_CLK_FLL_MCLK, + NAU8824_CLK_FLL_BLK, + NAU8824_CLK_FLL_FS, +}; + +struct nau8824 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct work_struct jdet_work; + struct semaphore jd_sem; + int fs; + int irq; + int micbias_voltage; + int vref_impedance; + int jkdet_polarity; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int sar_compare_time; + int sar_sampling_time; + int key_debounce; + int jack_eject_debounce; +}; + +struct nau8824_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau8824_fll_attr { + unsigned int param; + unsigned int val; +}; + +struct nau8824_osr_attr { + unsigned int osr; + unsigned int clk_src; +}; + + +int nau8824_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif /* _NAU8824_H */ + -- 2.6.4

2 2

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Add loadable module support for
by Subhransu S. Prusty 26 Apr '17

26 Apr '17

Module loading (which are not part of base firmware) is different for kbl platform compared to skl or bxtn. In kabylake library manifest contains the module information, so load the library to get the module information. As a pre-work, common code which can be used for kabylake are moved to helper and arguments to few functions are modified to support module load for kabylake. Separate dsp_fw_ops is registered and load library ops is added for kabylake. G Kranthi (2): ASoC: Intel: Skylake: Move sst common initialization to a helper function ASoC: Intel: Skylake: Modify arguments to reuse module transfer function Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add loadable module support on KBL platform Subhransu S. Prusty (3): ASoC: Intel: Skylake: Commonize library load ASoC: Intel: Skylake: Register dsp_fw_ops for kabylake ASoC: Intel: Skylake: Modify load_lib_ipc arguments for a nowait version sound/soc/intel/skylake/bxt-sst.c | 68 +++--------------- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-sst-dsp.h | 13 +++- sound/soc/intel/skylake/skl-sst-ipc.c | 10 ++- sound/soc/intel/skylake/skl-sst-ipc.h | 2 +- sound/soc/intel/skylake/skl-sst-utils.c | 80 +++++++++++++++++++++ sound/soc/intel/skylake/skl-sst.c | 124 ++++++++++++++++++++++++-------- 7 files changed, 207 insertions(+), 92 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Add loadable module support for Kabylake
by Subhransu S. Prusty 26 Apr '17

26 Apr '17

KBL module load is a bit different than rest as Library manifest containing module information is required to be loaded first before loading module. So common changes are moved to helpers and arguments are modified as a pre-work for adding module loading support. Also, separate dsp_fw_ops is registered and load library for kabylake is added to use code loader dma for library and module load. changes in v2: - Few minor fixes in comments - Fixed the cover letter subject G Kranthi (2): ASoC: Intel: Skylake: Move sst common initialization to a helper function ASoC: Intel: Skylake: Modify arguments to reuse module transfer function Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add loadable module support on KBL platform Subhransu S. Prusty (3): ASoC: Intel: Skylake: Commonize library load ASoC: Intel: Skylake: Register dsp_fw_ops for kabylake ASoC: Intel: Skylake: Modify load_lib_ipc arguments for a nowait version sound/soc/intel/skylake/bxt-sst.c | 68 +++-------------- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-sst-dsp.h | 13 +++- sound/soc/intel/skylake/skl-sst-ipc.c | 10 ++- sound/soc/intel/skylake/skl-sst-ipc.h | 2 +- sound/soc/intel/skylake/skl-sst-utils.c | 80 ++++++++++++++++++++ sound/soc/intel/skylake/skl-sst.c | 125 ++++++++++++++++++++++++-------- 7 files changed, 208 insertions(+), 92 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH] ASoC: hdmi-codec: Use devm_kcalloc() in hdmi_codec_probe()
by SF Markus Elfring 26 Apr '17

26 Apr '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 26 Apr 2017 12:26:06 +0200 A multiplication for the size determination of a memory allocation indicated that an array data structure should be processed. Thus use the corresponding function "devm_kcalloc". This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/hdmi-codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..99933488fa03 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -825,8 +825,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) hcp->hcd = *hcd; mutex_init(&hcp->current_stream_lock); - - hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv), + hcp->daidrv = devm_kcalloc(dev, dai_count, sizeof(*hcp->daidrv), GFP_KERNEL); if (!hcp->daidrv) return -ENOMEM; -- 2.12.2

1 0

[alsa-devel] ALSA / Soundcard latency
by Feilmeier Daniel (Ask De) 26 Apr '17

26 Apr '17

Hello, My company is a supplier of car manufacturers. We develop and produce digital audio amplifiers. We would like to build our test setup for echo order cancelation routines for our audio amplifiers with a real- time Linux and ALSA. We have very strong requirements regarding the maximum latency from audio input via line in to audio output. We would like to achieve a maximum latency < 1ms. The alsaloop example mentions a latency of 1 ms which is on our test setup with the latest rt kernel from This Mächler not working due to broken pipe errors. Is 1 ms feasible with a standard PC ? Currently we still don´t know what the lowest possible latency using ALSA is. We tried different soundcards and found out that they have an internal FIFO, which seems to be one of the limiting factors. The min. latency we can achieve is currently 5,6 ms, but this is still to high. Is there any possibility to switch this FIFO off via ALSA ? Is it possible at all to achieve such low latency values ? Kind regards, ___________________________________________________________________________________ Feilmeier Daniel R&D Software Electronics Tel.: +49-9962-2003-629 Fax.: +49-9962-2003-79 E-Mail: FeilmeierD(a)askgroup.de<mailto:FeilmeierD@askgroup.de> ASK Industries GmbH Hauptstrasse 73 D-94559 Niederwinkling HRG Straubing HRB 10703 Geschaeftsfuehrer: Ruggero Marchetti Franz Maier Juergen Pledl ___________________________________________________________________________________

2 1

[alsa-devel] [PATCH v4] ASoC: Add support for Maxim Integrated MAX98927 Amplifier
by Ryan Lee 26 Apr '17

26 Apr '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v4: * Removed support for SND_SOC_DAIFMT_CBS_CFM. * Fixed coding style for indention. * Removed variables if it has only one user. * Assigned ch_size directly. * Removed oring. * Put the return false in the switch statement. * Removed 'Monomix Output' and 'Speaker Source' controls. * Modified control names per control-names.rst. * Moved Revision ID check code to i2c_probe function. * Added 'Current Limit' control. * Removed 'devm__kfree' function. Changes since v3: * Combined MAX98926 and MAX98927 binding. Kept existing property name. Changes since v2: * Removed local register read/write function to avoid duplication of ASoC core function. .../devicetree/bindings/sound/max98925.txt | 22 - .../devicetree/bindings/sound/max98926.txt | 32 - .../devicetree/bindings/sound/max9892x.txt | 41 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98927.c | 841 +++++++++++++++++++++ sound/soc/codecs/max98927.h | 272 +++++++ 7 files changed, 1161 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max98925.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100755 Documentation/devicetree/bindings/sound/max9892x.txt mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/max98927.c create mode 100755 sound/soc/codecs/max98927.h diff --git a/Documentation/devicetree/bindings/sound/max98925.txt b/Documentation/devicetree/bindings/sound/max98925.txt deleted file mode 100644 index 27be63e..0000000 --- a/Documentation/devicetree/bindings/sound/max98925.txt +++ /dev/null @@ -1,22 +0,0 @@ -max98925 audio CODEC - -This device supports I2C. - -Required properties: - - - compatible : "maxim,max98925" - - - vmon-slot-no : slot number used to send voltage information - - - imon-slot-no : slot number used to send current information - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98925@1a { - compatible = "maxim,max98925"; - vmon-slot-no = <0>; - imon-slot-no = <2>; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt deleted file mode 100644 index 0b7f4e4..0000000 --- a/Documentation/devicetree/bindings/sound/max98926.txt +++ /dev/null @@ -1,32 +0,0 @@ -max98926 audio CODEC - -This device supports I2C. - -Required properties: - - - compatible : "maxim,max98926" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - - - imon-slot-no : slot number used to send current information - - - interleave-mode : When using two MAX98926 in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - - - reg : the I2C address of the device for I2C - -Example: - -codec: max98926@1a { - compatible = "maxim,max98926"; - vmon-slot-no = <0>; - imon-slot-no = <2>; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt new file mode 100755 index 0000000..f617159 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -0,0 +1,41 @@ +Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : should be one of the following + - "maxim,max98925" + - "maxim,max98926" + - "maxim,max98927" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - imon-slot-no : slot number used to send current information + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - interleave-mode : When using two MAX9892X in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + Range : 0 (off), 1 (on), Default : 0 + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98927@3a { + compatible = "maxim,max98927"; + vmon-slot-no = <0>; + imon-slot-no = <1>; + interleave-mode = <0>; + reg = <0x3a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index e49e9da..cbdc8aa --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -89,6 +89,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C + select SND_SOC_MAX98927 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C @@ -588,6 +589,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX98926 tristate +config SND_SOC_MAX98927 + tristate "Maxim Integrated MAX98927 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 1796cb9..187a639 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o +snd-soc-max98927-objs := max98927.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o @@ -313,6 +314,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o +obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c new file mode 100755 index 0000000..b5ee294 --- /dev/null +++ b/sound/soc/codecs/max98927.c @@ -0,0 +1,841 @@ +/* + * max98927.c -- MAX98927 ALSA Soc Audio driver + * + * Copyright (C) 2016 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/acpi.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98927.h" + +static struct reg_default max98927_reg[] = { + {MAX98927_R0001_INT_RAW1, 0x00}, + {MAX98927_R0002_INT_RAW2, 0x00}, + {MAX98927_R0003_INT_RAW3, 0x00}, + {MAX98927_R0004_INT_STATE1, 0x00}, + {MAX98927_R0005_INT_STATE2, 0x00}, + {MAX98927_R0006_INT_STATE3, 0x00}, + {MAX98927_R0007_INT_FLAG1, 0x00}, + {MAX98927_R0008_INT_FLAG2, 0x00}, + {MAX98927_R0009_INT_FLAG3, 0x00}, + {MAX98927_R000A_INT_EN1, 0x00}, + {MAX98927_R000B_INT_EN2, 0x00}, + {MAX98927_R000C_INT_EN3, 0x00}, + {MAX98927_R000D_INT_FLAG_CLR1, 0x00}, + {MAX98927_R000E_INT_FLAG_CLR2, 0x00}, + {MAX98927_R000F_INT_FLAG_CLR3, 0x00}, + {MAX98927_R0010_IRQ_CTRL, 0x00}, + {MAX98927_R0011_CLK_MON, 0x00}, + {MAX98927_R0012_WDOG_CTRL, 0x00}, + {MAX98927_R0013_WDOG_RST, 0x00}, + {MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH, 0x00}, + {MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH, 0x00}, + {MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS, 0x00}, + {MAX98927_R0017_PIN_CFG, 0x55}, + {MAX98927_R0018_PCM_RX_EN_A, 0x00}, + {MAX98927_R0019_PCM_RX_EN_B, 0x00}, + {MAX98927_R001A_PCM_TX_EN_A, 0x00}, + {MAX98927_R001B_PCM_TX_EN_B, 0x00}, + {MAX98927_R001C_PCM_TX_HIZ_CTRL_A, 0x00}, + {MAX98927_R001D_PCM_TX_HIZ_CTRL_B, 0x00}, + {MAX98927_R001E_PCM_TX_CH_SRC_A, 0x00}, + {MAX98927_R001F_PCM_TX_CH_SRC_B, 0x00}, + {MAX98927_R0020_PCM_MODE_CFG, 0x40}, + {MAX98927_R0021_PCM_MASTER_MODE, 0x00}, + {MAX98927_R0022_PCM_CLK_SETUP, 0x22}, + {MAX98927_R0023_PCM_SR_SETUP1, 0x00}, + {MAX98927_R0024_PCM_SR_SETUP2, 0x00}, + {MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, 0x00}, + {MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, 0x00}, + {MAX98927_R0027_ICC_RX_EN_A, 0x00}, + {MAX98927_R0028_ICC_RX_EN_B, 0x00}, + {MAX98927_R002B_ICC_TX_EN_A, 0x00}, + {MAX98927_R002C_ICC_TX_EN_B, 0x00}, + {MAX98927_R002E_ICC_HIZ_MANUAL_MODE, 0x00}, + {MAX98927_R002F_ICC_TX_HIZ_EN_A, 0x00}, + {MAX98927_R0030_ICC_TX_HIZ_EN_B, 0x00}, + {MAX98927_R0031_ICC_LNK_EN, 0x00}, + {MAX98927_R0032_PDM_TX_EN, 0x00}, + {MAX98927_R0033_PDM_TX_HIZ_CTRL, 0x00}, + {MAX98927_R0034_PDM_TX_CTRL, 0x00}, + {MAX98927_R0035_PDM_RX_CTRL, 0x00}, + {MAX98927_R0036_AMP_VOL_CTRL, 0x00}, + {MAX98927_R0037_AMP_DSP_CFG, 0x02}, + {MAX98927_R0038_TONE_GEN_DC_CFG, 0x00}, + {MAX98927_R0039_DRE_CTRL, 0x01}, + {MAX98927_R003A_AMP_EN, 0x00}, + {MAX98927_R003B_SPK_SRC_SEL, 0x00}, + {MAX98927_R003C_SPK_GAIN, 0x00}, + {MAX98927_R003D_SSM_CFG, 0x01}, + {MAX98927_R003E_MEAS_EN, 0x00}, + {MAX98927_R003F_MEAS_DSP_CFG, 0x04}, + {MAX98927_R0040_BOOST_CTRL0, 0x00}, + {MAX98927_R0041_BOOST_CTRL3, 0x00}, + {MAX98927_R0042_BOOST_CTRL1, 0x00}, + {MAX98927_R0043_MEAS_ADC_CFG, 0x00}, + {MAX98927_R0044_MEAS_ADC_BASE_MSB, 0x00}, + {MAX98927_R0045_MEAS_ADC_BASE_LSB, 0x00}, + {MAX98927_R0046_ADC_CH0_DIVIDE, 0x00}, + {MAX98927_R0047_ADC_CH1_DIVIDE, 0x00}, + {MAX98927_R0048_ADC_CH2_DIVIDE, 0x00}, + {MAX98927_R0049_ADC_CH0_FILT_CFG, 0x00}, + {MAX98927_R004A_ADC_CH1_FILT_CFG, 0x00}, + {MAX98927_R004B_ADC_CH2_FILT_CFG, 0x00}, + {MAX98927_R004C_MEAS_ADC_CH0_READ, 0x00}, + {MAX98927_R004D_MEAS_ADC_CH1_READ, 0x00}, + {MAX98927_R004E_MEAS_ADC_CH2_READ, 0x00}, + {MAX98927_R0051_BROWNOUT_STATUS, 0x00}, + {MAX98927_R0052_BROWNOUT_EN, 0x00}, + {MAX98927_R0053_BROWNOUT_INFINITE_HOLD, 0x00}, + {MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR, 0x00}, + {MAX98927_R0055_BROWNOUT_LVL_HOLD, 0x00}, + {MAX98927_R005A_BROWNOUT_LVL1_THRESH, 0x00}, + {MAX98927_R005B_BROWNOUT_LVL2_THRESH, 0x00}, + {MAX98927_R005C_BROWNOUT_LVL3_THRESH, 0x00}, + {MAX98927_R005D_BROWNOUT_LVL4_THRESH, 0x00}, + {MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS, 0x00}, + {MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL, 0x00}, + {MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL, 0x00}, + {MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE, 0x00}, + {MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT, 0x00}, + {MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1, 0x00}, + {MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2, 0x00}, + {MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3, 0x00}, + {MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT, 0x00}, + {MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1, 0x00}, + {MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2, 0x00}, + {MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3, 0x00}, + {MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT, 0x00}, + {MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1, 0x00}, + {MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2, 0x00}, + {MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3, 0x00}, + {MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT, 0x00}, + {MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, 0x00}, + {MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2, 0x00}, + {MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3, 0x00}, + {MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0x00}, + {MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY, 0x00}, + {MAX98927_R0084_ENV_TRACK_REL_RATE, 0x00}, + {MAX98927_R0085_ENV_TRACK_HOLD_RATE, 0x00}, + {MAX98927_R0086_ENV_TRACK_CTRL, 0x00}, + {MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, 0x00}, + {MAX98927_R00FF_GLOBAL_SHDN, 0x00}, + {MAX98927_R0100_SOFT_RESET, 0x00}, + {MAX98927_R01FF_REV_ID, 0x40}, +}; + +static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int mode = 0; + unsigned int format = 0; + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + mode = MAX98927_PCM_MASTER_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98927->master = true; + mode = MAX98927_PCM_MASTER_MODE_MASTER; + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MASK, + mode); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + max98927->iface |= SND_SOC_DAIFMT_I2S; + format = MAX98927_PCM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + max98927->iface |= SND_SOC_DAIFMT_LEFT_J; + format = MAX98927_PCM_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_PDM: + max98927->iface |= SND_SOC_DAIFMT_PDM; + break; + default: + return -EINVAL; + } + + /* pcm channel configuration */ + if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + regmap_update_bits(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN); + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_FORMAT_MASK, + format << MAX98927_PCM_MODE_CFG_FORMAT_SHIFT); + + regmap_update_bits(max98927->regmap, + MAX98927_R003B_SPK_SRC_SEL, + MAX98927_SPK_SRC_MASK, 0); + + } else + regmap_update_bits(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + /* pdm channel configuration */ + if (max98927->iface & SND_SOC_DAIFMT_PDM) { + regmap_update_bits(max98927->regmap, + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 1); + + regmap_update_bits(max98927->regmap, + MAX98927_R003B_SPK_SRC_SEL, + MAX98927_SPK_SRC_MASK, 3); + } else + regmap_update_bits(max98927->regmap, + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + return 0; +} + +/* codec MCLK rate in master mode */ +static const int rate_table[] = { + 5644800, 6000000, 6144000, 6500000, + 9600000, 11289600, 12000000, 12288000, + 13000000, 19200000, +}; + +static int max98927_set_clock(struct max98927_priv *max98927, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_codec *codec = max98927->codec; + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98927->ch_size; + int value; + + if (max98927->master) { + int i; + /* match rate to closest value */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i] >= max98927->sysclk) + break; + } + if (i == ARRAY_SIZE(rate_table)) { + dev_err(codec->dev, "failed to find proper clock rate.\n"); + return -EINVAL; + } + regmap_update_bits(max98927->regmap, + MAX98927_R0021_PCM_MASTER_MODE, + MAX98927_PCM_MASTER_MODE_MCLK_MASK, + i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); + } + + switch (blr_clk_ratio) { + case 32: + value = 2; + break; + case 48: + value = 3; + break; + case 64: + value = 4; + break; + default: + return -EINVAL; + } + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); + return 0; +} + +static int max98927_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d", + params_format(params)); + goto err; + } + + max98927->ch_size = snd_pcm_format_width(params_format(params)); + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(codec->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_8000; + break; + case 11025: + sampling_rate = MAX98927_PCM_SR_SET1_SR_11025; + break; + case 12000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_12000; + break; + case 16000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_16000; + break; + case 22050: + sampling_rate = MAX98927_PCM_SR_SET1_SR_22050; + break; + case 24000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_24000; + break; + case 32000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_32000; + break; + case 44100: + sampling_rate = MAX98927_PCM_SR_SET1_SR_44100; + break; + case 48000: + sampling_rate = MAX98927_PCM_SR_SET1_SR_48000; + break; + default: + dev_err(codec->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98927->regmap, + MAX98927_R0023_PCM_SR_SETUP1, + MAX98927_PCM_SR_SET1_SR_MASK, + sampling_rate); + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_SR_MASK, + sampling_rate << MAX98927_PCM_SR_SET2_SR_SHIFT); + + /* set sampling rate of IV */ + if (max98927->interleave_mode && + sampling_rate > MAX98927_PCM_SR_SET1_SR_16000) + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate - 3); + else + regmap_update_bits(max98927->regmap, + MAX98927_R0024_PCM_SR_SETUP2, + MAX98927_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate); + return max98927_set_clock(max98927, params); +err: + return -EINVAL; +} + +#define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->sysclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops max98927_dai_ops = { + .set_sysclk = max98927_dai_set_sysclk, + .set_fmt = max98927_dai_set_fmt, + .hw_params = max98927_dai_hw_params, +}; + +static int max98927_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 1); + /* enable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN); + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 1); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98927->regmap, + MAX98927_R00FF_GLOBAL_SHDN, + MAX98927_GLOBAL_EN_MASK, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R003A_AMP_EN, + MAX98927_AMP_EN_MASK, 0); + /* disable VMON and IMON */ + regmap_update_bits(max98927->regmap, + MAX98927_R003E_MEAS_EN, + MAX98927_MEAS_V_EN | MAX98927_MEAS_I_EN, 0); + break; + default: + return 0; + } + return 0; +} + +static const char * const max98927_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT, + 3, max98927_switch_text); + +static const struct snd_kcontrol_new max98927_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_soc_dapm_widget max98927_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98927_R003A_AMP_EN, + 0, 0, max98927_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, + &max98927_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static DECLARE_TLV_DB_SCALE(max98927_spk_tlv, 300, 300, 0); +static DECLARE_TLV_DB_SCALE(max98927_digital_tlv, -1600, 25, 0); + +static bool max98927_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0028_ICC_RX_EN_B: + case MAX98927_R002B_ICC_TX_EN_A ... MAX98927_R002C_ICC_TX_EN_B: + case MAX98927_R002E_ICC_HIZ_MANUAL_MODE + ... MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS + ... MAX98927_R0055_BROWNOUT_LVL_HOLD: + case MAX98927_R005A_BROWNOUT_LVL1_THRESH + ... MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE: + case MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT + ... MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R00FF_GLOBAL_SHDN: + case MAX98927_R0100_SOFT_RESET: + case MAX98927_R01FF_REV_ID: + return true; + default: + return false; + } +}; + +static bool max98927_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + return true; + default: + return false; + } +} + +static const char * const max98927_boost_voltage_text[] = { + "6.5V", "6.625V", "6.75V", "6.875V", "7V", "7.125V", "7.25V", "7.375V", + "7.5V", "7.625V", "7.75V", "7.875V", "8V", "8.125V", "8.25V", "8.375V", + "8.5V", "8.625V", "8.75V", "8.875V", "9V", "9.125V", "9.25V", "9.375V", + "9.5V", "9.625V", "9.75V", "9.875V", "10V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_boost_voltage, + MAX98927_R0040_BOOST_CTRL0, 0, + max98927_boost_voltage_text); + +static const char * const max98927_current_limit_text[] = { + "1.00A", "1.10A", "1.20A", "1.30A", "1.40A", "1.50A", "1.60A", "1.70A", + "1.80A", "1.90A", "2.00A", "2.10A", "2.20A", "2.30A", "2.40A", "2.50A", + "2.60A", "2.70A", "2.80A", "2.90A", "3.00A", "3.10A", "3.20A", "3.30A", + "3.40A", "3.50A", "3.60A", "3.70A", "3.80A", "3.90A", "4.00A", "4.10A" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_current_limit, + MAX98927_R0042_BOOST_CTRL1, 1, + max98927_current_limit_text); + +static const struct snd_kcontrol_new max98927_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, + 0, 6, 0, + max98927_spk_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98927_R0036_AMP_VOL_CTRL, + 0, (1<<MAX98927_AMP_VOL_WIDTH)-1, 0, + max98927_digital_tlv), + SOC_SINGLE("Amp DSP Switch", MAX98927_R0052_BROWNOUT_EN, + MAX98927_BROWNOUT_DSP_SHIFT, 1, 0), + SOC_SINGLE("Ramp Switch", MAX98927_R0037_AMP_DSP_CFG, + MAX98927_AMP_DSP_CFG_RMP_SHIFT, 1, 0), + SOC_SINGLE("DRE Switch", MAX98927_R0039_DRE_CTRL, + MAX98927_DRE_EN_SHIFT, 1, 0), + SOC_SINGLE("Volume Location Switch", MAX98927_R0036_AMP_VOL_CTRL, + MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), + SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), + SOC_ENUM("Current Limit", max98927_current_limit), +}; + +static const struct snd_soc_dapm_route max98927_audio_map[] = { + {"Amp Enable", NULL, "DAI_OUT"}, + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98927_dai[] = { + { + .name = "max98927-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .ops = &max98927_dai_ops, + } +}; + +static int max98927_probe(struct snd_soc_codec *codec) +{ + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->codec = codec; + codec->control_data = max98927->regmap; + codec->cache_bypass = 1; + + /* Software Reset */ + regmap_write(max98927->regmap, + MAX98927_R0100_SOFT_RESET, MAX98927_SOFT_RESET); + + /* IV default slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0025_PCM_TO_SPK_MONOMIX_A, + 0x80); + regmap_write(max98927->regmap, + MAX98927_R0026_PCM_TO_SPK_MONOMIX_B, + 0x1); + /* Set inital volume (+13dB) */ + regmap_write(max98927->regmap, + MAX98927_R0036_AMP_VOL_CTRL, + 0x38); + regmap_write(max98927->regmap, + MAX98927_R003C_SPK_GAIN, + 0x05); + /* Enable DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R0037_AMP_DSP_CFG, + 0x03); + /* Enable IMON VMON DC blocker */ + regmap_write(max98927->regmap, + MAX98927_R003F_MEAS_DSP_CFG, + 0xF7); + /* Boost Output Voltage & Current limit */ + regmap_write(max98927->regmap, + MAX98927_R0040_BOOST_CTRL0, + 0x1C); + regmap_write(max98927->regmap, + MAX98927_R0042_BOOST_CTRL1, + 0x3E); + /* Measurement ADC config */ + regmap_write(max98927->regmap, + MAX98927_R0043_MEAS_ADC_CFG, + 0x04); + regmap_write(max98927->regmap, + MAX98927_R0044_MEAS_ADC_BASE_MSB, + 0x00); + regmap_write(max98927->regmap, + MAX98927_R0045_MEAS_ADC_BASE_LSB, + 0x24); + /* Brownout Level */ + regmap_write(max98927->regmap, + MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1, + 0x06); + /* Envelope Tracking configuration */ + regmap_write(max98927->regmap, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, + 0x08); + regmap_write(max98927->regmap, + MAX98927_R0086_ENV_TRACK_CTRL, + 0x01); + regmap_write(max98927->regmap, + MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, + 0x10); + + /* voltage, current slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001E_PCM_TX_CH_SRC_A, + (max98927->i_l_slot<<MAX98927_PCM_TX_CH_SRC_A_I_SHIFT| + max98927->v_l_slot)&0xFF); + + if (max98927->v_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->v_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->v_l_slot, + 1 << max98927->v_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->v_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->v_l_slot - 8), + 1 << (max98927->v_l_slot - 8)); + } + + if (max98927->i_l_slot < 8) { + regmap_update_bits(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + 1 << max98927->i_l_slot, 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + 1 << max98927->i_l_slot, + 1 << max98927->i_l_slot); + } else { + regmap_update_bits(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + 1 << (max98927->i_l_slot - 8), 0); + regmap_update_bits(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + 1 << (max98927->i_l_slot - 8), + 1 << (max98927->i_l_slot - 8)); + } + + /* Set interleave mode */ + if (max98927->interleave_mode) + regmap_update_bits(max98927->regmap, + MAX98927_R001F_PCM_TX_CH_SRC_B, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK, + MAX98927_PCM_TX_CH_INTERLEAVE_MASK); + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_dev_max98927 = { + .probe = max98927_probe, + .component_driver = { + .controls = max98927_snd_controls, + .num_controls = ARRAY_SIZE(max98927_snd_controls), + .dapm_widgets = max98927_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98927_dapm_widgets), + .dapm_routes = max98927_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98927_audio_map), + }, +}; + +static const struct regmap_config max98927_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98927_R01FF_REV_ID, + .reg_defaults = max98927_reg, + .num_reg_defaults = ARRAY_SIZE(max98927_reg), + .readable_reg = max98927_readable_register, + .volatile_reg = max98927_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static void max98927_slot_config(struct i2c_client *i2c, + struct max98927_priv *max98927) +{ + int value; + + if (!of_property_read_u32(i2c->dev.of_node, + "vmon-slot-no", &value)) + max98927->v_l_slot = value & 0xF; + else + max98927->v_l_slot = 0; + if (!of_property_read_u32(i2c->dev.of_node, + "imon-slot-no", &value)) + max98927->i_l_slot = value & 0xF; + else + max98927->i_l_slot = 1; +} + +static int max98927_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + + int ret = 0, value; + int reg = 0; + struct max98927_priv *max98927 = NULL; + + max98927 = devm_kzalloc(&i2c->dev, + sizeof(*max98927), GFP_KERNEL); + + if (!max98927) { + ret = -ENOMEM; + return ret; + } + i2c_set_clientdata(i2c, max98927); + + /* update interleave mode info */ + if (!of_property_read_u32(i2c->dev.of_node, + "interleave_mode", &value)) { + if (value > 0) + max98927->interleave_mode = 1; + else + max98927->interleave_mode = 0; + } else + max98927->interleave_mode = 0; + + /* regmap initialization */ + max98927->regmap + = devm_regmap_init_i2c(i2c, &max98927_regmap); + if (IS_ERR(max98927->regmap)) { + ret = PTR_ERR(max98927->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + /* Check Revision ID */ + ret = regmap_read(max98927->regmap, + MAX98927_R01FF_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, + "Failed to read: 0x%02X\n", MAX98927_R01FF_REV_ID); + return ret; + } + dev_info(&i2c->dev, "MAX98927 revisionID: 0x%02X\n", reg); + + /* voltage/current slot configuration */ + max98927_slot_config(i2c, max98927); + + /* codec registeration */ + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98927, + max98927_dai, ARRAY_SIZE(max98927_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int max98927_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98927_i2c_id[] = { + { "max98927", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98927_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98927_of_match[] = { + { .compatible = "maxim,max98927", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98927_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98927_acpi_match[] = { + { "MX98927", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98927_acpi_match); +#endif + +static struct i2c_driver max98927_i2c_driver = { + .driver = { + .name = "max98927", + .of_match_table = of_match_ptr(max98927_of_match), + .acpi_match_table = ACPI_PTR(max98927_acpi_match), + .pm = NULL, + }, + .probe = max98927_i2c_probe, + .remove = max98927_i2c_remove, + .id_table = max98927_i2c_id, +}; + +module_i2c_driver(max98927_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98927 driver"); +MODULE_AUTHOR("Ryan Lee <ryans.lee(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h new file mode 100755 index 0000000..ece6a60 --- /dev/null +++ b/sound/soc/codecs/max98927.h @@ -0,0 +1,272 @@ +/* + * max98927.h -- MAX98927 ALSA Soc Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ +#ifndef _MAX98927_H +#define _MAX98927_H + +/* Register Values */ +#define MAX98927_R0001_INT_RAW1 0x0001 +#define MAX98927_R0002_INT_RAW2 0x0002 +#define MAX98927_R0003_INT_RAW3 0x0003 +#define MAX98927_R0004_INT_STATE1 0x0004 +#define MAX98927_R0005_INT_STATE2 0x0005 +#define MAX98927_R0006_INT_STATE3 0x0006 +#define MAX98927_R0007_INT_FLAG1 0x0007 +#define MAX98927_R0008_INT_FLAG2 0x0008 +#define MAX98927_R0009_INT_FLAG3 0x0009 +#define MAX98927_R000A_INT_EN1 0x000A +#define MAX98927_R000B_INT_EN2 0x000B +#define MAX98927_R000C_INT_EN3 0x000C +#define MAX98927_R000D_INT_FLAG_CLR1 0x000D +#define MAX98927_R000E_INT_FLAG_CLR2 0x000E +#define MAX98927_R000F_INT_FLAG_CLR3 0x000F +#define MAX98927_R0010_IRQ_CTRL 0x0010 +#define MAX98927_R0011_CLK_MON 0x0011 +#define MAX98927_R0012_WDOG_CTRL 0x0012 +#define MAX98927_R0013_WDOG_RST 0x0013 +#define MAX98927_R0014_MEAS_ADC_THERM_WARN_THRESH 0x0014 +#define MAX98927_R0015_MEAS_ADC_THERM_SHDN_THRESH 0x0015 +#define MAX98927_R0016_MEAS_ADC_THERM_HYSTERESIS 0x0016 +#define MAX98927_R0017_PIN_CFG 0x0017 +#define MAX98927_R0018_PCM_RX_EN_A 0x0018 +#define MAX98927_R0019_PCM_RX_EN_B 0x0019 +#define MAX98927_R001A_PCM_TX_EN_A 0x001A +#define MAX98927_R001B_PCM_TX_EN_B 0x001B +#define MAX98927_R001C_PCM_TX_HIZ_CTRL_A 0x001C +#define MAX98927_R001D_PCM_TX_HIZ_CTRL_B 0x001D +#define MAX98927_R001E_PCM_TX_CH_SRC_A 0x001E +#define MAX98927_R001F_PCM_TX_CH_SRC_B 0x001F +#define MAX98927_R0020_PCM_MODE_CFG 0x0020 +#define MAX98927_R0021_PCM_MASTER_MODE 0x0021 +#define MAX98927_R0022_PCM_CLK_SETUP 0x0022 +#define MAX98927_R0023_PCM_SR_SETUP1 0x0023 +#define MAX98927_R0024_PCM_SR_SETUP2 0x0024 +#define MAX98927_R0025_PCM_TO_SPK_MONOMIX_A 0x0025 +#define MAX98927_R0026_PCM_TO_SPK_MONOMIX_B 0x0026 +#define MAX98927_R0027_ICC_RX_EN_A 0x0027 +#define MAX98927_R0028_ICC_RX_EN_B 0x0028 +#define MAX98927_R002B_ICC_TX_EN_A 0x002B +#define MAX98927_R002C_ICC_TX_EN_B 0x002C +#define MAX98927_R002E_ICC_HIZ_MANUAL_MODE 0x002E +#define MAX98927_R002F_ICC_TX_HIZ_EN_A 0x002F +#define MAX98927_R0030_ICC_TX_HIZ_EN_B 0x0030 +#define MAX98927_R0031_ICC_LNK_EN 0x0031 +#define MAX98927_R0032_PDM_TX_EN 0x0032 +#define MAX98927_R0033_PDM_TX_HIZ_CTRL 0x0033 +#define MAX98927_R0034_PDM_TX_CTRL 0x0034 +#define MAX98927_R0035_PDM_RX_CTRL 0x0035 +#define MAX98927_R0036_AMP_VOL_CTRL 0x0036 +#define MAX98927_R0037_AMP_DSP_CFG 0x0037 +#define MAX98927_R0038_TONE_GEN_DC_CFG 0x0038 +#define MAX98927_R0039_DRE_CTRL 0x0039 +#define MAX98927_R003A_AMP_EN 0x003A +#define MAX98927_R003B_SPK_SRC_SEL 0x003B +#define MAX98927_R003C_SPK_GAIN 0x003C +#define MAX98927_R003D_SSM_CFG 0x003D +#define MAX98927_R003E_MEAS_EN 0x003E +#define MAX98927_R003F_MEAS_DSP_CFG 0x003F +#define MAX98927_R0040_BOOST_CTRL0 0x0040 +#define MAX98927_R0041_BOOST_CTRL3 0x0041 +#define MAX98927_R0042_BOOST_CTRL1 0x0042 +#define MAX98927_R0043_MEAS_ADC_CFG 0x0043 +#define MAX98927_R0044_MEAS_ADC_BASE_MSB 0x0044 +#define MAX98927_R0045_MEAS_ADC_BASE_LSB 0x0045 +#define MAX98927_R0046_ADC_CH0_DIVIDE 0x0046 +#define MAX98927_R0047_ADC_CH1_DIVIDE 0x0047 +#define MAX98927_R0048_ADC_CH2_DIVIDE 0x0048 +#define MAX98927_R0049_ADC_CH0_FILT_CFG 0x0049 +#define MAX98927_R004A_ADC_CH1_FILT_CFG 0x004A +#define MAX98927_R004B_ADC_CH2_FILT_CFG 0x004B +#define MAX98927_R004C_MEAS_ADC_CH0_READ 0x004C +#define MAX98927_R004D_MEAS_ADC_CH1_READ 0x004D +#define MAX98927_R004E_MEAS_ADC_CH2_READ 0x004E +#define MAX98927_R0051_BROWNOUT_STATUS 0x0051 +#define MAX98927_R0052_BROWNOUT_EN 0x0052 +#define MAX98927_R0053_BROWNOUT_INFINITE_HOLD 0x0053 +#define MAX98927_R0054_BROWNOUT_INFINITE_HOLD_CLR 0x0054 +#define MAX98927_R0055_BROWNOUT_LVL_HOLD 0x0055 +#define MAX98927_R005A_BROWNOUT_LVL1_THRESH 0x005A +#define MAX98927_R005B_BROWNOUT_LVL2_THRESH 0x005B +#define MAX98927_R005C_BROWNOUT_LVL3_THRESH 0x005C +#define MAX98927_R005D_BROWNOUT_LVL4_THRESH 0x005D +#define MAX98927_R005E_BROWNOUT_THRESH_HYSTERYSIS 0x005E +#define MAX98927_R005F_BROWNOUT_AMP_LIMITER_ATK_REL 0x005F +#define MAX98927_R0060_BROWNOUT_AMP_GAIN_ATK_REL 0x0060 +#define MAX98927_R0061_BROWNOUT_AMP1_CLIP_MODE 0x0061 +#define MAX98927_R0072_BROWNOUT_LVL1_CUR_LIMIT 0x0072 +#define MAX98927_R0073_BROWNOUT_LVL1_AMP1_CTRL1 0x0073 +#define MAX98927_R0074_BROWNOUT_LVL1_AMP1_CTRL2 0x0074 +#define MAX98927_R0075_BROWNOUT_LVL1_AMP1_CTRL3 0x0075 +#define MAX98927_R0076_BROWNOUT_LVL2_CUR_LIMIT 0x0076 +#define MAX98927_R0077_BROWNOUT_LVL2_AMP1_CTRL1 0x0077 +#define MAX98927_R0078_BROWNOUT_LVL2_AMP1_CTRL2 0x0078 +#define MAX98927_R0079_BROWNOUT_LVL2_AMP1_CTRL3 0x0079 +#define MAX98927_R007A_BROWNOUT_LVL3_CUR_LIMIT 0x007A +#define MAX98927_R007B_BROWNOUT_LVL3_AMP1_CTRL1 0x007B +#define MAX98927_R007C_BROWNOUT_LVL3_AMP1_CTRL2 0x007C +#define MAX98927_R007D_BROWNOUT_LVL3_AMP1_CTRL3 0x007D +#define MAX98927_R007E_BROWNOUT_LVL4_CUR_LIMIT 0x007E +#define MAX98927_R007F_BROWNOUT_LVL4_AMP1_CTRL1 0x007F +#define MAX98927_R0080_BROWNOUT_LVL4_AMP1_CTRL2 0x0080 +#define MAX98927_R0081_BROWNOUT_LVL4_AMP1_CTRL3 0x0081 +#define MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM 0x0082 +#define MAX98927_R0083_ENV_TRACK_BOOST_VOUT_DELAY 0x0083 +#define MAX98927_R0084_ENV_TRACK_REL_RATE 0x0084 +#define MAX98927_R0085_ENV_TRACK_HOLD_RATE 0x0085 +#define MAX98927_R0086_ENV_TRACK_CTRL 0x0086 +#define MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ 0x0087 +#define MAX98927_R00FF_GLOBAL_SHDN 0x00FF +#define MAX98927_R0100_SOFT_RESET 0x0100 +#define MAX98927_R01FF_REV_ID 0x01FF + +/* MAX98927_R0018_PCM_RX_EN_A */ +#define MAX98927_PCM_RX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_RX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_RX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_RX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_RX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_RX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_RX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_RX_CH7_EN (0x1 << 7) + +/* MAX98927_R001A_PCM_TX_EN_A */ +#define MAX98927_PCM_TX_CH0_EN (0x1 << 0) +#define MAX98927_PCM_TX_CH1_EN (0x1 << 1) +#define MAX98927_PCM_TX_CH2_EN (0x1 << 2) +#define MAX98927_PCM_TX_CH3_EN (0x1 << 3) +#define MAX98927_PCM_TX_CH4_EN (0x1 << 4) +#define MAX98927_PCM_TX_CH5_EN (0x1 << 5) +#define MAX98927_PCM_TX_CH6_EN (0x1 << 6) +#define MAX98927_PCM_TX_CH7_EN (0x1 << 7) + +/* MAX98927_R001E_PCM_TX_CH_SRC_A */ +#define MAX98927_PCM_TX_CH_SRC_A_V_SHIFT (0) +#define MAX98927_PCM_TX_CH_SRC_A_I_SHIFT (4) + +/* MAX98927_R001F_PCM_TX_CH_SRC_B */ +#define MAX98927_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 5) + +/* MAX98927_R0020_PCM_MODE_CFG */ +#define MAX98927_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 2) +#define MAX98927_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) +#define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) +#define MAX98927_PCM_FORMAT_I2S (0x0 << 0) +#define MAX98927_PCM_FORMAT_LJ (0x1 << 0) + +#define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98927_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98927_R0021_PCM_MASTER_MODE */ +#define MAX98927_PCM_MASTER_MODE_MASK (0x3 << 0) +#define MAX98927_PCM_MASTER_MODE_SLAVE (0x0 << 0) +#define MAX98927_PCM_MASTER_MODE_MASTER (0x3 << 0) + +#define MAX98927_PCM_MASTER_MODE_MCLK_MASK (0xF << 2) +#define MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT (2) + +/* MAX98927_R0022_PCM_CLK_SETUP */ +#define MAX98927_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98927_R0023_PCM_SR_SETUP1 */ +#define MAX98927_PCM_SR_SET1_SR_MASK (0xF << 0) + +#define MAX98927_PCM_SR_SET1_SR_8000 (0x0 << 0) +#define MAX98927_PCM_SR_SET1_SR_11025 (0x1 << 0) +#define MAX98927_PCM_SR_SET1_SR_12000 (0x2 << 0) +#define MAX98927_PCM_SR_SET1_SR_16000 (0x3 << 0) +#define MAX98927_PCM_SR_SET1_SR_22050 (0x4 << 0) +#define MAX98927_PCM_SR_SET1_SR_24000 (0x5 << 0) +#define MAX98927_PCM_SR_SET1_SR_32000 (0x6 << 0) +#define MAX98927_PCM_SR_SET1_SR_44100 (0x7 << 0) +#define MAX98927_PCM_SR_SET1_SR_48000 (0x8 << 0) + +/* MAX98927_R0024_PCM_SR_SETUP2 */ +#define MAX98927_PCM_SR_SET2_SR_MASK (0xF << 4) +#define MAX98927_PCM_SR_SET2_SR_SHIFT (4) +#define MAX98927_PCM_SR_SET2_IVADC_SR_MASK (0xf << 0) + +/* MAX98927_R0025_PCM_TO_SPK_MONOMIX_A */ +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_MASK (0x3 << 6) +#define MAX98927_PCM_TO_SPK_MONOMIX_CFG_SHIFT (6) + +/* MAX98927_R0035_PDM_RX_CTRL */ +#define MAX98927_PDM_RX_EN_MASK (0x1 << 0) + +/* MAX98927_R0036_AMP_VOL_CTRL */ +#define MAX98927_AMP_VOL_SEL (0x1 << 7) +#define MAX98927_AMP_VOL_SEL_WIDTH (1) +#define MAX98927_AMP_VOL_SEL_SHIFT (7) +#define MAX98927_AMP_VOL_MASK (0x7f << 0) +#define MAX98927_AMP_VOL_WIDTH (7) +#define MAX98927_AMP_VOL_SHIFT (0) + +/* MAX98927_R0037_AMP_DSP_CFG */ +#define MAX98927_AMP_DSP_CFG_DCBLK_EN (0x1 << 0) +#define MAX98927_AMP_DSP_CFG_DITH_EN (0x1 << 1) +#define MAX98927_AMP_DSP_CFG_RMP_BYPASS (0x1 << 4) +#define MAX98927_AMP_DSP_CFG_DAC_INV (0x1 << 5) +#define MAX98927_AMP_DSP_CFG_RMP_SHIFT (4) + +/* MAX98927_R0039_DRE_CTRL */ +#define MAX98927_DRE_CTRL_DRE_EN (0x1 << 0) +#define MAX98927_DRE_EN_SHIFT 0x1 + +/* MAX98927_R003A_AMP_EN */ +#define MAX98927_AMP_EN_MASK (0x1 << 0) + +/* MAX98927_R003B_SPK_SRC_SEL */ +#define MAX98927_SPK_SRC_MASK (0x3 << 0) + +/* MAX98927_R003C_SPK_GAIN */ +#define MAX98927_SPK_PCM_GAIN_MASK (0x7 << 0) +#define MAX98927_SPK_PDM_GAIN_MASK (0x7 << 4) +#define MAX98927_SPK_GAIN_WIDTH (3) + +/* MAX98927_R003E_MEAS_EN */ +#define MAX98927_MEAS_V_EN (0x1 << 0) +#define MAX98927_MEAS_I_EN (0x1 << 1) + +/* MAX98927_R0040_BOOST_CTRL0 */ +#define MAX98927_BOOST_CTRL0_VOUT_MASK (0x1f << 0) +#define MAX98927_BOOST_CTRL0_PVDD_MASK (0x1 << 7) +#define MAX98927_BOOST_CTRL0_PVDD_EN_SHIFT (7) + +/* MAX98927_R0052_BROWNOUT_EN */ +#define MAX98927_BROWNOUT_BDE_EN (0x1 << 0) +#define MAX98927_BROWNOUT_AMP_EN (0x1 << 1) +#define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) +#define MAX98927_BROWNOUT_DSP_SHIFT (2) + +/* MAX98927_R0100_SOFT_RESET */ +#define MAX98927_SOFT_RESET (0x1 << 0) + +/* MAX98927_R00FF_GLOBAL_SHDN */ +#define MAX98927_GLOBAL_EN_MASK (0x1 << 0) + +struct max98927_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98927_pdata *pdata; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int v_l_slot; + unsigned int i_l_slot; + bool interleave_mode; + unsigned int ch_size; + unsigned int rate; + unsigned int iface; + unsigned int master; + unsigned int digital_gain; +}; +#endif -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC fixes for v4.11
by Mark Brown 25 Apr '17

25 Apr '17

The following changes since commit c1ae3cfa0e89fa1a7ecc4c99031f5e9ae99d9201: Linux 4.11-rc1 (2017-03-05 12:59:56 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.11-rc7 for you to fetch changes up to dc9617cb81c4130858b6dd026039f42a39e93c18: Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/topology' and 'asoc/fix/sti' into asoc-linus (2017-04-25 16:25:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.11 A few last minute fixes for v4.11, the STI fix is relatively large but driver specific and has been cooking in -next for a little while now: - A fix from Takashi for some suspend/resume related crashes in the Intel drivers. - A fix from Mousumi Jana for issues with incorrectly created enumeration controls generated from topology files which could cause problems for userspace. - Fixes from Arnaud Pouliquen for some crashes due to races with the interrupt handler in the STI driver. ---------------------------------------------------------------- Arnaud Pouliquen (2): ASoC: STI: Fix reader substream pointer set ASoC: STI: Fix null ptr deference in IRQ handler Mark Brown (1): Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/topology' and 'asoc/fix/sti' into asoc-linus Mousumi Jana (1): ASoC: topology: Fix to store enum text values Takashi Iwai (1): ASoC: intel: Fix PM and non-atomic crash in bytcr drivers sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 2 -- sound/soc/soc-topology.c | 1 + sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 35 ++++++++++++++++++++++++----------- sound/soc/sti/uniperif_reader.c | 27 +++++++++++++++++++++++---- 6 files changed, 51 insertions(+), 19 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: bytcr_rt5640: Fix a typo and quirk parameter type
by Takashi Iwai 25 Apr '17

25 Apr '17

The previous patch for adding the quirk module option had a typo in its info print, which results in a weird output. Also, the parameter type should be rather unsigned int instead of signed int. Fixes: 9f2cf73ed65b ("ASoC: bytcr_rt5640: Allow quirk set via module option") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 0ac32788f216..5ca09cadf39f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -60,7 +60,7 @@ struct byt_rt5640_private { static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; static unsigned int quirk_override; -module_param_named(quirk, quirk_override, int, 0444); +module_param_named(quirk, quirk_override, uint, 0444); MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) @@ -811,7 +811,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* check quirks before creating card */ dmi_check_system(byt_rt5640_quirk_table); if (quirk_override) { - dev_info(&pdev->dev, "Overriding quirk %0x => 0x%x\n", + dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n", (unsigned int)byt_rt5640_quirk, quirk_override); byt_rt5640_quirk = quirk_override; } -- 2.11.1

2 1

[alsa-devel] [PATCH] ASoC: bytcr_rt5640: log quirk configuration errors
by Pierre-Louis Bossart 25 Apr '17

25 Apr '17

Now that quirks can be overridden with a module parameter, log errors so that non-sensical quirks introduced by mistake are identified. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 100 +++++++++++++++++++++++++--------- 1 file changed, 75 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b1f0ccb..44ec96a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -57,6 +57,7 @@ enum { struct byt_rt5640_private { struct clk *mclk; }; +static bool is_bytcr; static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; static unsigned int quirk_override; @@ -65,30 +66,79 @@ MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC1_MAP) - dev_info(dev, "quirk DMIC1_MAP enabled"); - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC2_MAP) - dev_info(dev, "quirk DMIC2_MAP enabled"); - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN1_MAP) - dev_info(dev, "quirk IN1_MAP enabled"); - if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN3_MAP) - dev_info(dev, "quirk IN3_MAP enabled"); - if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) - dev_info(dev, "quirk DMIC enabled"); + int map; + bool has_dmic = false; + bool has_mclk = false; + bool has_ssp0 = false; + bool has_ssp0_aif1 = false; + bool has_ssp0_aif2 = false; + bool has_ssp2_aif2 = false; + + map = BYT_RT5640_MAP(byt_rt5640_quirk); + switch (map) { + case BYT_RT5640_DMIC1_MAP: + dev_info(dev, "quirk DMIC1_MAP enabled\n"); + has_dmic = true; + break; + case BYT_RT5640_DMIC2_MAP: + dev_info(dev, "quirk DMIC2_MAP enabled\n"); + has_dmic = true; + break; + case BYT_RT5640_IN1_MAP: + dev_info(dev, "quirk IN1_MAP enabled\n"); + break; + case BYT_RT5640_IN3_MAP: + dev_info(dev, "quirk IN3_MAP enabled\n"); + break; + default: + dev_err(dev, "quirk map 0x%x is not supported, microphone input will not work\n", map); + break; + } + if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { + if (has_dmic) + dev_info(dev, "quirk DMIC enabled\n"); + else + dev_err(dev, "quirk DMIC enabled but no DMIC input set, will be ignored\n"); + } if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) - dev_info(dev, "quirk MONO_SPEAKER enabled"); - if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) - dev_info(dev, "quirk DIFF_MIC enabled"); - if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) - dev_info(dev, "quirk SSP2_AIF2 enabled"); - if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) - dev_info(dev, "quirk SSP0_AIF1 enabled"); - if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2) - dev_info(dev, "quirk SSP0_AIF2 enabled"); - if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) - dev_info(dev, "quirk MCLK_EN enabled"); - if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) - dev_info(dev, "quirk MCLK_25MHZ enabled"); + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) { + if (!has_dmic) + dev_info(dev, "quirk DIFF_MIC enabled\n"); + else + dev_info(dev, "quirk DIFF_MIC enabled but DMIC input selected, will be ignored\n"); + } + if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) { + dev_info(dev, "quirk SSP0_AIF1 enabled\n"); + has_ssp0 = true; + has_ssp0_aif1 = true; + } + if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2) { + dev_info(dev, "quirk SSP0_AIF2 enabled\n"); + has_ssp0 = true; + has_ssp0_aif2 = true; + } + if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) { + dev_info(dev, "quirk SSP2_AIF2 enabled\n"); + has_ssp2_aif2 = true; + } + if (is_bytcr && !has_ssp0) + dev_err(dev, "Invalid routing, bytcr detected but no SSP0-based quirk, audio cannot work with SSP2 on bytcr\n"); + if (has_ssp0_aif1 && has_ssp0_aif2) + dev_err(dev, "Invalid routing, SSP0 cannot be connected to both AIF1 and AIF2\n"); + if (has_ssp0 && has_ssp2_aif2) + dev_err(dev, "Invalid routing, cannot have both SSP0 and SSP2 connected to codec\n"); + + if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) { + dev_info(dev, "quirk MCLK_EN enabled\n"); + has_mclk = true; + } + if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) { + if (has_mclk) + dev_info(dev, "quirk MCLK_25MHZ enabled\n"); + else + dev_err(dev, "quirk MCLK_25MHZ enabled but quirk MCLK not selected, will be ignored\n"); + } } @@ -132,7 +182,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, ret = clk_prepare_enable(priv->mclk); if (ret < 0) { dev_err(card->dev, - "could not configure MCLK state"); + "could not configure MCLK state\n"); return ret; } } @@ -714,8 +764,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) int i; int dai_index; struct byt_rt5640_private *priv; - bool is_bytcr = false; + is_bytcr = false; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); if (!priv) return -ENOMEM; -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: tas2552: Propagate the error code in suspend/resume
by Fabio Estevam 25 Apr '17

25 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> tas2552_suspend() and tas2552_resume() currently always return success, even though they may fail. Fix this behaviour by always propagating the error code. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/tas2552.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index fd5251e..8840f72 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -637,7 +637,7 @@ static int tas2552_suspend(struct snd_soc_codec *codec) if (ret != 0) dev_err(codec->dev, "Failed to disable supplies: %d\n", ret); - return 0; + return ret; } static int tas2552_resume(struct snd_soc_codec *codec) @@ -653,7 +653,7 @@ static int tas2552_resume(struct snd_soc_codec *codec) ret); } - return 0; + return ret; } #else #define tas2552_suspend NULL -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] Relax bitclk computation when using PLL
by Daniel Baluta 25 Apr '17

25 Apr '17

Using strict bitclk requirements we cannot support all promised rates and formats. For this reason we relax bitclk computation by choosing the best available bitclk. First patch in the series is based on Arnd's patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119899.html Second one does the actual bitclk relaxation. Daniel Baluta (2): ASoC: codec: wm9860: avoid maybe-uninitialized warning ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 29 ++++++++++++++++++++--------- 1 file changed, 20 insertions(+), 9 deletions(-) -- 2.7.4

3 7

[alsa-devel] PM issue with Intel SST Atom driver
by Takashi Iwai 25 Apr '17

25 Apr '17

Hi, I noticed that the unstable PM behavior on my Cherrytrail laptop actually comes from the sound driver. First off, the atom/sst/sst.c has the following suspend code: static int intel_sst_suspend(struct device *dev) { .... /* * check if any stream is active and running * they should already by suspend by soc_suspend */ for (i = 1; i <= ctx->info.max_streams; i++) { struct stream_info *stream = &ctx->streams[i]; if (stream->status == STREAM_RUNNING) { dev_err(dev, "stream %d is running, can't suspend, abort\n", i); return -EBUSY; } } This doesn't look good, and I actually hit this when I tried to suspend while playing something. In general, the driver shouldn't reject at this point, because this is the PM suspend callback, i.e. the user wants to suspend the device inevitably. The driver should return an error only when it faces to a fatal error. But I wondered why this happened at all, and noticed that the machine driver (in my case bytcr_rt5640) has no its own PM ops. But hooking the snd_soc_pm_ops there seems causing a hang up at suspend by some reason. Maybe this wasn't a big problem until now since the BYT/CHT didn't support the suspend/resume properly in the past. But now PM suspend is supported on these devices, so the problem surfaced more often. Could you guys check the status of these drivers? thanks, Takashi

3 15

[alsa-devel] [PATCH] ASoC: intel: Fix PM and non-atomic crash in bytcr drivers
by Takashi Iwai 25 Apr '17

25 Apr '17

The FE setups of Intel SST bytcr_rt5640 and bytcr_rt5651 drivers carry the ignore_suspend flag, and this prevents the suspend/resume working properly while the stream is running, since SST core code has the check of the running streams and returns -EBUSY. Drop these superfluous flags for fixing the behavior. Also, the bytcr_rt5640 driver lacks of nonatomic flag in some FE definitions, which leads to the kernel Oops at suspend/resume like: BUG: scheduling while atomic: systemd-sleep/3144/0x00000003 Call Trace: dump_stack+0x5c/0x7a __schedule_bug+0x55/0x70 __schedule+0x63c/0x8c0 schedule+0x3d/0x90 schedule_timeout+0x16b/0x320 ? del_timer_sync+0x50/0x50 ? sst_wait_timeout+0xa9/0x170 [snd_intel_sst_core] ? sst_wait_timeout+0xa9/0x170 [snd_intel_sst_core] ? remove_wait_queue+0x60/0x60 ? sst_prepare_and_post_msg+0x275/0x960 [snd_intel_sst_core] ? sst_pause_stream+0x9b/0x110 [snd_intel_sst_core] .... This patch addresses these appropriately, too. Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 2 -- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c7219fb3aa8..9e2a3404a836 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -621,7 +621,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -634,7 +634,6 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, @@ -661,6 +660,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { | SND_SOC_DAIFMT_CBS_CFS, .be_hw_params_fixup = byt_rt5640_codec_fixup, .ignore_suspend = 1, + .nonatomic = true, .dpcm_playback = 1, .dpcm_capture = 1, .init = byt_rt5640_init, diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 3186f015939f..8164bec63bf1 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -235,7 +235,6 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, @@ -249,7 +248,6 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, -- 2.11.1

4 6

[alsa-devel] [PATCH v2] ASoC: imx-wm8962: Remove global variables
by Fabio Estevam 24 Apr '17

24 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> Currently the following variables are global: - card_priv, sample_rate and sample_format ,which is not a good idea as it prevents the usage of multiple instances. Make sample_rate and sample_format part of the imx_priv structure and allocate imx_priv via the standard devm_kzalloc() mechanism inside the probe function. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Keep the original variable types for sample_rate and sample_format. sound/soc/fsl/imx-wm8962.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52659fa..206b898 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -38,8 +38,9 @@ struct imx_wm8962_data { struct imx_priv { struct platform_device *pdev; + int sample_rate; + snd_pcm_format_t sample_format; }; -static struct imx_priv card_priv; static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -48,14 +49,14 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; -static int sample_rate = 44100; -static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; - static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - sample_rate = params_rate(params); - sample_format = params_format(params); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); return 0; } @@ -70,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct imx_priv *priv = &card_priv; + struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; @@ -84,10 +85,10 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; else - pll_out = sample_rate * 256; + pll_out = priv->sample_rate * 256; ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, data->clk_frequency, @@ -139,7 +140,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct imx_priv *priv = &card_priv; + struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -159,14 +160,20 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; - struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_wm8962_data *data; + struct imx_priv *priv; struct clk *codec_clk; int int_port, ext_port; int ret; + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + priv->pdev = pdev; + priv->sample_rate = 44100; + priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; ret = of_property_read_u32(np, "mux-int-port", &int_port); if (ret) { -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Provide a dummy wrapper of snd_soc_set_dmi_name()
by Takashi Iwai 24 Apr '17

24 Apr '17

For systems without DMI, it makes no sense to have the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 2 ++ 2 files changed, 10 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index cdfb55f7aede..915c06cb2b32 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -497,7 +497,15 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream); int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt); +#ifdef CONFIG_DMI int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour); +#else +static inline int snd_soc_set_dmi_name(struct snd_soc_card *card, + const char *flavour) +{ + return 0; +} +#endif /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2722bb0c5573..aac73ff8a92a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1918,6 +1918,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); +#ifdef CONFIG_DMI /* Trim special characters, and replace '-' with '_' since '-' is used to * separate different DMI fields in the card long name. Only number and * alphabet characters and a few separator characters are kept. @@ -2049,6 +2050,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) return 0; } EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); +#endif /* CONFIG_DMI */ static int snd_soc_instantiate_card(struct snd_soc_card *card) { -- 2.11.1

5 6

[alsa-devel] [PATCH v2 0/2] ASoC: Give prettier longname based on DMI
by Takashi Iwai 24 Apr '17

24 Apr '17

Hi, this is the revised patchset to set the card longname based on DMI so that a unique UCM profile can be picked up. v1->v2: Fix the missnig NULL argument in the function call Takashi === Takashi Iwai (2): ASoC: Provide a dummy wrapper of snd_soc_set_dmi_name() ASoC: Call snd_soc_set_dmi_name() unconditionally include/sound/soc.h | 8 ++++++++ sound/soc/intel/boards/broadwell.c | 3 --- sound/soc/soc-core.c | 5 +++++ 3 files changed, 13 insertions(+), 3 deletions(-) -- 2.11.1

3 4

[alsa-devel] [PATCH v3] ASoC: bytcr_rt5640: Allow quirk set via module option
by Takashi Iwai 24 Apr '17

24 Apr '17

The bytcr-rt5640 driver has a few quirk setups depending on the board, where the quirk value is set by DMI matching. When you have a new device to add the support, you often experience to try the different quirk by trial-and-error. Or, you may have a development model that still has no proper DMI string. In either case, you'd need to compile the driver at each time. This patch introduces a module option to override the quirk value on the fly. User can boot like snd-soc-sst-bytcr-rt5640.quirk=0x4004 to override the default value without recompilation. It's a raw value, so user needs to check the source code for the meaning of each bit. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: override the quirk value completely v2->v3: print the override information sound/soc/intel/boards/bytcr_rt5640.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c7219fb3aa8..0ac32788f216 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -19,6 +19,7 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/device.h> @@ -58,6 +59,9 @@ struct byt_rt5640_private { }; static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; +static unsigned int quirk_override; +module_param_named(quirk, quirk_override, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { @@ -806,6 +810,11 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* check quirks before creating card */ dmi_check_system(byt_rt5640_quirk_table); + if (quirk_override) { + dev_info(&pdev->dev, "Overriding quirk %0x => 0x%x\n", + (unsigned int)byt_rt5640_quirk, quirk_override); + byt_rt5640_quirk = quirk_override; + } log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || -- 2.11.1

3 2

[alsa-devel] [PATCH] ASoC: wm5100: fix spelling mistake: "micropone" -> "microphone"
by Colin King 24 Apr '17

24 Apr '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm5100.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 560575000cc5..138a84efdd54 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2014,7 +2014,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) ret = regmap_read(wm5100->regmap, WM5100_MIC_DETECT_3, &val); if (ret != 0) { - dev_err(wm5100->dev, "Failed to read micropone status: %d\n", + dev_err(wm5100->dev, "Failed to read microphone status: %d\n", ret); return; } -- 2.11.0

4 3

[alsa-devel] [PATCH] ASoC: fix odd_ptr_err.cocci warnings (fwd)
by Julia Lawall 24 Apr '17

24 Apr '17

I don't think the patch is correct, so I haven't bothered signing off on it. Probably the & is not correct in the IS_ERR call? julia ---------- Forwarded message ---------- Date: Sat, 22 Apr 2017 06:09:26 +0800 From: kbuild test robot <fengguang.wu(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: [PATCH] ASoC: fix odd_ptr_err.cocci warnings PTR_ERR should access the value just tested by IS_ERR Semantic patch information: There can be false positives in the patch case, where it is the call to IS_ERR that is wrong. Generated by: scripts/coccinelle/tests/odd_ptr_err.cocci CC: Simon Ho <simon.ho(a)conexant.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- Please take the patch only if it's a positive warning. Thanks! cx2072x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -2183,7 +2183,7 @@ static int cx2072x_i2c_probe(struct i2c_ &cx2072x_regmap); if (IS_ERR(&cx2072x->regmap)) { - ret = PTR_ERR(cx2072x->regmap); + ret = PTR_ERR(&cx2072x->regmap); dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); return ret; }

3 2

[alsa-devel] Status of ALSA firewire stack at Linux v4.12 development period
by Takashi Sakamoto 24 Apr '17

24 Apr '17

Hi, We are now in the last week of development period for Linux v4.12. All of my patches were already merged into 'for-next' branch on maintainer's repository. This is a digest for the subject. ## New drivers Two drivers were newly added to sound subsystem: * A driver for MOTU FireWire series (snd-firewire-motu) * http://mailman.alsa-project.org/pipermail/alsa-devel/2017-March/119055.html * A driver for RME Fireface series (snd-fireface) * http://mailman.alsa-project.org/pipermail/alsa-devel/2017-March/119360.html Although both drivers support a part of units on each series, I won't work for the other units by my self anymore. I wish users are going to work for enhancement of support for more units. ## Changes for existent drivers Additionally, below drivers were widely changed: * A driver for Digidesign Digi00x series (snd-firewire-dg00x) * http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119397.html * A driver for TASCAM FireWire series (snd-firewire-tascam) * http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119748.html Purpose of the changes is mainly for maintenance. Especially, digi00x driver now got a proper support for its console models. Correspondingly, some feature of firewire-lib move to firewire-tascam driver. ## Additional Tracepoints Furthermore, ALSA firewire stack newly supports additional tracepoints: * snd_firewire_lib subsystem * in_packet_without_header/out_packet_without_header * snd_firewire_motu subsystem * in_data_block_sph/out_data_block_sph * in_data_block_message/out_data_block_mesasge These tracepoints are available to debug behaviours of drivers newly added in this development period. Of cource, below tracepoints are still available for existent drivers: * snd_firewire_lib subsystem * in_packet/out_packet These tracepoints are available via tracefs or perf_event_open(2). Please refer to proper documentation. ## Known issues As known issues of IEC 61883-1/6 engine on ALSA firewire stack, the engine takes target units to generate sound with periodical short noise. This can be observed except for units based on BeBoB or Fireface series, and this can not be observed every time of packet streaming. The tracepoints declare that the cause does _not_ come from packet queueing timing and throughput of the amount of data blocks. Thus, It comes from protocol implementation itself. When consider about a fact that this depends on the timing to start packet streaming, the cause is that each unit has a strong demand about sequence of the number of data blocks in a packet. It might tight on actual isochronous cycle and the other time clock somehow. Additionally, as I have reported, some of units based on Dice don't transfer/receive exactly the same number of total data block as sampling transmission frequency. In detail, seee: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html ## Useful stuffs For Linux kernel 4.4 or later, backported code is available in my github repository: https://github.com/takaswie/snd-firewire-improve Drivers in ALSA firewire stack support ALSA HwDep interface. There's a sample implementation for user space; libhinawa: https://github.com/takaswie/libhinawa In detail, please refer to documentation of the library. Some functionalities of supported units are controllable by tools in hinawa-utils. This is a sample application of the libhinawa with GOject Introspection support: https://github.com/takaswie/hinawa-utils Currently, they're maintained to assist my work for ALSA firewire stack. Although it can not satisfy all demands from usual users, I'm welcome to receive PRs to integrate the tools. $ git diff --shortstat v4.11-rc1.. sound/firewire/ 43 files changed, 5791 insertions(+), 502 deletions(-) Regards Takashi Sakamoto

1 0

[alsa-devel] [PATCH 00/16] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 24 Apr '17

24 Apr '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Mayuresh Kulkarni (1): ASoC: wm_adsp: add support for DSP region lock Richard Fitzgerald (15): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: madera-ldo1: LDO1 driver for Cirrus Logic Madera codecs regulator: madera-micsupp: Mic supply for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: wm_adsp: Add support for ADSP2V2 ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 .../devicetree/bindings/gpio/gpio-madera.txt | 24 + .../interrupt-controller/cirrus,madera.txt | 31 + Documentation/devicetree/bindings/mfd/madera.txt | 79 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 103 + .../devicetree/bindings/regulator/madera-ldo1.txt | 29 + .../bindings/regulator/madera-micsupp.txt | 27 + Documentation/devicetree/bindings/sound/madera.txt | 63 + MAINTAINERS | 28 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 173 + drivers/irqchip/Kconfig | 5 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 349 + drivers/mfd/Kconfig | 41 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1688 ++++ drivers/mfd/cs47l85-tables.c | 3169 +++++++ drivers/mfd/cs47l90-tables.c | 2830 +++++++ drivers/mfd/madera-core.c | 689 ++ drivers/mfd/madera-i2c.c | 130 + drivers/mfd/madera-spi.c | 131 + drivers/mfd/madera.h | 52 + drivers/pinctrl/Kconfig | 22 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-madera.c | 1092 +++ drivers/regulator/Kconfig | 16 + drivers/regulator/Makefile | 2 + drivers/regulator/madera-ldo1.c | 198 + drivers/regulator/madera-micsupp.c | 260 + include/dt-bindings/sound/madera.h | 18 + include/linux/irqchip/irq-madera-pdata.h | 19 + include/linux/irqchip/irq-madera.h | 96 + include/linux/mfd/madera/core.h | 175 + include/linux/mfd/madera/pdata.h | 88 + include/linux/mfd/madera/registers.h | 8832 ++++++++++++++++++++ include/linux/regulator/madera-ldo1.h | 24 + include/linux/regulator/madera-micsupp.h | 21 + include/sound/madera-pdata.h | 70 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1747 ++++ sound/soc/codecs/cs47l85.c | 2706 ++++++ sound/soc/codecs/cs47l90.c | 2645 ++++++ sound/soc/codecs/madera.c | 4430 ++++++++++ sound/soc/codecs/madera.h | 470 ++ sound/soc/codecs/wm_adsp.c | 324 +- sound/soc/codecs/wm_adsp.h | 24 + 48 files changed, 32930 insertions(+), 43 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/gpio-madera.txt create mode 100644 Documentation/devicetree/bindings/interrupt-controller/cirrus,madera.txt create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/regulator/madera-ldo1.txt create mode 100644 Documentation/devicetree/bindings/regulator/madera-micsupp.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/pinctrl-madera.c create mode 100644 drivers/regulator/madera-ldo1.c create mode 100644 drivers/regulator/madera-micsupp.c create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/linux/regulator/madera-ldo1.h create mode 100644 include/linux/regulator/madera-micsupp.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

7 46

[alsa-devel] [PATCH] ASoC: imx-wm8962: Remove global variables
by Fabio Estevam 24 Apr '17

24 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> Currently the following variables are global: - card_priv, sample_rate and sample_format ,which is not a good idea as it prevents the usage of multiple instances. Make sample_rate and sample_format part of the imx_priv structure and allocate imx_priv via the standard devm_kzalloc() mechanism inside the probe function. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/imx-wm8962.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52659fa..fbeada7 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -38,8 +38,9 @@ struct imx_wm8962_data { struct imx_priv { struct platform_device *pdev; + u32 sample_rate; + u32 sample_format; }; -static struct imx_priv card_priv; static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -48,14 +49,14 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; -static int sample_rate = 44100; -static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; - static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - sample_rate = params_rate(params); - sample_format = params_format(params); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); return 0; } @@ -70,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct imx_priv *priv = &card_priv; + struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; @@ -84,10 +85,10 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; else - pll_out = sample_rate * 256; + pll_out = priv->sample_rate * 256; ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, data->clk_frequency, @@ -139,7 +140,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct imx_priv *priv = &card_priv; + struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -159,14 +160,20 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; - struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_wm8962_data *data; + struct imx_priv *priv; struct clk *codec_clk; int int_port, ext_port; int ret; + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + priv->pdev = pdev; + priv->sample_rate = 44100; + priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; ret = of_property_read_u32(np, "mux-int-port", &int_port); if (ret) { -- 2.7.4

1 0

[alsa-devel] [PATCH 0/4] Workarounds for HD-audio dual codecs
by Takashi Iwai 23 Apr '17

23 Apr '17

Hi, recently the hardware vendors tend to put multiple codecs for the onboard audio, and it causes a problem with the current HD-audio auto-parser, as it supposes only a single codec. With the multiple codecs, some existing features are hard to implement, such as the vmaster or the stream switching because basically the multiple codecs mean to deal with multiple individual streams. In this patchset, some workarounds are implemented to make the multiple codecs more or less working. In this mode, the single master control or auto-mute/mic switching are disabled, and multiple PCM streams are created depending on the I/O. These differences have to be handled via UCM, and for providing the unique setup, the card longname is overridden in the quirk. Takashi === Takashi Iwai (4): ALSA: hda - A new flag to enforce prefix to each pin ALSA: hda - Allow to enable/disable vmaster build explicitly ALSA: hda - Support Gigabyte Gaming board with dual Realtek codecs ALSA: hda - Use a helper function for renaming kctl names Documentation/sound/hd-audio/notes.rst | 2 + sound/pci/hda/hda_auto_parser.c | 1 + sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_generic.c | 9 ++++- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 72 ++++++++++++++++++++++++++++++---- 6 files changed, 77 insertions(+), 9 deletions(-) -- 2.11.1

1 4

[alsa-devel] [PATCH v2] ASoC: bytcr_rt5640: Allow quirk set via module option
by Takashi Iwai 21 Apr '17

21 Apr '17

The bytcr-rt5640 driver has a few quirk setups depending on the board, where the quirk value is set by DMI matching. When you have a new device to add the support, you often experience to try the different quirk by trial-and-error. Or, you may have a development model that still has no proper DMI string. In either case, you'd need to compile the driver at each time. This patch introduces a module option to override the quirk value on the fly. User can boot like snd-soc-sst-bytcr-rt5640.quirk=0x4004 to override the default value without recompilation. It's a raw value, so user needs to check the source code for the meaning of each bit. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: override the quirk value completely sound/soc/intel/boards/bytcr_rt5640.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c7219fb3aa8..23f5ffdb3731 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -19,6 +19,7 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/device.h> @@ -58,6 +59,9 @@ struct byt_rt5640_private { }; static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; +static unsigned int quirk_override; +module_param_named(quirk, quirk_override, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { @@ -806,6 +810,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* check quirks before creating card */ dmi_check_system(byt_rt5640_quirk_table); + if (quirk_override) + byt_rt5640_quirk = quirk_override; log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || -- 2.11.1

2 2

[alsa-devel] [PATCH] ALSA: hda: fix a locking bug in dead code
by Dan Carpenter 21 Apr '17

21 Apr '17

We added an impossible return to silence a static checker warning but it introduced another warning because we need to drop the lock before returning. Fixes: 960a581e22d9 ("ALSA: hda: fix some klockwork scan warnings") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 90e4ff87445e..0dc18c4a3b1b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1808,9 +1808,12 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) /* Add sanity check to pass klockwork check. * This should never happen. */ - if (WARN_ON(spdif == NULL)) - return true; + if (WARN_ON(spdif == NULL)) { + non_pcm = true; + goto unlock; + } non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); +unlock: mutex_unlock(&codec->spdif_mutex); return non_pcm; }

1 0

[alsa-devel] [PATCH] ASoC: bytcr_rt5640: Allow quirk set via module option
by Takashi Iwai 21 Apr '17

21 Apr '17

The bytcr-rt5640 driver has a few quirk setups depending on the board, where the quirk value is set by DMI matching. When you have a new device to add the support, you often experience to try the different quirk by trial-and-error. Or, you may have a development model that still has no proper DMI string. In either case, you'd need to compile the driver at each time. This patch introduces a module option to override the quirk value on the fly. User can boot like snd-soc-sst-bytcr-rt5640.quirk=0x4004 to override the default value without recompilation. It's a raw value, so user needs to check the source code for the meaning of each bit. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/boards/bytcr_rt5640.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c7219fb3aa8..b848cbf575a1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -19,6 +19,7 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/device.h> @@ -57,7 +58,9 @@ struct byt_rt5640_private { struct clk *mclk; }; -static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; +static unsigned int byt_rt5640_quirk = BYT_RT5640_MCLK_EN; +module_param_named(quirk, byt_rt5640_quirk, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { -- 2.11.1

2 2

[alsa-devel] [PATCH] ASoC: rsnd: fix semicolon.cocci warnings
by kbuild test robot 21 Apr '17

21 Apr '17

sound/soc/sh/rcar/adg.c:462:54-55: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci CC: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- adg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -459,7 +459,7 @@ static void rsnd_adg_get_clkout(struct r * ADG supports BRRA/BRRB output only * this means all clkout0/1/2/3 will be same rate */ - prop = of_find_property(np, "clock-frequency", NULL);; + prop = of_find_property(np, "clock-frequency", NULL); req_size = prop->length / sizeof(u32); of_property_read_u32_array(np, "clock-frequency", req_rate, req_size);

3 2

[alsa-devel] Applied "ASoC: rsnd: Fix possible NULL pointer dereference" to the asoc tree
by Mark Brown 21 Apr '17

21 Apr '17

The patch ASoC: rsnd: Fix possible NULL pointer dereference 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 e8dffe6c2004278c588b3bb441a3dbe998a3f2e4 Mon Sep 17 00:00:00 2001 From: Marek Vasut <marek.vasut+renesas(a)gmail.com> Date: Fri, 21 Apr 2017 00:41:20 +0000 Subject: [PATCH] ASoC: rsnd: Fix possible NULL pointer dereference 25165f79adc76b812bfb4d8f2ab120aafb28d0e6 ("ASoC: rsnd: enable clock-frequency for both 44.1kHz/48kHz") supports both 44.1kHz/48kHz clock-frequency settings for ADG which will be used for AUDIO_OLKOUTn. But some board doesn't need it, thus, it is not mandatory. But, above patch didn't care about the case of "clock-frequency" DT property was not present. This patch ignores ADG settings if AUDIO_OLKOUTn was not used. Signed-off-by: Marek Vasut <marek.vasut+renesas(a)gmail.com> [Kuninori: tidyup not to break non AUDIO_OLKOUTn case] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/adg.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index faa1a4f09766..66203d107a11 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -453,13 +453,18 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, [CLKI] = 0x2, }; - of_property_read_u32(np, "#clock-cells", &count); + ckr = 0; + rbga = 2; /* default 1/6 */ + rbgb = 2; /* default 1/6 */ /* * ADG supports BRRA/BRRB output only * this means all clkout0/1/2/3 will be same rate */ prop = of_find_property(np, "clock-frequency", NULL); + if (!prop) + goto rsnd_adg_get_clkout_end; + req_size = prop->length / sizeof(u32); of_property_read_u32_array(np, "clock-frequency", req_rate, req_size); @@ -472,6 +477,9 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, req_48kHz_rate = req_rate[i]; } + if (req_rate[0] % 48000 == 0) + adg->flags = AUDIO_OUT_48; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. @@ -481,9 +489,6 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * rsnd_adg_ssi_clk_try_start() * rsnd_ssi_master_clk_start() */ - ckr = 0; - rbga = 2; /* default 1/6 */ - rbgb = 2; /* default 1/6 */ adg->rbga_rate_for_441khz = 0; adg->rbgb_rate_for_48khz = 0; for_each_rsnd_clk(clk, adg, i) { @@ -528,6 +533,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * this means all clkout0/1/2/3 will be * same rate */ + of_property_read_u32(np, "#clock-cells", &count); /* * for clkout */ @@ -557,13 +563,11 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, &adg->onecell); } +rsnd_adg_get_clkout_end: adg->ckr = ckr; adg->rbga = rbga; adg->rbgb = rbgb; - if (req_rate[0] % 48000 == 0) - adg->flags = AUDIO_OUT_48; - for_each_rsnd_clkout(clk, adg, i) dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Uninitialized variable in probe_codec()
by Dan Carpenter 21 Apr '17

21 Apr '17

My static checker complains that if snd_hdac_bus_get_response() returns -EIO then "res" is uninitialized. Fix this by initializing it to -1 so that the error is handled correctly. Fixes: d8c2dab8381d ("ASoC: Intel: Add Skylake HDA audio driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 0c57d4eaae3a..6df3b317a476 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -512,7 +512,7 @@ static int probe_codec(struct hdac_ext_bus *ebus, int addr) struct hdac_bus *bus = ebus_to_hbus(ebus); unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; - unsigned int res; + unsigned int res = -1; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd);

2 1

[alsa-devel] [PATCH] ASoC: mediatek: add I2C dependency for CS42XX8
by Arnd Bergmann 21 Apr '17

21 Apr '17

We should not select drivers that depend on I2C when that is disabled, as it results in a build error: warning: (SND_SOC_MT2701_WM8960) selects SND_SOC_WM8960 which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && I2C) sound/soc/codecs/wm8960.c:1469:1: error: data definition has no type or storage class [-Werror] sound/soc/codecs/wm8960.c:1469:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] Fixes: 8625c1dbd876 ("ASoC: mediatek: Add mt2701-wm8960 machine driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index c6f39040f71f..5c68797f36c4 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -24,7 +24,7 @@ config SND_SOC_MT2701_CS42448 config SND_SOC_MT2701_WM8960 tristate "ASoc Audio driver for MT2701 with WM8960 codec" - depends on SND_SOC_MT2701 + depends on SND_SOC_MT2701 && I2C select SND_SOC_WM8960 help This adds ASoC driver for Mediatek MT2701 boards -- 2.9.0

3 2

[alsa-devel] [PATCH 1/1] ASoC: fsl_asrc: use dma_transfer_direction enum when calling dmaengine_prep_dma_cyclic
by Nicolas Iooss 21 Apr '17

21 Apr '17

When fsl_asrc_dma_prepare_and_submit() calls dmaengine_prep_dma_cyclic(), it uses either DMA_TO_DEVICE or DMA_FROM_DEVICE. These values are both items of dma_data_direction enum, not of dma_data_direction enum, which is the type expected by dmaengine_prep_dma_cyclic(). Replace DMA_TO_DEVICE with DMA_MEM_TO_DEV and DMA_FROM_DEVICE with DMA_DEV_TO_MEM to fix this type mismatch issue. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- sound/soc/fsl/fsl_asrc_dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index dc30d780f874..282d841840b1 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -76,7 +76,7 @@ static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream) pair->dma_chan[!dir], runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), - dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags); + dir == OUT ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, flags); if (!pair->desc[!dir]) { dev_err(dev, "failed to prepare slave DMA for Front-End\n"); return -ENOMEM; -- 2.11.0

4 3

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix a couple user after free bugs
by Dan Carpenter 21 Apr '17

21 Apr '17

We need to use the _safe() version of list_for_each_entry() here because of the kfree(modules). Fixes: b8c722ddd548 ("ASoC: Intel: Skylake: Add support for deferred DSP module bind") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 600faad19bd4..d43d1976dd3b 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1323,10 +1323,10 @@ int skl_platform_unregister(struct device *dev) { struct hdac_ext_bus *ebus = dev_get_drvdata(dev); struct skl *skl = ebus_to_skl(ebus); - struct skl_module_deferred_bind *modules; + struct skl_module_deferred_bind *modules, *tmp; if (!list_empty(&skl->bind_list)) { - list_for_each_entry(modules, &skl->bind_list, node) { + list_for_each_entry_safe(modules, tmp, &skl->bind_list, node) { list_del(&modules->node); kfree(modules); } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 74b3acf3bb07..aea7917e791c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1091,7 +1091,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, struct skl_module_cfg *src_module = NULL, *dst_module; struct skl_sst *ctx = skl->skl_sst; struct skl_pipe *s_pipe = mconfig->pipe; - struct skl_module_deferred_bind *modules; + struct skl_module_deferred_bind *modules, *tmp; if (s_pipe->state == SKL_PIPE_INVALID) return -EINVAL; @@ -1105,7 +1105,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, src_module = w_module->w->priv; - list_for_each_entry(modules, &skl->bind_list, node) { + list_for_each_entry_safe(modules, tmp, &skl->bind_list, node) { /* * When the destination module is deleted, Unbind the * modules from deferred bind list.

2 1

[alsa-devel] [PATCH] ASoC: cs35l35: Allow user to configure IMON SCALE
by Charles Keepax 21 Apr '17

21 Apr '17

On the chip the IMON signal is a full 24-bits however normally only some of the bits will be sent over the bus. The chip provides a field to select which bits of the IMON will be sent back, this is the only feedback signal that has this feature. Add an additional entry to the cirrus,imon device tree property to allow the IMON scale parameter to be passed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- I realise this is changing the binding slightly, but thought we might be able to sneak it before cs35l35 ships as it hasn't shipped in a kernel yet. However, it would be relatively simple (although makes the code a little messier) to support both the 3 and 4 entry versions of the binding if people prefer? Thanks, Charles .../devicetree/bindings/sound/cs35l35.txt | 4 ++-- include/sound/cs35l35.h | 1 + sound/soc/codecs/cs35l35.c | 22 +++++++++++++++------- sound/soc/codecs/cs35l35.h | 3 +++ 4 files changed, 21 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt index 457d176..016b768 100644 --- a/Documentation/devicetree/bindings/sound/cs35l35.txt +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -118,8 +118,8 @@ Optional Monitor Signal Format sub-node: Sections 7.44 - 7.53 lists values for the depth, location, and frame for each monitoring signal. - - cirrus,imon : 3 8 bit values to set the depth, location, and frame - of the IMON monitor signal. + - cirrus,imon : 4 8 bit values to set the depth, location, frame and ADC + scale of the IMON monitor signal. - cirrus,vmon : 3 8 bit values to set the depth, location, and frame of the VMON monitor signal. diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h index 88744bb..29da899 100644 --- a/include/sound/cs35l35.h +++ b/include/sound/cs35l35.h @@ -57,6 +57,7 @@ struct monitor_cfg { u8 imon_dpth; u8 imon_loc; u8 imon_frm; + u8 imon_scale; u8 vmon_dpth; u8 vmon_loc; u8 vmon_frm; diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 6ecb7dd..88c48e2 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -918,6 +918,11 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) CS35L35_MON_FRM_MASK, monitor_config->imon_frm << CS35L35_MON_FRM_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_SCALE_CTL, + CS35L35_IMON_SCALE_MASK, + monitor_config->imon_scale << + CS35L35_IMON_SCALE_SHIFT); } if (monitor_config->vpmon_specs) { regmap_update_bits(cs35l35->regmap, @@ -1161,7 +1166,9 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, struct classh_cfg *classh_config = &pdata->classh_algo; struct monitor_cfg *monitor_config = &pdata->mon_cfg; unsigned int val32 = 0; - u8 monitor_array[3]; + u8 monitor_array[4]; + const int imon_array_size = ARRAY_SIZE(monitor_array); + const int mon_array_size = imon_array_size - 1; int ret = 0; if (!np) @@ -1302,15 +1309,16 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, monitor_config->is_present = signal_format ? true : false; if (monitor_config->is_present) { ret = of_property_read_u8_array(signal_format, "cirrus,imon", - monitor_array, ARRAY_SIZE(monitor_array)); + monitor_array, imon_array_size); if (!ret) { monitor_config->imon_specs = true; monitor_config->imon_dpth = monitor_array[0]; monitor_config->imon_loc = monitor_array[1]; monitor_config->imon_frm = monitor_array[2]; + monitor_config->imon_scale = monitor_array[3]; } ret = of_property_read_u8_array(signal_format, "cirrus,vmon", - monitor_array, ARRAY_SIZE(monitor_array)); + monitor_array, mon_array_size); if (!ret) { monitor_config->vmon_specs = true; monitor_config->vmon_dpth = monitor_array[0]; @@ -1318,7 +1326,7 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, monitor_config->vmon_frm = monitor_array[2]; } ret = of_property_read_u8_array(signal_format, "cirrus,vpmon", - monitor_array, ARRAY_SIZE(monitor_array)); + monitor_array, mon_array_size); if (!ret) { monitor_config->vpmon_specs = true; monitor_config->vpmon_dpth = monitor_array[0]; @@ -1326,7 +1334,7 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, monitor_config->vpmon_frm = monitor_array[2]; } ret = of_property_read_u8_array(signal_format, "cirrus,vbstmon", - monitor_array, ARRAY_SIZE(monitor_array)); + monitor_array, mon_array_size); if (!ret) { monitor_config->vbstmon_specs = true; monitor_config->vbstmon_dpth = monitor_array[0]; @@ -1334,7 +1342,7 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, monitor_config->vbstmon_frm = monitor_array[2]; } ret = of_property_read_u8_array(signal_format, "cirrus,vpbrstat", - monitor_array, ARRAY_SIZE(monitor_array)); + monitor_array, mon_array_size); if (!ret) { monitor_config->vpbrstat_specs = true; monitor_config->vpbrstat_dpth = monitor_array[0]; @@ -1342,7 +1350,7 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, monitor_config->vpbrstat_frm = monitor_array[2]; } ret = of_property_read_u8_array(signal_format, "cirrus,zerofill", - monitor_array, ARRAY_SIZE(monitor_array)); + monitor_array, mon_array_size); if (!ret) { monitor_config->zerofill_specs = true; monitor_config->zerofill_dpth = monitor_array[0]; diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index 54e9ac5..5a6e43a 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -148,6 +148,9 @@ #define CS35L35_MON_FRM_MASK 0x80 #define CS35L35_MON_FRM_SHIFT 7 +#define CS35L35_IMON_SCALE_MASK 0xF8 +#define CS35L35_IMON_SCALE_SHIFT 3 + #define CS35L35_MS_MASK 0x80 #define CS35L35_MS_SHIFT 7 #define CS35L35_SPMODE_MASK 0x40 -- 2.1.4

4 3

[alsa-devel] [PATCH] ASoC: qcom: move clock names into LPASS variant struct
by Linus Walleij 21 Apr '17

21 Apr '17

The clock names for the two supported codecs are either "mi2s-*" name variants generated by code. This naming scheme does not work for platforms like MSM8660 which has I2S channels named CODEC_I2S_SPKR (rather than just "MI2S tertiary" and other repetitive names) and consequently have clocks named "codec-i2s-spkr-osr-clk" and similar. Skip the runtime generation of clock names and replace it with name lookup tables encoded into the variant data. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 12 ++++++++++++ sound/soc/qcom/lpass-cpu.c | 22 +++++++--------------- sound/soc/qcom/lpass-ipq806x.c | 6 ++++++ sound/soc/qcom/lpass.h | 2 ++ 4 files changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 8aed72be3224..8a74844d99e2 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -231,6 +231,18 @@ static struct lpass_variant apq8016_data = { .wrdma_channels = 2, .dai_driver = apq8016_lpass_cpu_dai_driver, .num_dai = ARRAY_SIZE(apq8016_lpass_cpu_dai_driver), + .dai_osr_clk_names = (const char *[]) { + "mi2s-osr-clk0", + "mi2s-osr-clk1", + "mi2s-osr-clk2", + "mi2s-osr-clk3", + }, + .dai_bit_clk_names = (const char *[]) { + "mi2s-bit-clk0", + "mi2s-bit-clk1", + "mi2s-bit-clk2", + "mi2s-bit-clk3", + }, .init = apq8016_lpass_init, .exit = apq8016_lpass_exit, .alloc_dma_channel = apq8016_lpass_alloc_dma_channel, diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5202a584e0c6..292b103abada 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -429,7 +429,6 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) struct lpass_variant *variant; struct device *dev = &pdev->dev; const struct of_device_id *match; - char clk_name[16]; int ret, i, dai_id; dsp_of_node = of_parse_phandle(pdev->dev.of_node, "qcom,adsp", 0); @@ -477,31 +476,24 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) for (i = 0; i < variant->num_dai; i++) { dai_id = variant->dai_driver[i].id; - if (variant->num_dai > 1) - sprintf(clk_name, "mi2s-osr-clk%d", i); - else - sprintf(clk_name, "mi2s-osr-clk"); - drvdata->mi2s_osr_clk[dai_id] = devm_clk_get(&pdev->dev, - clk_name); + variant->dai_osr_clk_names[i]); if (IS_ERR(drvdata->mi2s_osr_clk[dai_id])) { dev_warn(&pdev->dev, - "error getting optional mi2s-osr-clk: %ld\n", + "%s() error getting optional %s: %ld\n", + __func__, + variant->dai_osr_clk_names[i], PTR_ERR(drvdata->mi2s_osr_clk[dai_id])); drvdata->mi2s_osr_clk[dai_id] = NULL; } - if (variant->num_dai > 1) - sprintf(clk_name, "mi2s-bit-clk%d", i); - else - sprintf(clk_name, "mi2s-bit-clk"); - drvdata->mi2s_bit_clk[dai_id] = devm_clk_get(&pdev->dev, - clk_name); + variant->dai_bit_clk_names[i]); if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) { dev_err(&pdev->dev, - "error getting mi2s-bit-clk: %ld\n", + "error getting %s: %ld\n", + variant->dai_bit_clk_names[i], PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 608c1a92af8a..ca1e1f2d2787 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -92,6 +92,12 @@ static struct lpass_variant ipq806x_data = { .wrdma_channels = 4, .dai_driver = &ipq806x_lpass_cpu_dai_driver, .num_dai = 1, + .dai_osr_clk_names = (const char *[]) { + "mi2s-osr-clk", + }, + .dai_bit_clk_names = (const char *[]) { + "mi2s-bit-clk", + }, .alloc_dma_channel = ipq806x_lpass_alloc_dma_channel, .free_dma_channel = ipq806x_lpass_free_dma_channel, }; diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 9b031352ea3c..b848db2d6c3d 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -91,6 +91,8 @@ struct lpass_variant { /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; int num_dai; + const char * const *dai_osr_clk_names; + const char * const *dai_bit_clk_names; }; /* register the platform driver from the CPU DAI driver */ -- 2.9.3

2 1

[alsa-devel] [alsa-lib][PATCH v2 0/3] timer/hwdep: update included headers with UAPI headers in recent kernel
by Takashi Sakamoto 21 Apr '17

21 Apr '17

Hi, In this subsystem, interfaces between kernel/userspace are described in UAPI headers of kernel source. User space library (alsa-lib) includes this header as its code base. When developers update UAPI headers in development cycle of kernel, corresponding headers in alsa-lib should be updated synchronously. The purpose of this patchset is to catch up recent UAPI header for userspace library. There're missing changes added at v4.7, v4.9 and v4.12 development period. v2 changes: - Update commit description. - Reorder commits so that aligned according to date of corresponding code change. Takashi Sakamoto (3): timer: obsolete legacy rtctimer instance hwdep: add Line6 USB series support hwdep: add support for MOTU FireWire series and RME Fireface series include/hwdep.h | 5 ++++- include/sound/asound.h | 7 +++++-- include/timer.h | 2 +- 3 files changed, 10 insertions(+), 4 deletions(-) -- 2.9.3

2 4

[alsa-devel] [alsa-lib][PATCH 0/3] hwdep: update included headers with UAPI headers in recent kernel
by Takashi Sakamoto 21 Apr '17

21 Apr '17

Hi, In this subsystem, interfaces between kernel/userspace are described in UAPI headers of kernel source. User space library (alsa-lib) includes this header as its code base. When developers update UAPI headers in development cycle of kernel, corresponding headers in alsa-lib should be updated. The purpose of this patchset is to catch up recent UAPI header for userspace library. There're missing changes added at v4.7, v4.9 and v4.12 development period. Takashi Sakamoto (3): hwdep: add Line6 USB series support hwdep: add support for MOTU FireWire series and RME Fireface series timer: obsolete legacy rtctimer instance include/hwdep.h | 5 ++++- include/sound/asound.h | 7 +++++-- include/timer.h | 2 +- 3 files changed, 10 insertions(+), 4 deletions(-) -- 2.9.3

1 4

[alsa-devel] [PATCH 0/4] topology: Support using "index" to group objects when generating binary for kernel
by fuweix.tang＠intel.com 21 Apr '17

21 Apr '17

From: Fuwei Tang <fuweix.tang(a)intel.com> The index value of an topology object can be used to group objects for different use cases. And so the audio kernel driver can ask topology core to load all objects of the specified index value(use case). The function is already supported in kernel. So in the user space, for topology objects of the same type but with different index values, they should be written into different blocks in the binary for kernel. The index value is stored in the block header. Fuwei Tang (4): topology: Insert new element based on its index value topology: Remove code parsing index value in paring each object topology: Look up references for an object based on its index topology: Group elements with the same index value into a block include/topology.h | 10 +++-- src/topology/builder.c | 94 +++++++++++++++++++++++++---------------------- src/topology/ctl.c | 31 +--------------- src/topology/dapm.c | 27 +++++--------- src/topology/data.c | 15 ++------ src/topology/elem.c | 74 +++++++++++++++++++++++++++---------- src/topology/pcm.c | 58 ++++++----------------------- src/topology/tplg_local.h | 3 +- 8 files changed, 141 insertions(+), 171 deletions(-) -- 2.7.4

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix headset microphone detection for ASUS N551 and N751
by Takashi Iwai 21 Apr '17

21 Apr '17

From: Mikhail Paulyshka <me(a)mixaill.tk> Headset microphone does not work out of the box on ASUS Nx51 laptops. This patch fixes it. Patch tested on Asus N551 laptop. Asus N751 part is not tested, but according to [1] this laptop uses the same audiosystem. 1. https://bugzilla.kernel.org/show_bug.cgi?id=117781 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195437 Signed-off-by: Mikhail Paulyshka <me(a)mixaill.tk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9371f1a95b33..05a26fba5ef2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6815,6 +6815,7 @@ enum { ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE, ALC668_FIXUP_ASUS_Nx51, ALC891_FIXUP_HEADSET_MODE, ALC891_FIXUP_DELL_MIC_NO_PRESENCE, @@ -7068,14 +7069,21 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_BASS_1A }, + [ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc668, + .chain_id = ALC662_FIXUP_BASS_CHMAP + }, [ALC668_FIXUP_ASUS_Nx51] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - {0x1a, 0x90170151}, /* bass speaker */ + { 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { 0x1a, 0x90170151 }, /* bass speaker */ + { 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */ {} }, .chained = true, - .chain_id = ALC662_FIXUP_BASS_CHMAP, + .chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE, }, [ALC891_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, -- 2.11.1

1 0

[alsa-devel] [PATCH v2] ASoC: rsnd: Fix possible NULL pointer dereference
by Kuninori Morimoto 21 Apr '17

21 Apr '17

From: Marek Vasut <marek.vasut+renesas(a)gmail.com> 25165f79adc76b812bfb4d8f2ab120aafb28d0e6 ("ASoC: rsnd: enable clock-frequency for both 44.1kHz/48kHz") supports both 44.1kHz/48kHz clock-frequency settings for ADG which will be used for AUDIO_OLKOUTn. But some board doesn't need it, thus, it is not mandatory. But, above patch didn't care about the case of "clock-frequency" DT property was not present. This patch ignores ADG settings if AUDIO_OLKOUTn was not used. Signed-off-by: Marek Vasut <marek.vasut+renesas(a)gmail.com> [Kuninori: tidyup not to break non AUDIO_OLKOUTn case] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Marek I think this patch solves your issue sound/soc/sh/rcar/adg.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 96fef91..277774c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -453,13 +453,18 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, [CLKI] = 0x2, }; - of_property_read_u32(np, "#clock-cells", &count); + ckr = 0; + rbga = 2; /* default 1/6 */ + rbgb = 2; /* default 1/6 */ /* * ADG supports BRRA/BRRB output only * this means all clkout0/1/2/3 will be same rate */ prop = of_find_property(np, "clock-frequency", NULL);; + if (!prop) + goto rsnd_adg_get_clkout_end; + req_size = prop->length / sizeof(u32); of_property_read_u32_array(np, "clock-frequency", req_rate, req_size); @@ -472,6 +477,9 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, req_48kHz_rate = req_rate[i]; } + if (req_rate[0] % 48000 == 0) + adg->flags = AUDIO_OUT_48; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. @@ -481,9 +489,6 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * rsnd_adg_ssi_clk_try_start() * rsnd_ssi_master_clk_start() */ - ckr = 0; - rbga = 2; /* default 1/6 */ - rbgb = 2; /* default 1/6 */ adg->rbga_rate_for_441khz = 0; adg->rbgb_rate_for_48khz = 0; for_each_rsnd_clk(clk, adg, i) { @@ -528,6 +533,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * this means all clkout0/1/2/3 will be * same rate */ + of_property_read_u32(np, "#clock-cells", &count); /* * for clkout */ @@ -557,13 +563,11 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, &adg->onecell); } +rsnd_adg_get_clkout_end: adg->ckr = ckr; adg->rbga = rbga; adg->rbgb = rbgb; - if (req_rate[0] % 48000 == 0) - adg->flags = AUDIO_OUT_48; - for_each_rsnd_clkout(clk, adg, i) dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: Fix possible NULL pointer dereference
by Marek Vasut 21 Apr '17

21 Apr '17

In case the "clock-frequency" DT property is not present, the of_find_property(np, "clock-frequency", NULL) will return NULL and the subsequent req_size = prop->length / sizeof(u32); will trigger a NULL pointer dereference. This patch adds check for the NULL return value and propagates the error into the caller of the function. Signed-off-by: Marek Vasut <marek.vasut+renesas(a)gmail.com> Cc: Geert Uytterhoeven <geert+renesas(a)glider.be> Cc: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linux-renesas-soc(a)vger.kernel.org --- sound/soc/sh/rcar/adg.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 96fef91b480c..a7cbe7e89dfc 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -425,8 +425,7 @@ static void rsnd_adg_get_clkin(struct rsnd_priv *priv, dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); } -static void rsnd_adg_get_clkout(struct rsnd_priv *priv, - struct rsnd_adg *adg) +static int rsnd_adg_get_clkout(struct rsnd_priv *priv, struct rsnd_adg *adg) { struct clk *clk; struct device *dev = rsnd_priv_to_dev(priv); @@ -459,7 +458,10 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * ADG supports BRRA/BRRB output only * this means all clkout0/1/2/3 will be same rate */ - prop = of_find_property(np, "clock-frequency", NULL);; + prop = of_find_property(np, "clock-frequency", NULL); + if (!prop) + return -EINVAL; + req_size = prop->length / sizeof(u32); of_property_read_u32_array(np, "clock-frequency", req_rate, req_size); @@ -568,6 +570,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", ckr, rbga, rbgb); + return 0; } int rsnd_adg_probe(struct rsnd_priv *priv) @@ -589,7 +592,9 @@ int rsnd_adg_probe(struct rsnd_priv *priv) return ret; rsnd_adg_get_clkin(priv, adg); - rsnd_adg_get_clkout(priv, adg); + ret = rsnd_adg_get_clkout(priv, adg); + if (ret) + return ret; if (of_get_property(np, "clkout-lr-asynchronous", NULL)) adg->flags = LRCLK_ASYNC; -- 2.11.0

2 2

[alsa-devel] [PATCH] ALSA: firewire-motu: mark trace helpers as __maybe_unused
by Arnd Bergmann 20 Apr '17

20 Apr '17

Two functions were introduced for the purpose of tracing but cause warnings when tracing is disabled: sound/firewire/motu/amdtp-motu.c:284:13: error: 'copy_message' defined but not used [-Werror=unused-function] static void copy_message(u64 *frames, __be32 *buffer, unsigned int data_blocks, sound/firewire/motu/amdtp-motu.c:271:13: error: 'copy_sph' defined but not used [-Werror=unused-function] static void copy_sph(u32 *frames, __be32 *buffer, unsigned int data_blocks, Marking them as __maybe_unused will do the right thing here. Fixes: 17909c1b3058 ("ALSA: firewire-motu: add tracepoints for SPH in IEC 61883-1 fashion") Fixes: c6b0b9e65f09 ("ALSA: firewire-motu: add tracepoints for messages for unique protocol") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/firewire/motu/amdtp-motu.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 996b5f818918..96f0091144bb 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -268,8 +268,9 @@ static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer, } /* For tracepoints. */ -static void copy_sph(u32 *frames, __be32 *buffer, unsigned int data_blocks, - unsigned int data_block_quadlets) +static void __maybe_unused copy_sph(u32 *frames, __be32 *buffer, + unsigned int data_blocks, + unsigned int data_block_quadlets) { unsigned int i; @@ -281,8 +282,9 @@ static void copy_sph(u32 *frames, __be32 *buffer, unsigned int data_blocks, } /* For tracepoints. */ -static void copy_message(u64 *frames, __be32 *buffer, unsigned int data_blocks, - unsigned int data_block_quadlets) +static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, + unsigned int data_blocks, + unsigned int data_block_quadlets) { unsigned int i; -- 2.9.0

3 3

[alsa-devel] [PATCH] ASoC: fsl_asrc_dma: use correct direction enum type
by Stefan Agner 20 Apr '17

20 Apr '17

The direction argument is of type enum dma_transfer_direction, and not enum dma_data_direction. The enumeration values are the same so this did not had an effect in practise. Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/fsl/fsl_asrc_dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index dc30d780f874..282d841840b1 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -76,7 +76,7 @@ static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream) pair->dma_chan[!dir], runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), - dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags); + dir == OUT ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, flags); if (!pair->desc[!dir]) { dev_err(dev, "failed to prepare slave DMA for Front-End\n"); return -ENOMEM; -- 2.12.2

2 1

[alsa-devel] [PATCH v2] clk: x86: add "mclk" alias for Baytrail/Cherrytrail
by Pierre-Louis Bossart 19 Apr '17

19 Apr '17

Due to timing requirements, TI and Conexant manage the audio reference clock from their ASoC codec drivers using the "mclk" string. This patch adds another lookup for the "pmc_plt_clk_3" clock to avoid Intel-specific tests in those codec drivers and use code as-is. To avoid a leak, clk_add_alias() is not used in this patch. Instead the lookup is created manually as part of the .probe() step and dropped in the .remove() step. "pmc_plt_clk_3" is used exclusively for audio on all known Baytrail/CherryTrail designs and is e.g. routed on the MCLK (pin 26) of the MinnowBoardMAX Turbot LSE connector. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/clk/x86/clk-pmc-atom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/clk/x86/clk-pmc-atom.c b/drivers/clk/x86/clk-pmc-atom.c index 2b60577..fafc991 100644 --- a/drivers/clk/x86/clk-pmc-atom.c +++ b/drivers/clk/x86/clk-pmc-atom.c @@ -54,6 +54,7 @@ struct clk_plt_data { struct clk_plt_fixed **parents; u8 nparents; struct clk_plt *clks[PMC_CLK_NUM]; + struct clk_lookup *mclk_lookup; }; /* Return an index in parent table */ @@ -337,6 +338,11 @@ static int plt_clk_probe(struct platform_device *pdev) goto err_unreg_clk_plt; } } + data->mclk_lookup = clkdev_hw_create(&data->clks[3]->hw, "mclk", NULL); + if (IS_ERR(data->mclk_lookup)) { + err = PTR_ERR(data->mclk_lookup); + goto err_unreg_clk_plt; + } plt_clk_free_parent_names_loop(parent_names, data->nparents); @@ -356,6 +362,7 @@ static int plt_clk_remove(struct platform_device *pdev) data = platform_get_drvdata(pdev); + clkdev_drop(data->mclk_lookup); plt_clk_unregister_loop(data, PMC_CLK_NUM); plt_clk_unregister_parents(data); return 0; -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: ak4613: use snd_soc_update_bits() to avoid Reserve bit on I/O CTRL
by Kuninori Morimoto 19 Apr '17

19 Apr '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index e819dd8..b2dfdde 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -75,6 +75,12 @@ #define DFS_DOUBLE_SPEED (1 << 2) #define DFS_QUAD_SPEED (2 << 2) +/* ICTRL */ +#define ICTRL_MASK (0x3) + +/* OCTRL */ +#define OCTRL_MASK (0x3F) + struct ak4613_formats { unsigned int width; unsigned int fmt; @@ -365,8 +371,8 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); snd_soc_update_bits(codec, CTRL2, DFS_MASK, ctrl2); - snd_soc_write(codec, ICTRL, priv->ic); - snd_soc_write(codec, OCTRL, priv->oc); + snd_soc_update_bits(codec, ICTRL, ICTRL_MASK, priv->ic); + snd_soc_update_bits(codec, OCTRL, OCTRL_MASK, priv->oc); hw_params_end: if (ret < 0) -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: rsnd_mod_make_sure() is not under DEBUG
by Kuninori Morimoto 19 Apr '17

19 Apr '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_mod_make_sure() will be used any situation, thus, under DEBUG is not realistic. This patch move it to non DEBUG area Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 -- sound/soc/sh/rcar/rsnd.h | 2 +- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8b0a16a..e36a6b5 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -110,7 +110,6 @@ /* * rsnd_mod functions */ -#ifdef DEBUG void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type) { if (mod->type != type) { @@ -121,7 +120,6 @@ void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type) rsnd_mod_name(mod), rsnd_mod_id(mod)); } } -#endif char *rsnd_mod_name(struct rsnd_mod *mod) { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc6b46b..af6d57b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -740,8 +740,8 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); struct rsnd_mod *rsnd_cmd_mod_get(struct rsnd_priv *priv, int id); -#ifdef DEBUG void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); +#ifdef DEBUG #define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) #define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) #define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: enable clock-frequency for both 44.1kHz/48kHz
by Kuninori Morimoto 19 Apr '17

19 Apr '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current clock-frequency allows only 1 clock, but ADG can handle both 44.1kHz/48kHz base clocks. This patch enables these. On Salvator-X board, AUDIO_CLKOUT which is generated by ADG is connected to ak4613 MCKI, and it should be synchronized with LRCK. Thus, we need both 44.1kHz/48kHz base clock-frequency. Otherwise, either one sounds strange in high frequency sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 39 ++++++++++++++++++++++++++------------- 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 214a9ce..96fef91 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -43,6 +43,7 @@ struct rsnd_adg { }; #define LRCLK_ASYNC (1 << 0) +#define AUDIO_OUT_48 (1 << 1) #define adg_mode_flags(adg) (adg->flags) #define for_each_rsnd_clk(pos, adg, i) \ @@ -364,7 +365,10 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) rsnd_adg_set_ssi_clk(ssi_mod, data); - if (!(adg_mode_flags(adg) & LRCLK_ASYNC)) { + if (adg_mode_flags(adg) & LRCLK_ASYNC) { + if (adg_mode_flags(adg) & AUDIO_OUT_48) + ckr = 0x80000000; + } else { if (0 == (rate % 8000)) ckr = 0x80000000; } @@ -427,11 +431,14 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, struct clk *clk; struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; + struct property *prop; u32 ckr, rbgx, rbga, rbgb; - u32 rate, req_rate = 0, div; + u32 rate, div; +#define REQ_SIZE 2 + u32 req_rate[REQ_SIZE] = {}; uint32_t count = 0; unsigned long req_48kHz_rate, req_441kHz_rate; - int i; + int i, req_size; const char *parent_clk_name = NULL; static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", @@ -452,13 +459,18 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * ADG supports BRRA/BRRB output only * this means all clkout0/1/2/3 will be same rate */ - of_property_read_u32(np, "clock-frequency", &req_rate); + prop = of_find_property(np, "clock-frequency", NULL);; + req_size = prop->length / sizeof(u32); + + of_property_read_u32_array(np, "clock-frequency", req_rate, req_size); req_48kHz_rate = 0; req_441kHz_rate = 0; - if (0 == (req_rate % 44100)) - req_441kHz_rate = req_rate; - if (0 == (req_rate % 48000)) - req_48kHz_rate = req_rate; + for (i = 0; i < req_size; i++) { + if (0 == (req_rate[i] % 44100)) + req_441kHz_rate = req_rate[i]; + if (0 == (req_rate[i] % 48000)) + req_48kHz_rate = req_rate[i]; + } /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC @@ -505,10 +517,8 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, rbgb = rbgx; adg->rbgb_rate_for_48khz = rate / div; ckr |= brg_table[i] << 16; - if (req_48kHz_rate) { + if (req_48kHz_rate) parent_clk_name = __clk_get_name(clk); - ckr |= 0x80000000; - } } } } @@ -523,7 +533,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, 0, req_rate); + parent_clk_name, 0, req_rate[0]); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -536,7 +546,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, - req_rate); + req_rate[0]); adg->clkout[i] = ERR_PTR(-ENOENT); if (!IS_ERR(clk)) adg->clkout[i] = clk; @@ -551,6 +561,9 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->rbga = rbga; adg->rbgb = rbgb; + if (req_rate[0] % 48000 == 0) + adg->flags = AUDIO_OUT_48; + for_each_rsnd_clkout(clk, adg, i) dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", -- 1.9.1

2 1

[alsa-devel] [resend][PATCH] ALSA: hda - Add HP ZBook 15u G3 Conexant CX20724 GPIO mute leds
by jeronimo＠borque.com.ar 19 Apr '17

19 Apr '17

From: Jerónimo Borque <jeronimo(a)borque.com.ar> The HP ZBook 15u G3 has a Conexant CX20724 with mute led on GPIO1 and mic mute led on GPIO2. Adding CXT_FIXUP_MUTE_LED_GPIO inspired on patch_realtek's one. Signed-off-by: Jerónimo Borque <jeronimo(a)borque.com.ar> --- sound/pci/hda/patch_conexant.c | 81 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 81 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 69266b8..e825373 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -52,6 +52,12 @@ struct conexant_spec { bool dc_enable; unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */ struct nid_path *dc_mode_path; + + int mute_led_polarity; + unsigned int gpio_led; + unsigned int gpio_mute_led_mask; + unsigned int gpio_mic_led_mask; + }; @@ -264,6 +270,7 @@ enum { CXT_FIXUP_HP_DOCK, CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, + CXT_FIXUP_MUTE_LED_GPIO, }; /* for hda_fixup_thinkpad_acpi() */ @@ -646,6 +653,74 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, snd_hda_jack_set_gating_jack(codec, 0x19, 0x16); } +/* update LED status via GPIO */ +static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, + bool enabled) +{ + struct conexant_spec *spec = codec->spec; + unsigned int oldval = spec->gpio_led; + + if (spec->mute_led_polarity) + enabled = !enabled; + + if (enabled) + spec->gpio_led &= ~mask; + else + spec->gpio_led |= mask; + if (spec->gpio_led != oldval) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); +} + +/* turn on/off mute LED via GPIO per vmaster hook */ +static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct conexant_spec *spec = codec->spec; + + cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); +} + +/* turn on/off mic-mute LED via GPIO per capture hook */ +static void cxt_fixup_gpio_mic_mute_hook(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct conexant_spec *spec = codec->spec; + + if (ucontrol) + cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, + ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]); +} + + +static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x03 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, + {} + }; + codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; + spec->gen.cap_sync_hook = cxt_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x01; + spec->gpio_mic_led_mask = 0x02; + } + snd_hda_add_verbs(codec, gpio_init); + if (spec->gpio_led) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -799,6 +874,10 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_hp_gate_mic_jack, }, + [CXT_FIXUP_MUTE_LED_GPIO] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_mute_led_gpio, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -851,6 +930,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), + SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), @@ -882,6 +962,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, + { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, {} }; -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix dereference before NULL check on ebus
by Colin King 19 Apr '17

19 Apr '17

From: Colin Ian King <colin.king(a)canonical.com> ebus is being NULL checked however it is being dereferenced earlier on the assignment bus = ebus_to_hbus(ebus). Fix this by moving the assignment to after the NULL check on ebus to avoid a NULL pointer dereference error. Fixes: c5a76a246989c8 ("ASoC: Intel: Skylake: Add shutdown callback") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 0c57d4eaae3a..21369f60fbaa 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -818,7 +818,7 @@ static int skl_probe(struct pci_dev *pci, static void skl_shutdown(struct pci_dev *pci) { struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus; struct hdac_stream *s; struct hdac_ext_stream *stream; struct skl *skl; @@ -826,6 +826,7 @@ static void skl_shutdown(struct pci_dev *pci) if (ebus == NULL) return; + bus = ebus_to_hbus(ebus); skl = ebus_to_skl(ebus); if (skl->init_failed) -- 2.11.0

3 2

[alsa-devel] [sound:topic/gpd-win-jack-fix 1/4] sound/soc//codecs/rt5645.c:3252:4: error: break statement not within loop or switch
by kbuild test robot 19 Apr '17

19 Apr '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/gpd-win-jack-fix head: ba696ca0c6864a74842e0534cf5a26c38b55172e commit: 867babbf447f6c5ad188b7245f63e96f1905ff00 [1/4] ASoC: rt5645: read jd1_1 status for jd detection config: arm-multi_v7_defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 867babbf447f6c5ad188b7245f63e96f1905ff00 # save the attached .config to linux build tree make.cross ARCH=arm Note: the sound/topic/gpd-win-jack-fix HEAD ba696ca0c6864a74842e0534cf5a26c38b55172e builds fine. It only hurts bisectibility. All errors (new ones prefixed by >>): sound/soc//codecs/rt5645.c: In function 'rt5645_jack_detect_work': >> sound/soc//codecs/rt5645.c:3252:4: error: break statement not within loop or switch break; ^~~~~ vim +3252 sound/soc//codecs/rt5645.c 6e747d53 Bard Liao 2015-04-28 3236 snd_soc_jack_report(rt5645->hp_jack, 6e747d53 Bard Liao 2015-04-28 3237 report, SND_JACK_HEADPHONE); 6e747d53 Bard Liao 2015-04-28 3238 snd_soc_jack_report(rt5645->mic_jack, 6e747d53 Bard Liao 2015-04-28 3239 report, SND_JACK_MICROPHONE); f312bc59 Nicolas Boichat 2015-07-14 3240 return; 867babbf Bard Liao 2017-03-20 3241 default: /* read rt5645 jd1_1 status */ 867babbf Bard Liao 2017-03-20 3242 val = snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x1000; 6e747d53 Bard Liao 2015-04-28 3243 break; 6e747d53 Bard Liao 2015-04-28 3244 6e747d53 Bard Liao 2015-04-28 3245 } 6e747d53 Bard Liao 2015-04-28 3246 867babbf Bard Liao 2017-03-20 3247 if (!val) { 6e747d53 Bard Liao 2015-04-28 3248 /* jack in */ 6e747d53 Bard Liao 2015-04-28 3249 if (rt5645->jack_type == 0) { 6e747d53 Bard Liao 2015-04-28 3250 report = rt5645_jack_detect(rt5645->codec, 1); 6e747d53 Bard Liao 2015-04-28 3251 /* for push button and jack out */ 6e747d53 Bard Liao 2015-04-28 @3252 break; 6e747d53 Bard Liao 2015-04-28 3253 } 6e747d53 Bard Liao 2015-04-28 3254 btn_type = 0; 6e747d53 Bard Liao 2015-04-28 3255 if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { 6e747d53 Bard Liao 2015-04-28 3256 /* button pressed */ 6e747d53 Bard Liao 2015-04-28 3257 report = SND_JACK_HEADSET; 6e747d53 Bard Liao 2015-04-28 3258 btn_type = rt5645_button_detect(rt5645->codec); 6e747d53 Bard Liao 2015-04-28 3259 /* rt5650 can report three kinds of button behavior, 6e747d53 Bard Liao 2015-04-28 3260 one click, double click and hold. However, :::::: The code at line 3252 was first introduced by commit :::::: 6e747d5311fc67b5fe7e2d7d242329c1bdff3318 ASoC: rt5645: Adds push button support for rt5650 :::::: TO: Bard Liao <bardliao(a)realtek.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] ALSA: hda - Add HP ZBook 15u G3 Conexant CX20724 GPIO mute leds
by Jerónimo Borque 19 Apr '17

19 Apr '17

The HP ZBook 15u G3 has a Conexant CX20724 with mute led on GPIO1 and mic mute led on GPIO2. Adding CXT_FIXUP_MUTE_LED_GPIO inspired on patch_realtek's one. Signed-off-by: Jerónimo Borque <jeronimo(a)borque.com.ar> --- diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 69266b8..e825373 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -52,6 +52,12 @@ struct conexant_spec { bool dc_enable; unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */ struct nid_path *dc_mode_path; + + int mute_led_polarity; + unsigned int gpio_led; + unsigned int gpio_mute_led_mask; + unsigned int gpio_mic_led_mask; + }; @@ -264,6 +270,7 @@ enum { CXT_FIXUP_HP_DOCK, CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, + CXT_FIXUP_MUTE_LED_GPIO, }; /* for hda_fixup_thinkpad_acpi() */ @@ -646,6 +653,74 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, snd_hda_jack_set_gating_jack(codec, 0x19, 0x16); } +/* update LED status via GPIO */ +static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, + bool enabled) +{ + struct conexant_spec *spec = codec->spec; + unsigned int oldval = spec->gpio_led; + + if (spec->mute_led_polarity) + enabled = !enabled; + + if (enabled) + spec->gpio_led &= ~mask; + else + spec->gpio_led |= mask; + if (spec->gpio_led != oldval) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); +} + +/* turn on/off mute LED via GPIO per vmaster hook */ +static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct conexant_spec *spec = codec->spec; + + cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); +} + +/* turn on/off mic-mute LED via GPIO per capture hook */ +static void cxt_fixup_gpio_mic_mute_hook(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct conexant_spec *spec = codec->spec; + + if (ucontrol) + cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, + ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]); +} + + +static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x03 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, + {} + }; + codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; + spec->gen.cap_sync_hook = cxt_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x01; + spec->gpio_mic_led_mask = 0x02; + } + snd_hda_add_verbs(codec, gpio_init); + if (spec->gpio_led) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -799,6 +874,10 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_hp_gate_mic_jack, }, + [CXT_FIXUP_MUTE_LED_GPIO] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_mute_led_gpio, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -851,6 +930,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), + SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), @@ -882,6 +962,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, + { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, {} };

2 1

[alsa-devel] [PATCH v2] ASoC: rsnd: control kctrl items acceptance anytime/runtime
by Kuninori Morimoto 19 Apr '17

19 Apr '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SRC/DVC/CTU adds kctrl for each device, and SRC can adjust its sampling rate during playback, thus, this feature should be enabled only *during* playback. This patch controls it more clearly Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - tidyup log comment sound/soc/sh/rcar/core.c | 17 +++++++++++++++++ sound/soc/sh/rcar/ctu.c | 6 ++++++ sound/soc/sh/rcar/dvc.c | 5 +++++ sound/soc/sh/rcar/rsnd.h | 16 ++++++++++------ sound/soc/sh/rcar/src.c | 2 ++ 5 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f8eb9d3..92c9884 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -984,6 +984,9 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); int i, change = 0; + if (!cfg->accept(cfg->io)) + return 0; + for (i = 0; i < cfg->size; i++) { if (cfg->texts) { change |= (uc->value.enumerated.item[i] != cfg->val[i]); @@ -1000,6 +1003,18 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, return change; } +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) +{ + return 1; +} + +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + return !!runtime; +} + struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; @@ -1018,6 +1033,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -1052,6 +1068,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->texts = texts; cfg->max = max; cfg->size = size; + cfg->accept = accept; cfg->update = update; cfg->card = card; cfg->kctrl = kctrl; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 9dcc1f9..4ba8f2f 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -279,12 +279,14 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* CTU Pass */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + rsnd_kctrl_accept_anytime, NULL, &ctu->pass, RSND_MAX_CHANNELS, 0xC); /* ROW0 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv0, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -293,6 +295,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW1 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv1, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -301,6 +304,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW2 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv2, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -309,6 +313,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW3 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv3, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -317,6 +322,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* Reset */ ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_kctrl_accept_anytime, rsnd_ctu_value_reset, &ctu->reset, 1); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 463de83..75af6e7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -257,6 +257,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->volume, slots, 0x00800000 - 1); @@ -267,6 +268,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->mute, slots, 1); @@ -277,6 +279,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_s(mod, io, rtd, is_play ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) @@ -285,6 +288,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rup, dvc_ramp_rate); @@ -294,6 +298,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rdown, dvc_ramp_rate); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3dc9e06..b4d3bc0 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -594,6 +594,7 @@ struct rsnd_kctrl_cfg { unsigned int size; u32 *val; const char * const *texts; + int (*accept)(struct rsnd_dai_stream *io); void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod); struct rsnd_dai_stream *io; struct snd_card *card; @@ -611,12 +612,15 @@ struct rsnd_kctrl_cfg_s { u32 val; }; +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -624,16 +628,16 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int size, u32 max); -#define rsnd_kctrl_new_m(mod, io, rtd, name, update, cfg, size, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_m(cfg), \ +#define rsnd_kctrl_new_m(mod, io, rtd, name, accept, update, cfg, size, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_m(cfg), \ NULL, size, max) -#define rsnd_kctrl_new_s(mod, io, rtd, name, update, cfg, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_s(mod, io, rtd, name, accept, update, cfg, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ NULL, 1, max) -#define rsnd_kctrl_new_e(mod, io, rtd, name, update, cfg, texts) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_e(mod, io, rtd, name, accept, update, cfg, texts) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ texts, 1, ARRAY_SIZE(texts)) /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 20b5b2e..8be0415 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -507,6 +507,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate Switch" : "SRC In Rate Switch", + rsnd_kctrl_accept_anytime, rsnd_src_set_convert_rate, &src->sen, 1); if (ret < 0) @@ -516,6 +517,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", + rsnd_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); -- 1.9.1

1 0

[alsa-devel] Applied "ASoC: nau8540: fix tab conversion problem" to the asoc tree
by Mark Brown 18 Apr '17

18 Apr '17

The patch ASoC: nau8540: fix tab conversion problem 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 babd658503a69c423aeab440f7d385f03ac54f7e Mon Sep 17 00:00:00 2001 From: John Hsu <supercraig0719(a)gmail.com> Date: Tue, 18 Apr 2017 10:32:04 +0800 Subject: [PATCH] ASoC: nau8540: fix tab conversion problem Fix the tab converting to space problem. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8540.c | 1224 ++++++++++++++++++++++---------------------- sound/soc/codecs/nau8540.h | 310 +++++------ 2 files changed, 767 insertions(+), 767 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 9e8f0f4aa51a..c8bcb1db966d 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -39,147 +39,147 @@ /* scaling for mclk from sysclk_src output */ static const struct nau8540_fll_attr mclk_src_scaling[] = { - { 1, 0x0 }, - { 2, 0x2 }, - { 4, 0x3 }, - { 8, 0x4 }, - { 16, 0x5 }, - { 32, 0x6 }, - { 3, 0x7 }, - { 6, 0xa }, - { 12, 0xb }, - { 24, 0xc }, + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, }; /* ratio for input clk freq */ static const struct nau8540_fll_attr fll_ratio[] = { - { 512000, 0x01 }, - { 256000, 0x02 }, - { 128000, 0x04 }, - { 64000, 0x08 }, - { 32000, 0x10 }, - { 8000, 0x20 }, - { 4000, 0x40 }, + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, }; static const struct nau8540_fll_attr fll_pre_scalar[] = { - { 1, 0x0 }, - { 2, 0x1 }, - { 4, 0x2 }, - { 8, 0x3 }, + { 1, 0x0 }, + { 2, 0x1 }, + { 4, 0x2 }, + { 8, 0x3 }, }; /* over sampling rate */ static const struct nau8540_osr_attr osr_adc_sel[] = { - { 32, 3 }, /* OSR 32, SRC 1/8 */ - { 64, 2 }, /* OSR 64, SRC 1/4 */ - { 128, 1 }, /* OSR 128, SRC 1/2 */ - { 256, 0 }, /* OSR 256, SRC 1 */ + { 32, 3 }, /* OSR 32, SRC 1/8 */ + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 256, 0 }, /* OSR 256, SRC 1 */ }; static const struct reg_default nau8540_reg_defaults[] = { - {NAU8540_REG_POWER_MANAGEMENT, 0x0000}, - {NAU8540_REG_CLOCK_CTRL, 0x0000}, - {NAU8540_REG_CLOCK_SRC, 0x0000}, - {NAU8540_REG_FLL1, 0x0001}, - {NAU8540_REG_FLL2, 0x3126}, - {NAU8540_REG_FLL3, 0x0008}, - {NAU8540_REG_FLL4, 0x0010}, - {NAU8540_REG_FLL5, 0xC000}, - {NAU8540_REG_FLL6, 0x6000}, - {NAU8540_REG_FLL_VCO_RSV, 0xF13C}, - {NAU8540_REG_PCM_CTRL0, 0x000B}, - {NAU8540_REG_PCM_CTRL1, 0x3010}, - {NAU8540_REG_PCM_CTRL2, 0x0800}, - {NAU8540_REG_PCM_CTRL3, 0x0000}, - {NAU8540_REG_PCM_CTRL4, 0x000F}, - {NAU8540_REG_ALC_CONTROL_1, 0x0000}, - {NAU8540_REG_ALC_CONTROL_2, 0x700B}, - {NAU8540_REG_ALC_CONTROL_3, 0x0022}, - {NAU8540_REG_ALC_CONTROL_4, 0x1010}, - {NAU8540_REG_ALC_CONTROL_5, 0x1010}, - {NAU8540_REG_NOTCH_FIL1_CH1, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH1, 0x0000}, - {NAU8540_REG_NOTCH_FIL1_CH2, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH2, 0x0000}, - {NAU8540_REG_NOTCH_FIL1_CH3, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH3, 0x0000}, - {NAU8540_REG_NOTCH_FIL1_CH4, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH4, 0x0000}, - {NAU8540_REG_HPF_FILTER_CH12, 0x0000}, - {NAU8540_REG_HPF_FILTER_CH34, 0x0000}, - {NAU8540_REG_ADC_SAMPLE_RATE, 0x0002}, - {NAU8540_REG_DIGITAL_GAIN_CH1, 0x0400}, - {NAU8540_REG_DIGITAL_GAIN_CH2, 0x0400}, - {NAU8540_REG_DIGITAL_GAIN_CH3, 0x0400}, - {NAU8540_REG_DIGITAL_GAIN_CH4, 0x0400}, - {NAU8540_REG_DIGITAL_MUX, 0x00E4}, - {NAU8540_REG_GPIO_CTRL, 0x0000}, - {NAU8540_REG_MISC_CTRL, 0x0000}, - {NAU8540_REG_I2C_CTRL, 0xEFFF}, - {NAU8540_REG_VMID_CTRL, 0x0000}, - {NAU8540_REG_MUTE, 0x0000}, - {NAU8540_REG_ANALOG_ADC1, 0x0011}, - {NAU8540_REG_ANALOG_ADC2, 0x0020}, - {NAU8540_REG_ANALOG_PWR, 0x0000}, - {NAU8540_REG_MIC_BIAS, 0x0004}, - {NAU8540_REG_REFERENCE, 0x0000}, - {NAU8540_REG_FEPGA1, 0x0000}, - {NAU8540_REG_FEPGA2, 0x0000}, - {NAU8540_REG_FEPGA3, 0x0101}, - {NAU8540_REG_FEPGA4, 0x0101}, - {NAU8540_REG_PWR, 0x0000}, + {NAU8540_REG_POWER_MANAGEMENT, 0x0000}, + {NAU8540_REG_CLOCK_CTRL, 0x0000}, + {NAU8540_REG_CLOCK_SRC, 0x0000}, + {NAU8540_REG_FLL1, 0x0001}, + {NAU8540_REG_FLL2, 0x3126}, + {NAU8540_REG_FLL3, 0x0008}, + {NAU8540_REG_FLL4, 0x0010}, + {NAU8540_REG_FLL5, 0xC000}, + {NAU8540_REG_FLL6, 0x6000}, + {NAU8540_REG_FLL_VCO_RSV, 0xF13C}, + {NAU8540_REG_PCM_CTRL0, 0x000B}, + {NAU8540_REG_PCM_CTRL1, 0x3010}, + {NAU8540_REG_PCM_CTRL2, 0x0800}, + {NAU8540_REG_PCM_CTRL3, 0x0000}, + {NAU8540_REG_PCM_CTRL4, 0x000F}, + {NAU8540_REG_ALC_CONTROL_1, 0x0000}, + {NAU8540_REG_ALC_CONTROL_2, 0x700B}, + {NAU8540_REG_ALC_CONTROL_3, 0x0022}, + {NAU8540_REG_ALC_CONTROL_4, 0x1010}, + {NAU8540_REG_ALC_CONTROL_5, 0x1010}, + {NAU8540_REG_NOTCH_FIL1_CH1, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH1, 0x0000}, + {NAU8540_REG_NOTCH_FIL1_CH2, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH2, 0x0000}, + {NAU8540_REG_NOTCH_FIL1_CH3, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH3, 0x0000}, + {NAU8540_REG_NOTCH_FIL1_CH4, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH4, 0x0000}, + {NAU8540_REG_HPF_FILTER_CH12, 0x0000}, + {NAU8540_REG_HPF_FILTER_CH34, 0x0000}, + {NAU8540_REG_ADC_SAMPLE_RATE, 0x0002}, + {NAU8540_REG_DIGITAL_GAIN_CH1, 0x0400}, + {NAU8540_REG_DIGITAL_GAIN_CH2, 0x0400}, + {NAU8540_REG_DIGITAL_GAIN_CH3, 0x0400}, + {NAU8540_REG_DIGITAL_GAIN_CH4, 0x0400}, + {NAU8540_REG_DIGITAL_MUX, 0x00E4}, + {NAU8540_REG_GPIO_CTRL, 0x0000}, + {NAU8540_REG_MISC_CTRL, 0x0000}, + {NAU8540_REG_I2C_CTRL, 0xEFFF}, + {NAU8540_REG_VMID_CTRL, 0x0000}, + {NAU8540_REG_MUTE, 0x0000}, + {NAU8540_REG_ANALOG_ADC1, 0x0011}, + {NAU8540_REG_ANALOG_ADC2, 0x0020}, + {NAU8540_REG_ANALOG_PWR, 0x0000}, + {NAU8540_REG_MIC_BIAS, 0x0004}, + {NAU8540_REG_REFERENCE, 0x0000}, + {NAU8540_REG_FEPGA1, 0x0000}, + {NAU8540_REG_FEPGA2, 0x0000}, + {NAU8540_REG_FEPGA3, 0x0101}, + {NAU8540_REG_FEPGA4, 0x0101}, + {NAU8540_REG_PWR, 0x0000}, }; static bool nau8540_readable_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case NAU8540_REG_POWER_MANAGEMENT ... NAU8540_REG_FLL_VCO_RSV: - case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: - case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: - case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ADC_SAMPLE_RATE: - case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: - case NAU8540_REG_P2P_CH1 ... NAU8540_REG_I2C_CTRL: - case NAU8540_REG_I2C_DEVICE_ID: - case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: - case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: - return true; - default: - return false; - } + switch (reg) { + case NAU8540_REG_POWER_MANAGEMENT ... NAU8540_REG_FLL_VCO_RSV: + case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: + case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: + case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ADC_SAMPLE_RATE: + case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: + case NAU8540_REG_P2P_CH1 ... NAU8540_REG_I2C_CTRL: + case NAU8540_REG_I2C_DEVICE_ID: + case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: + case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: + return true; + default: + return false; + } } static bool nau8540_writeable_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case NAU8540_REG_SW_RESET ... NAU8540_REG_FLL_VCO_RSV: - case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: - case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: - case NAU8540_REG_NOTCH_FIL1_CH1 ... NAU8540_REG_ADC_SAMPLE_RATE: - case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: - case NAU8540_REG_GPIO_CTRL ... NAU8540_REG_I2C_CTRL: - case NAU8540_REG_RST: - case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: - case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: - return true; - default: - return false; - } + switch (reg) { + case NAU8540_REG_SW_RESET ... NAU8540_REG_FLL_VCO_RSV: + case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: + case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: + case NAU8540_REG_NOTCH_FIL1_CH1 ... NAU8540_REG_ADC_SAMPLE_RATE: + case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: + case NAU8540_REG_GPIO_CTRL ... NAU8540_REG_I2C_CTRL: + case NAU8540_REG_RST: + case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: + case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: + return true; + default: + return false; + } } static bool nau8540_volatile_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case NAU8540_REG_SW_RESET: - case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ALC_STATUS: - case NAU8540_REG_P2P_CH1 ... NAU8540_REG_PEAK_CH4: - case NAU8540_REG_I2C_DEVICE_ID: - case NAU8540_REG_RST: - return true; - default: - return false; - } + switch (reg) { + case NAU8540_REG_SW_RESET: + case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ALC_STATUS: + case NAU8540_REG_P2P_CH1 ... NAU8540_REG_PEAK_CH4: + case NAU8540_REG_I2C_DEVICE_ID: + case NAU8540_REG_RST: + return true; + default: + return false; + } } @@ -187,255 +187,255 @@ static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -12800, 3600); static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); static const struct snd_kcontrol_new nau8540_snd_controls[] = { - SOC_SINGLE_TLV("Mic1 Volume", NAU8540_REG_DIGITAL_GAIN_CH1, - 0, 0x520, 0, adc_vol_tlv), - SOC_SINGLE_TLV("Mic2 Volume", NAU8540_REG_DIGITAL_GAIN_CH2, - 0, 0x520, 0, adc_vol_tlv), - SOC_SINGLE_TLV("Mic3 Volume", NAU8540_REG_DIGITAL_GAIN_CH3, - 0, 0x520, 0, adc_vol_tlv), - SOC_SINGLE_TLV("Mic4 Volume", NAU8540_REG_DIGITAL_GAIN_CH4, - 0, 0x520, 0, adc_vol_tlv), - - SOC_SINGLE_TLV("Frontend PGA1 Volume", NAU8540_REG_FEPGA3, - 0, 0x25, 0, fepga_gain_tlv), - SOC_SINGLE_TLV("Frontend PGA2 Volume", NAU8540_REG_FEPGA3, - 8, 0x25, 0, fepga_gain_tlv), - SOC_SINGLE_TLV("Frontend PGA3 Volume", NAU8540_REG_FEPGA4, - 0, 0x25, 0, fepga_gain_tlv), - SOC_SINGLE_TLV("Frontend PGA4 Volume", NAU8540_REG_FEPGA4, - 8, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Mic1 Volume", NAU8540_REG_DIGITAL_GAIN_CH1, + 0, 0x520, 0, adc_vol_tlv), + SOC_SINGLE_TLV("Mic2 Volume", NAU8540_REG_DIGITAL_GAIN_CH2, + 0, 0x520, 0, adc_vol_tlv), + SOC_SINGLE_TLV("Mic3 Volume", NAU8540_REG_DIGITAL_GAIN_CH3, + 0, 0x520, 0, adc_vol_tlv), + SOC_SINGLE_TLV("Mic4 Volume", NAU8540_REG_DIGITAL_GAIN_CH4, + 0, 0x520, 0, adc_vol_tlv), + + SOC_SINGLE_TLV("Frontend PGA1 Volume", NAU8540_REG_FEPGA3, + 0, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Frontend PGA2 Volume", NAU8540_REG_FEPGA3, + 8, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Frontend PGA3 Volume", NAU8540_REG_FEPGA4, + 0, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Frontend PGA4 Volume", NAU8540_REG_FEPGA4, + 8, 0x25, 0, fepga_gain_tlv), }; static const char * const adc_channel[] = { - "ADC channel 1", "ADC channel 2", "ADC channel 3", "ADC channel 4" + "ADC channel 1", "ADC channel 2", "ADC channel 3", "ADC channel 4" }; static SOC_ENUM_SINGLE_DECL( - digital_ch4_enum, NAU8540_REG_DIGITAL_MUX, 6, adc_channel); + digital_ch4_enum, NAU8540_REG_DIGITAL_MUX, 6, adc_channel); static const struct snd_kcontrol_new digital_ch4_mux = - SOC_DAPM_ENUM("Digital CH4 Select", digital_ch4_enum); + SOC_DAPM_ENUM("Digital CH4 Select", digital_ch4_enum); static SOC_ENUM_SINGLE_DECL( - digital_ch3_enum, NAU8540_REG_DIGITAL_MUX, 4, adc_channel); + digital_ch3_enum, NAU8540_REG_DIGITAL_MUX, 4, adc_channel); static const struct snd_kcontrol_new digital_ch3_mux = - SOC_DAPM_ENUM("Digital CH3 Select", digital_ch3_enum); + SOC_DAPM_ENUM("Digital CH3 Select", digital_ch3_enum); static SOC_ENUM_SINGLE_DECL( - digital_ch2_enum, NAU8540_REG_DIGITAL_MUX, 2, adc_channel); + digital_ch2_enum, NAU8540_REG_DIGITAL_MUX, 2, adc_channel); static const struct snd_kcontrol_new digital_ch2_mux = - SOC_DAPM_ENUM("Digital CH2 Select", digital_ch2_enum); + SOC_DAPM_ENUM("Digital CH2 Select", digital_ch2_enum); static SOC_ENUM_SINGLE_DECL( - digital_ch1_enum, NAU8540_REG_DIGITAL_MUX, 0, adc_channel); + digital_ch1_enum, NAU8540_REG_DIGITAL_MUX, 0, adc_channel); static const struct snd_kcontrol_new digital_ch1_mux = - SOC_DAPM_ENUM("Digital CH1 Select", digital_ch1_enum); + SOC_DAPM_ENUM("Digital CH1 Select", digital_ch1_enum); static const struct snd_soc_dapm_widget nau8540_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("MICBIAS2", NAU8540_REG_MIC_BIAS, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MICBIAS1", NAU8540_REG_MIC_BIAS, 10, 0, NULL, 0), - - SND_SOC_DAPM_INPUT("MIC1"), - SND_SOC_DAPM_INPUT("MIC2"), - SND_SOC_DAPM_INPUT("MIC3"), - SND_SOC_DAPM_INPUT("MIC4"), - - SND_SOC_DAPM_PGA("Frontend PGA1", NAU8540_REG_PWR, 12, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA2", NAU8540_REG_PWR, 13, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA3", NAU8540_REG_PWR, 14, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA4", NAU8540_REG_PWR, 15, 0, NULL, 0), - - SND_SOC_DAPM_ADC("ADC1", NULL, - NAU8540_REG_POWER_MANAGEMENT, 0, 0), - SND_SOC_DAPM_ADC("ADC2", NULL, - NAU8540_REG_POWER_MANAGEMENT, 1, 0), - SND_SOC_DAPM_ADC("ADC3", NULL, - NAU8540_REG_POWER_MANAGEMENT, 2, 0), - SND_SOC_DAPM_ADC("ADC4", NULL, - NAU8540_REG_POWER_MANAGEMENT, 3, 0), - - SND_SOC_DAPM_PGA("ADC CH1", NAU8540_REG_ANALOG_PWR, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH2", NAU8540_REG_ANALOG_PWR, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH3", NAU8540_REG_ANALOG_PWR, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH4", NAU8540_REG_ANALOG_PWR, 3, 0, NULL, 0), - - SND_SOC_DAPM_MUX("Digital CH4 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch4_mux), - SND_SOC_DAPM_MUX("Digital CH3 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch3_mux), - SND_SOC_DAPM_MUX("Digital CH2 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch2_mux), - SND_SOC_DAPM_MUX("Digital CH1 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch1_mux), - - SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", NAU8540_REG_MIC_BIAS, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", NAU8540_REG_MIC_BIAS, 10, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_PGA("Frontend PGA1", NAU8540_REG_PWR, 12, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA2", NAU8540_REG_PWR, 13, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA3", NAU8540_REG_PWR, 14, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA4", NAU8540_REG_PWR, 15, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC1", NULL, + NAU8540_REG_POWER_MANAGEMENT, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, + NAU8540_REG_POWER_MANAGEMENT, 1, 0), + SND_SOC_DAPM_ADC("ADC3", NULL, + NAU8540_REG_POWER_MANAGEMENT, 2, 0), + SND_SOC_DAPM_ADC("ADC4", NULL, + NAU8540_REG_POWER_MANAGEMENT, 3, 0), + + SND_SOC_DAPM_PGA("ADC CH1", NAU8540_REG_ANALOG_PWR, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC CH2", NAU8540_REG_ANALOG_PWR, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC CH3", NAU8540_REG_ANALOG_PWR, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC CH4", NAU8540_REG_ANALOG_PWR, 3, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Digital CH4 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch4_mux), + SND_SOC_DAPM_MUX("Digital CH3 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch3_mux), + SND_SOC_DAPM_MUX("Digital CH2 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch2_mux), + SND_SOC_DAPM_MUX("Digital CH1 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch1_mux), + + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { - {"Frontend PGA1", NULL, "MIC1"}, - {"Frontend PGA2", NULL, "MIC2"}, - {"Frontend PGA3", NULL, "MIC3"}, - {"Frontend PGA4", NULL, "MIC4"}, - - {"ADC1", NULL, "Frontend PGA1"}, - {"ADC2", NULL, "Frontend PGA2"}, - {"ADC3", NULL, "Frontend PGA3"}, - {"ADC4", NULL, "Frontend PGA4"}, - - {"ADC CH1", NULL, "ADC1"}, - {"ADC CH2", NULL, "ADC2"}, - {"ADC CH3", NULL, "ADC3"}, - {"ADC CH4", NULL, "ADC4"}, - - {"ADC1", NULL, "MICBIAS1"}, - {"ADC2", NULL, "MICBIAS1"}, - {"ADC3", NULL, "MICBIAS2"}, - {"ADC4", NULL, "MICBIAS2"}, - - {"Digital CH1 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH1 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH1 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH1 Mux", "ADC channel 4", "ADC CH4"}, - - {"Digital CH2 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH2 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH2 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH2 Mux", "ADC channel 4", "ADC CH4"}, - - {"Digital CH3 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH3 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH3 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH3 Mux", "ADC channel 4", "ADC CH4"}, - - {"Digital CH4 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH4 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH4 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH4 Mux", "ADC channel 4", "ADC CH4"}, - - {"AIFTX", NULL, "Digital CH1 Mux"}, - {"AIFTX", NULL, "Digital CH2 Mux"}, - {"AIFTX", NULL, "Digital CH3 Mux"}, - {"AIFTX", NULL, "Digital CH4 Mux"}, + {"Frontend PGA1", NULL, "MIC1"}, + {"Frontend PGA2", NULL, "MIC2"}, + {"Frontend PGA3", NULL, "MIC3"}, + {"Frontend PGA4", NULL, "MIC4"}, + + {"ADC1", NULL, "Frontend PGA1"}, + {"ADC2", NULL, "Frontend PGA2"}, + {"ADC3", NULL, "Frontend PGA3"}, + {"ADC4", NULL, "Frontend PGA4"}, + + {"ADC CH1", NULL, "ADC1"}, + {"ADC CH2", NULL, "ADC2"}, + {"ADC CH3", NULL, "ADC3"}, + {"ADC CH4", NULL, "ADC4"}, + + {"ADC1", NULL, "MICBIAS1"}, + {"ADC2", NULL, "MICBIAS1"}, + {"ADC3", NULL, "MICBIAS2"}, + {"ADC4", NULL, "MICBIAS2"}, + + {"Digital CH1 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH1 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH1 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH1 Mux", "ADC channel 4", "ADC CH4"}, + + {"Digital CH2 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH2 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH2 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH2 Mux", "ADC channel 4", "ADC CH4"}, + + {"Digital CH3 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH3 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH3 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH3 Mux", "ADC channel 4", "ADC CH4"}, + + {"Digital CH4 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH4 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH4 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH4 Mux", "ADC channel 4", "ADC CH4"}, + + {"AIFTX", NULL, "Digital CH1 Mux"}, + {"AIFTX", NULL, "Digital CH2 Mux"}, + {"AIFTX", NULL, "Digital CH3 Mux"}, + {"AIFTX", NULL, "Digital CH4 Mux"}, }; static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) { - int osrate; + int osrate; - if (osr >= ARRAY_SIZE(osr_adc_sel)) - return -EINVAL; - osrate = osr_adc_sel[osr].osr; + if (osr >= ARRAY_SIZE(osr_adc_sel)) + return -EINVAL; + osrate = osr_adc_sel[osr].osr; - if (rate * osr > CLK_ADC_MAX) { - dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); - return -EINVAL; - } + if (rate * osr > CLK_ADC_MAX) { + dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); + return -EINVAL; + } - return 0; + return 0; } static int nau8540_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, osr; - - /* CLK_ADC = OSR * FS - * ADC clock frequency is defined as Over Sampling Rate (OSR) - * multiplied by the audio sample rate (Fs). Note that the OSR and Fs - * values must be selected such that the maximum frequency is less - * than 6.144 MHz. - */ - regmap_read(nau8540->regmap, NAU8540_REG_ADC_SAMPLE_RATE, &osr); - osr &= NAU8540_ADC_OSR_MASK; - if (nau8540_clock_check(nau8540, params_rate(params), osr)) - return -EINVAL; - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_ADC_SRC_MASK, - osr_adc_sel[osr].clk_src << NAU8540_CLK_ADC_SRC_SFT); - - switch (params_width(params)) { - case 16: - val_len |= NAU8540_I2S_DL_16; - break; - case 20: - val_len |= NAU8540_I2S_DL_20; - break; - case 24: - val_len |= NAU8540_I2S_DL_24; - break; - case 32: - val_len |= NAU8540_I2S_DL_32; - break; - default: - return -EINVAL; - } - - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, - NAU8540_I2S_DL_MASK, val_len); - - return 0; + struct snd_soc_codec *codec = dai->codec; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, osr; + + /* CLK_ADC = OSR * FS + * ADC clock frequency is defined as Over Sampling Rate (OSR) + * multiplied by the audio sample rate (Fs). Note that the OSR and Fs + * values must be selected such that the maximum frequency is less + * than 6.144 MHz. + */ + regmap_read(nau8540->regmap, NAU8540_REG_ADC_SAMPLE_RATE, &osr); + osr &= NAU8540_ADC_OSR_MASK; + if (nau8540_clock_check(nau8540, params_rate(params), osr)) + return -EINVAL; + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_ADC_SRC_MASK, + osr_adc_sel[osr].clk_src << NAU8540_CLK_ADC_SRC_SFT); + + switch (params_width(params)) { + case 16: + val_len |= NAU8540_I2S_DL_16; + break; + case 20: + val_len |= NAU8540_I2S_DL_20; + break; + case 24: + val_len |= NAU8540_I2S_DL_24; + break; + case 32: + val_len |= NAU8540_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, + NAU8540_I2S_DL_MASK, val_len); + + return 0; } static int nau8540_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - struct snd_soc_codec *codec = dai->codec; - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - unsigned int ctrl1_val = 0, ctrl2_val = 0; - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - ctrl2_val |= NAU8540_I2S_MS_MASTER; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_IB_NF: - ctrl1_val |= NAU8540_I2S_BP_INV; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - ctrl1_val |= NAU8540_I2S_DF_I2S; - break; - case SND_SOC_DAIFMT_LEFT_J: - ctrl1_val |= NAU8540_I2S_DF_LEFT; - break; - case SND_SOC_DAIFMT_RIGHT_J: - ctrl1_val |= NAU8540_I2S_DF_RIGTH; - break; - case SND_SOC_DAIFMT_DSP_A: - ctrl1_val |= NAU8540_I2S_DF_PCM_AB; - break; - case SND_SOC_DAIFMT_DSP_B: - ctrl1_val |= NAU8540_I2S_DF_PCM_AB; - ctrl1_val |= NAU8540_I2S_PCMB_EN; - break; - default: - return -EINVAL; - } - - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, - NAU8540_I2S_DL_MASK | NAU8540_I2S_DF_MASK | - NAU8540_I2S_BP_INV | NAU8540_I2S_PCMB_EN, ctrl1_val); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, - NAU8540_I2S_MS_MASK | NAU8540_I2S_DO12_OE, ctrl2_val); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, - NAU8540_I2S_DO34_OE, 0); - - return 0; + struct snd_soc_codec *codec = dai->codec; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8540_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8540_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8540_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8540_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8540_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8540_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8540_I2S_DF_PCM_AB; + ctrl1_val |= NAU8540_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, + NAU8540_I2S_DL_MASK | NAU8540_I2S_DF_MASK | + NAU8540_I2S_BP_INV | NAU8540_I2S_PCMB_EN, ctrl1_val); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, + NAU8540_I2S_MS_MASK | NAU8540_I2S_DO12_OE, ctrl2_val); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, + NAU8540_I2S_DO34_OE, 0); + + return 0; } /** @@ -451,55 +451,55 @@ static int nau8540_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) * Configures a DAI for TDM operation. Only support 4 slots TDM. */ static int nau8540_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct snd_soc_codec *codec = dai->codec; - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - unsigned int ctrl2_val = 0, ctrl4_val = 0; - - if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf))) - return -EINVAL; - - ctrl4_val |= (NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN); - if (tx_mask & 0xf0) { - ctrl2_val = 4 * slot_width; - ctrl4_val |= (tx_mask >> 4); - } else { - ctrl4_val |= tx_mask; - } - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL4, - NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN | - NAU8540_TDM_TX_MASK, ctrl4_val); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, - NAU8540_I2S_DO12_OE, NAU8540_I2S_DO12_OE); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, - NAU8540_I2S_DO34_OE | NAU8540_I2S_TSLOT_L_MASK, - NAU8540_I2S_DO34_OE | ctrl2_val); - - return 0; + struct snd_soc_codec *codec = dai->codec; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl2_val = 0, ctrl4_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf))) + return -EINVAL; + + ctrl4_val |= (NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + ctrl2_val = 4 * slot_width; + ctrl4_val |= (tx_mask >> 4); + } else { + ctrl4_val |= tx_mask; + } + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL4, + NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN | + NAU8540_TDM_TX_MASK, ctrl4_val); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, + NAU8540_I2S_DO12_OE, NAU8540_I2S_DO12_OE); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, + NAU8540_I2S_DO34_OE | NAU8540_I2S_TSLOT_L_MASK, + NAU8540_I2S_DO34_OE | ctrl2_val); + + return 0; } static const struct snd_soc_dai_ops nau8540_dai_ops = { - .hw_params = nau8540_hw_params, - .set_fmt = nau8540_set_fmt, - .set_tdm_slot = nau8540_set_tdm_slot, + .hw_params = nau8540_hw_params, + .set_fmt = nau8540_set_fmt, + .set_tdm_slot = nau8540_set_tdm_slot, }; #define NAU8540_RATES SNDRV_PCM_RATE_8000_48000 #define NAU8540_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ - | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver nau8540_dai = { - .name = "nau8540-hifi", - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 4, - .rates = NAU8540_RATES, - .formats = NAU8540_FORMATS, - }, - .ops = &nau8540_dai_ops, + .name = "nau8540-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = NAU8540_RATES, + .formats = NAU8540_FORMATS, + }, + .ops = &nau8540_dai_ops, }; /** @@ -513,320 +513,320 @@ static struct snd_soc_dai_driver nau8540_dai = { * Returns 0 for success or negative error code. */ static int nau8540_calc_fll_param(unsigned int fll_in, - unsigned int fs, struct nau8540_fll *fll_param) + unsigned int fs, struct nau8540_fll *fll_param) { - u64 fvco, fvco_max; - unsigned int fref, i, fvco_sel; - - /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing - * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. - * FREF = freq_in / NAU8540_FLL_REF_DIV_MASK - */ - for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { - fref = fll_in / fll_pre_scalar[i].param; - if (fref <= NAU_FREF_MAX) - break; - } - if (i == ARRAY_SIZE(fll_pre_scalar)) - return -EINVAL; - fll_param->clk_ref_div = fll_pre_scalar[i].val; - - /* Choose the FLL ratio based on FREF */ - for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { - if (fref >= fll_ratio[i].param) - break; - } - if (i == ARRAY_SIZE(fll_ratio)) - return -EINVAL; - fll_param->ratio = fll_ratio[i].val; - - /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 124MHz or the FFL cannot be - * guaranteed across the full range of operation. - * FDCO = freq_out * 2 * mclk_src_scaling - */ - fvco_max = 0; - fvco_sel = ARRAY_SIZE(mclk_src_scaling); - for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && - fvco_max < fvco) { - fvco_max = fvco; - fvco_sel = i; - } - } - if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) - return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; - - /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional - * input based on FDCO, FREF and FLL ratio. - */ - fvco = div_u64(fvco_max << 16, fref * fll_param->ratio); - fll_param->fll_int = (fvco >> 16) & 0x3FF; - fll_param->fll_frac = fvco & 0xFFFF; - return 0; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing + * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8540_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in / fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } + } + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; + + /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco_max << 16, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 16) & 0x3FF; + fll_param->fll_frac = fvco & 0xFFFF; + return 0; } static void nau8540_fll_apply(struct regmap *regmap, - struct nau8540_fll *fll_param) + struct nau8540_fll *fll_param) { - regmap_update_bits(regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK | NAU8540_CLK_MCLK_SRC_MASK, - NAU8540_CLK_SRC_MCLK | fll_param->mclk_src); - regmap_update_bits(regmap, NAU8540_REG_FLL1, - NAU8540_FLL_RATIO_MASK, fll_param->ratio); - /* FLL 16-bit fractional input */ - regmap_write(regmap, NAU8540_REG_FLL2, fll_param->fll_frac); - /* FLL 10-bit integer input */ - regmap_update_bits(regmap, NAU8540_REG_FLL3, - NAU8540_FLL_INTEGER_MASK, fll_param->fll_int); - /* FLL pre-scaler */ - regmap_update_bits(regmap, NAU8540_REG_FLL4, - NAU8540_FLL_REF_DIV_MASK, - fll_param->clk_ref_div << NAU8540_FLL_REF_DIV_SFT); - regmap_update_bits(regmap, NAU8540_REG_FLL5, - NAU8540_FLL_CLK_SW_MASK, NAU8540_FLL_CLK_SW_REF); - regmap_update_bits(regmap, - NAU8540_REG_FLL6, NAU8540_DCO_EN, 0); - if (fll_param->fll_frac) { - regmap_update_bits(regmap, NAU8540_REG_FLL5, - NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | - NAU8540_FLL_FTR_SW_MASK, - NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | - NAU8540_FLL_FTR_SW_FILTER); - regmap_update_bits(regmap, NAU8540_REG_FLL6, - NAU8540_SDM_EN, NAU8540_SDM_EN); - } else { - regmap_update_bits(regmap, NAU8540_REG_FLL5, - NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | - NAU8540_FLL_FTR_SW_MASK, NAU8540_FLL_FTR_SW_ACCU); - regmap_update_bits(regmap, - NAU8540_REG_FLL6, NAU8540_SDM_EN, 0); - } + regmap_update_bits(regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK | NAU8540_CLK_MCLK_SRC_MASK, + NAU8540_CLK_SRC_MCLK | fll_param->mclk_src); + regmap_update_bits(regmap, NAU8540_REG_FLL1, + NAU8540_FLL_RATIO_MASK, fll_param->ratio); + /* FLL 16-bit fractional input */ + regmap_write(regmap, NAU8540_REG_FLL2, fll_param->fll_frac); + /* FLL 10-bit integer input */ + regmap_update_bits(regmap, NAU8540_REG_FLL3, + NAU8540_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(regmap, NAU8540_REG_FLL4, + NAU8540_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8540_FLL_REF_DIV_SFT); + regmap_update_bits(regmap, NAU8540_REG_FLL5, + NAU8540_FLL_CLK_SW_MASK, NAU8540_FLL_CLK_SW_REF); + regmap_update_bits(regmap, + NAU8540_REG_FLL6, NAU8540_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(regmap, NAU8540_REG_FLL5, + NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | + NAU8540_FLL_FTR_SW_MASK, + NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | + NAU8540_FLL_FTR_SW_FILTER); + regmap_update_bits(regmap, NAU8540_REG_FLL6, + NAU8540_SDM_EN, NAU8540_SDM_EN); + } else { + regmap_update_bits(regmap, NAU8540_REG_FLL5, + NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | + NAU8540_FLL_FTR_SW_MASK, NAU8540_FLL_FTR_SW_ACCU); + regmap_update_bits(regmap, + NAU8540_REG_FLL6, NAU8540_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ static int nau8540_set_pll(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out) + unsigned int freq_in, unsigned int freq_out) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - struct nau8540_fll fll_param; - int ret, fs; - - switch (pll_id) { - case NAU8540_CLK_FLL_MCLK: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_MCLK); - break; - - case NAU8540_CLK_FLL_BLK: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_BLK); - break; - - case NAU8540_CLK_FLL_FS: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_FS); - break; - - default: - dev_err(nau8540->dev, "Invalid clock id (%d)\n", pll_id); - return -EINVAL; - } - dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", - freq_out, pll_id); - - fs = freq_out / 256; - ret = nau8540_calc_fll_param(freq_in, fs, &fll_param); - if (ret < 0) { - dev_err(nau8540->dev, "Unsupported input clock %d\n", freq_in); - return ret; - } - dev_dbg(nau8540->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", - fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, - fll_param.fll_int, fll_param.clk_ref_div); - - nau8540_fll_apply(nau8540->regmap, &fll_param); - mdelay(2); - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); - - return 0; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + struct nau8540_fll fll_param; + int ret, fs; + + switch (pll_id) { + case NAU8540_CLK_FLL_MCLK: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, + NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_MCLK); + break; + + case NAU8540_CLK_FLL_BLK: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, + NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_BLK); + break; + + case NAU8540_CLK_FLL_FS: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, + NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_FS); + break; + + default: + dev_err(nau8540->dev, "Invalid clock id (%d)\n", pll_id); + return -EINVAL; + } + dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", + freq_out, pll_id); + + fs = freq_out / 256; + ret = nau8540_calc_fll_param(freq_in, fs, &fll_param); + if (ret < 0) { + dev_err(nau8540->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_dbg(nau8540->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, + fll_param.fll_int, fll_param.clk_ref_div); + + nau8540_fll_apply(nau8540->regmap, &fll_param); + mdelay(2); + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); + + return 0; } static int nau8540_set_sysclk(struct snd_soc_codec *codec, - int clk_id, int source, unsigned int freq, int dir) + int clk_id, int source, unsigned int freq, int dir) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - - switch (clk_id) { - case NAU8540_CLK_DIS: - case NAU8540_CLK_MCLK: - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_MCLK); - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, - NAU8540_DCO_EN, 0); - break; - - case NAU8540_CLK_INTERNAL: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, - NAU8540_DCO_EN, NAU8540_DCO_EN); - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); - break; - - default: - dev_err(nau8540->dev, "Invalid clock id (%d)\n", clk_id); - return -EINVAL; - } - - dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", - freq, clk_id); - - return 0; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case NAU8540_CLK_DIS: + case NAU8540_CLK_MCLK: + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_MCLK); + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, + NAU8540_DCO_EN, 0); + break; + + case NAU8540_CLK_INTERNAL: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, + NAU8540_DCO_EN, NAU8540_DCO_EN); + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); + break; + + default: + dev_err(nau8540->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; } static void nau8540_reset_chip(struct regmap *regmap) { - regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); - regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); + regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); + regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); } static void nau8540_init_regs(struct nau8540 *nau8540) { - struct regmap *regmap = nau8540->regmap; - - /* Enable Bias/VMID/VMID Tieoff */ - regmap_update_bits(regmap, NAU8540_REG_VMID_CTRL, - NAU8540_VMID_EN | NAU8540_VMID_SEL_MASK, - NAU8540_VMID_EN | (0x2 << NAU8540_VMID_SEL_SFT)); - regmap_update_bits(regmap, NAU8540_REG_REFERENCE, - NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN, - NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN); - mdelay(2); - regmap_update_bits(regmap, NAU8540_REG_MIC_BIAS, - NAU8540_PU_PRE, NAU8540_PU_PRE); - regmap_update_bits(regmap, NAU8540_REG_CLOCK_CTRL, - NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN, - NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN); - /* ADC OSR selection, CLK_ADC = Fs * OSR */ - regmap_update_bits(regmap, NAU8540_REG_ADC_SAMPLE_RATE, - NAU8540_ADC_OSR_MASK, NAU8540_ADC_OSR_64); + struct regmap *regmap = nau8540->regmap; + + /* Enable Bias/VMID/VMID Tieoff */ + regmap_update_bits(regmap, NAU8540_REG_VMID_CTRL, + NAU8540_VMID_EN | NAU8540_VMID_SEL_MASK, + NAU8540_VMID_EN | (0x2 << NAU8540_VMID_SEL_SFT)); + regmap_update_bits(regmap, NAU8540_REG_REFERENCE, + NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN, + NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN); + mdelay(2); + regmap_update_bits(regmap, NAU8540_REG_MIC_BIAS, + NAU8540_PU_PRE, NAU8540_PU_PRE); + regmap_update_bits(regmap, NAU8540_REG_CLOCK_CTRL, + NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN, + NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN); + /* ADC OSR selection, CLK_ADC = Fs * OSR */ + regmap_update_bits(regmap, NAU8540_REG_ADC_SAMPLE_RATE, + NAU8540_ADC_OSR_MASK, NAU8540_ADC_OSR_64); } static int __maybe_unused nau8540_suspend(struct snd_soc_codec *codec) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(nau8540->regmap, true); - regcache_mark_dirty(nau8540->regmap); + regcache_cache_only(nau8540->regmap, true); + regcache_mark_dirty(nau8540->regmap); - return 0; + return 0; } static int __maybe_unused nau8540_resume(struct snd_soc_codec *codec) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(nau8540->regmap, false); - regcache_sync(nau8540->regmap); + regcache_cache_only(nau8540->regmap, false); + regcache_sync(nau8540->regmap); - return 0; + return 0; } static struct snd_soc_codec_driver nau8540_codec_driver = { - .set_sysclk = nau8540_set_sysclk, - .set_pll = nau8540_set_pll, - .suspend = nau8540_suspend, - .resume = nau8540_resume, - .suspend_bias_off = true, - - .component_driver = { - .controls = nau8540_snd_controls, - .num_controls = ARRAY_SIZE(nau8540_snd_controls), - .dapm_widgets = nau8540_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(nau8540_dapm_widgets), - .dapm_routes = nau8540_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(nau8540_dapm_routes), - }, + .set_sysclk = nau8540_set_sysclk, + .set_pll = nau8540_set_pll, + .suspend = nau8540_suspend, + .resume = nau8540_resume, + .suspend_bias_off = true, + + .component_driver = { + .controls = nau8540_snd_controls, + .num_controls = ARRAY_SIZE(nau8540_snd_controls), + .dapm_widgets = nau8540_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8540_dapm_widgets), + .dapm_routes = nau8540_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8540_dapm_routes), + }, }; static const struct regmap_config nau8540_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = 16, + .reg_bits = 16, - .max_register = NAU8540_REG_MAX, - .readable_reg = nau8540_readable_reg, - .writeable_reg = nau8540_writeable_reg, - .volatile_reg = nau8540_volatile_reg, + .max_register = NAU8540_REG_MAX, + .readable_reg = nau8540_readable_reg, + .writeable_reg = nau8540_writeable_reg, + .volatile_reg = nau8540_volatile_reg, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = nau8540_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(nau8540_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8540_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8540_reg_defaults), }; static int nau8540_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { - struct device *dev = &i2c->dev; - struct nau8540 *nau8540 = dev_get_platdata(dev); - int ret, value; - - if (!nau8540) { - nau8540 = devm_kzalloc(dev, sizeof(*nau8540), GFP_KERNEL); - if (!nau8540) - return -ENOMEM; - } - i2c_set_clientdata(i2c, nau8540); - - nau8540->regmap = devm_regmap_init_i2c(i2c, &nau8540_regmap_config); - if (IS_ERR(nau8540->regmap)) - return PTR_ERR(nau8540->regmap); - ret = regmap_read(nau8540->regmap, NAU8540_REG_I2C_DEVICE_ID, &value); - if (ret < 0) { - dev_err(dev, "Failed to read device id from the NAU85L40: %d\n", - ret); - return ret; - } - - nau8540->dev = dev; - nau8540_reset_chip(nau8540->regmap); - nau8540_init_regs(nau8540); - - return snd_soc_register_codec(dev, - &nau8540_codec_driver, &nau8540_dai, 1); + struct device *dev = &i2c->dev; + struct nau8540 *nau8540 = dev_get_platdata(dev); + int ret, value; + + if (!nau8540) { + nau8540 = devm_kzalloc(dev, sizeof(*nau8540), GFP_KERNEL); + if (!nau8540) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8540); + + nau8540->regmap = devm_regmap_init_i2c(i2c, &nau8540_regmap_config); + if (IS_ERR(nau8540->regmap)) + return PTR_ERR(nau8540->regmap); + ret = regmap_read(nau8540->regmap, NAU8540_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU85L40: %d\n", + ret); + return ret; + } + + nau8540->dev = dev; + nau8540_reset_chip(nau8540->regmap); + nau8540_init_regs(nau8540); + + return snd_soc_register_codec(dev, + &nau8540_codec_driver, &nau8540_dai, 1); } static int nau8540_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); - return 0; + snd_soc_unregister_codec(&client->dev); + return 0; } static const struct i2c_device_id nau8540_i2c_ids[] = { - { "nau8540", 0 }, - { } + { "nau8540", 0 }, + { } }; MODULE_DEVICE_TABLE(i2c, nau8540_i2c_ids); #ifdef CONFIG_OF static const struct of_device_id nau8540_of_ids[] = { - { .compatible = "nuvoton,nau8540", }, - {} + { .compatible = "nuvoton,nau8540", }, + {} }; MODULE_DEVICE_TABLE(of, nau8540_of_ids); #endif static struct i2c_driver nau8540_i2c_driver = { - .driver = { - .name = "nau8540", - .of_match_table = of_match_ptr(nau8540_of_ids), - }, - .probe = nau8540_i2c_probe, - .remove = nau8540_i2c_remove, - .id_table = nau8540_i2c_ids, + .driver = { + .name = "nau8540", + .of_match_table = of_match_ptr(nau8540_of_ids), + }, + .probe = nau8540_i2c_probe, + .remove = nau8540_i2c_remove, + .id_table = nau8540_i2c_ids, }; module_i2c_driver(nau8540_i2c_driver); diff --git a/sound/soc/codecs/nau8540.h b/sound/soc/codecs/nau8540.h index d06e65188cd5..5db5b224944d 100644 --- a/sound/soc/codecs/nau8540.h +++ b/sound/soc/codecs/nau8540.h @@ -12,211 +12,211 @@ #ifndef __NAU8540_H__ #define __NAU8540_H__ -#define NAU8540_REG_SW_RESET 0x00 -#define NAU8540_REG_POWER_MANAGEMENT 0x01 -#define NAU8540_REG_CLOCK_CTRL 0x02 -#define NAU8540_REG_CLOCK_SRC 0x03 -#define NAU8540_REG_FLL1 0x04 -#define NAU8540_REG_FLL2 0x05 -#define NAU8540_REG_FLL3 0x06 -#define NAU8540_REG_FLL4 0x07 -#define NAU8540_REG_FLL5 0x08 -#define NAU8540_REG_FLL6 0x09 -#define NAU8540_REG_FLL_VCO_RSV 0x0A -#define NAU8540_REG_PCM_CTRL0 0x10 -#define NAU8540_REG_PCM_CTRL1 0x11 -#define NAU8540_REG_PCM_CTRL2 0x12 -#define NAU8540_REG_PCM_CTRL3 0x13 -#define NAU8540_REG_PCM_CTRL4 0x14 -#define NAU8540_REG_ALC_CONTROL_1 0x20 -#define NAU8540_REG_ALC_CONTROL_2 0x21 -#define NAU8540_REG_ALC_CONTROL_3 0x22 -#define NAU8540_REG_ALC_CONTROL_4 0x23 -#define NAU8540_REG_ALC_CONTROL_5 0x24 -#define NAU8540_REG_ALC_GAIN_CH12 0x2D -#define NAU8540_REG_ALC_GAIN_CH34 0x2E -#define NAU8540_REG_ALC_STATUS 0x2F -#define NAU8540_REG_NOTCH_FIL1_CH1 0x30 -#define NAU8540_REG_NOTCH_FIL2_CH1 0x31 -#define NAU8540_REG_NOTCH_FIL1_CH2 0x32 -#define NAU8540_REG_NOTCH_FIL2_CH2 0x33 -#define NAU8540_REG_NOTCH_FIL1_CH3 0x34 -#define NAU8540_REG_NOTCH_FIL2_CH3 0x35 -#define NAU8540_REG_NOTCH_FIL1_CH4 0x36 -#define NAU8540_REG_NOTCH_FIL2_CH4 0x37 -#define NAU8540_REG_HPF_FILTER_CH12 0x38 -#define NAU8540_REG_HPF_FILTER_CH34 0x39 -#define NAU8540_REG_ADC_SAMPLE_RATE 0x3A -#define NAU8540_REG_DIGITAL_GAIN_CH1 0x40 -#define NAU8540_REG_DIGITAL_GAIN_CH2 0x41 -#define NAU8540_REG_DIGITAL_GAIN_CH3 0x42 -#define NAU8540_REG_DIGITAL_GAIN_CH4 0x43 -#define NAU8540_REG_DIGITAL_MUX 0x44 -#define NAU8540_REG_P2P_CH1 0x48 -#define NAU8540_REG_P2P_CH2 0x49 -#define NAU8540_REG_P2P_CH3 0x4A -#define NAU8540_REG_P2P_CH4 0x4B -#define NAU8540_REG_PEAK_CH1 0x4C -#define NAU8540_REG_PEAK_CH2 0x4D -#define NAU8540_REG_PEAK_CH3 0x4E -#define NAU8540_REG_PEAK_CH4 0x4F -#define NAU8540_REG_GPIO_CTRL 0x50 -#define NAU8540_REG_MISC_CTRL 0x51 -#define NAU8540_REG_I2C_CTRL 0x52 -#define NAU8540_REG_I2C_DEVICE_ID 0x58 -#define NAU8540_REG_RST 0x5A -#define NAU8540_REG_VMID_CTRL 0x60 -#define NAU8540_REG_MUTE 0x61 -#define NAU8540_REG_ANALOG_ADC1 0x64 -#define NAU8540_REG_ANALOG_ADC2 0x65 -#define NAU8540_REG_ANALOG_PWR 0x66 -#define NAU8540_REG_MIC_BIAS 0x67 -#define NAU8540_REG_REFERENCE 0x68 -#define NAU8540_REG_FEPGA1 0x69 -#define NAU8540_REG_FEPGA2 0x6A -#define NAU8540_REG_FEPGA3 0x6B -#define NAU8540_REG_FEPGA4 0x6C -#define NAU8540_REG_PWR 0x6D -#define NAU8540_REG_MAX NAU8540_REG_PWR +#define NAU8540_REG_SW_RESET 0x00 +#define NAU8540_REG_POWER_MANAGEMENT 0x01 +#define NAU8540_REG_CLOCK_CTRL 0x02 +#define NAU8540_REG_CLOCK_SRC 0x03 +#define NAU8540_REG_FLL1 0x04 +#define NAU8540_REG_FLL2 0x05 +#define NAU8540_REG_FLL3 0x06 +#define NAU8540_REG_FLL4 0x07 +#define NAU8540_REG_FLL5 0x08 +#define NAU8540_REG_FLL6 0x09 +#define NAU8540_REG_FLL_VCO_RSV 0x0A +#define NAU8540_REG_PCM_CTRL0 0x10 +#define NAU8540_REG_PCM_CTRL1 0x11 +#define NAU8540_REG_PCM_CTRL2 0x12 +#define NAU8540_REG_PCM_CTRL3 0x13 +#define NAU8540_REG_PCM_CTRL4 0x14 +#define NAU8540_REG_ALC_CONTROL_1 0x20 +#define NAU8540_REG_ALC_CONTROL_2 0x21 +#define NAU8540_REG_ALC_CONTROL_3 0x22 +#define NAU8540_REG_ALC_CONTROL_4 0x23 +#define NAU8540_REG_ALC_CONTROL_5 0x24 +#define NAU8540_REG_ALC_GAIN_CH12 0x2D +#define NAU8540_REG_ALC_GAIN_CH34 0x2E +#define NAU8540_REG_ALC_STATUS 0x2F +#define NAU8540_REG_NOTCH_FIL1_CH1 0x30 +#define NAU8540_REG_NOTCH_FIL2_CH1 0x31 +#define NAU8540_REG_NOTCH_FIL1_CH2 0x32 +#define NAU8540_REG_NOTCH_FIL2_CH2 0x33 +#define NAU8540_REG_NOTCH_FIL1_CH3 0x34 +#define NAU8540_REG_NOTCH_FIL2_CH3 0x35 +#define NAU8540_REG_NOTCH_FIL1_CH4 0x36 +#define NAU8540_REG_NOTCH_FIL2_CH4 0x37 +#define NAU8540_REG_HPF_FILTER_CH12 0x38 +#define NAU8540_REG_HPF_FILTER_CH34 0x39 +#define NAU8540_REG_ADC_SAMPLE_RATE 0x3A +#define NAU8540_REG_DIGITAL_GAIN_CH1 0x40 +#define NAU8540_REG_DIGITAL_GAIN_CH2 0x41 +#define NAU8540_REG_DIGITAL_GAIN_CH3 0x42 +#define NAU8540_REG_DIGITAL_GAIN_CH4 0x43 +#define NAU8540_REG_DIGITAL_MUX 0x44 +#define NAU8540_REG_P2P_CH1 0x48 +#define NAU8540_REG_P2P_CH2 0x49 +#define NAU8540_REG_P2P_CH3 0x4A +#define NAU8540_REG_P2P_CH4 0x4B +#define NAU8540_REG_PEAK_CH1 0x4C +#define NAU8540_REG_PEAK_CH2 0x4D +#define NAU8540_REG_PEAK_CH3 0x4E +#define NAU8540_REG_PEAK_CH4 0x4F +#define NAU8540_REG_GPIO_CTRL 0x50 +#define NAU8540_REG_MISC_CTRL 0x51 +#define NAU8540_REG_I2C_CTRL 0x52 +#define NAU8540_REG_I2C_DEVICE_ID 0x58 +#define NAU8540_REG_RST 0x5A +#define NAU8540_REG_VMID_CTRL 0x60 +#define NAU8540_REG_MUTE 0x61 +#define NAU8540_REG_ANALOG_ADC1 0x64 +#define NAU8540_REG_ANALOG_ADC2 0x65 +#define NAU8540_REG_ANALOG_PWR 0x66 +#define NAU8540_REG_MIC_BIAS 0x67 +#define NAU8540_REG_REFERENCE 0x68 +#define NAU8540_REG_FEPGA1 0x69 +#define NAU8540_REG_FEPGA2 0x6A +#define NAU8540_REG_FEPGA3 0x6B +#define NAU8540_REG_FEPGA4 0x6C +#define NAU8540_REG_PWR 0x6D +#define NAU8540_REG_MAX NAU8540_REG_PWR /* POWER_MANAGEMENT (0x01) */ -#define NAU8540_ADC4_EN (0x1 << 3) -#define NAU8540_ADC3_EN (0x1 << 2) -#define NAU8540_ADC2_EN (0x1 << 1) -#define NAU8540_ADC1_EN 0x1 +#define NAU8540_ADC4_EN (0x1 << 3) +#define NAU8540_ADC3_EN (0x1 << 2) +#define NAU8540_ADC2_EN (0x1 << 1) +#define NAU8540_ADC1_EN 0x1 /* CLOCK_CTRL (0x02) */ -#define NAU8540_CLK_ADC_EN (0x1 << 15) -#define NAU8540_CLK_I2S_EN (0x1 << 1) +#define NAU8540_CLK_ADC_EN (0x1 << 15) +#define NAU8540_CLK_I2S_EN (0x1 << 1) /* CLOCK_SRC (0x03) */ -#define NAU8540_CLK_SRC_SFT 15 -#define NAU8540_CLK_SRC_MASK (1 << NAU8540_CLK_SRC_SFT) -#define NAU8540_CLK_SRC_VCO (1 << NAU8540_CLK_SRC_SFT) -#define NAU8540_CLK_SRC_MCLK (0 << NAU8540_CLK_SRC_SFT) -#define NAU8540_CLK_ADC_SRC_SFT 6 -#define NAU8540_CLK_ADC_SRC_MASK (0x3 << NAU8540_CLK_ADC_SRC_SFT) -#define NAU8540_CLK_MCLK_SRC_MASK 0xf +#define NAU8540_CLK_SRC_SFT 15 +#define NAU8540_CLK_SRC_MASK (1 << NAU8540_CLK_SRC_SFT) +#define NAU8540_CLK_SRC_VCO (1 << NAU8540_CLK_SRC_SFT) +#define NAU8540_CLK_SRC_MCLK (0 << NAU8540_CLK_SRC_SFT) +#define NAU8540_CLK_ADC_SRC_SFT 6 +#define NAU8540_CLK_ADC_SRC_MASK (0x3 << NAU8540_CLK_ADC_SRC_SFT) +#define NAU8540_CLK_MCLK_SRC_MASK 0xf /* FLL1 (0x04) */ -#define NAU8540_FLL_RATIO_MASK 0x7f +#define NAU8540_FLL_RATIO_MASK 0x7f /* FLL3 (0x06) */ -#define NAU8540_FLL_CLK_SRC_SFT 10 -#define NAU8540_FLL_CLK_SRC_MASK (0x3 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_CLK_SRC_MCLK (0 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_CLK_SRC_BLK (0x2 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_CLK_SRC_FS (0x3 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_INTEGER_MASK 0x3ff +#define NAU8540_FLL_CLK_SRC_SFT 10 +#define NAU8540_FLL_CLK_SRC_MASK (0x3 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_CLK_SRC_MCLK (0 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_CLK_SRC_BLK (0x2 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_CLK_SRC_FS (0x3 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_INTEGER_MASK 0x3ff /* FLL4 (0x07) */ -#define NAU8540_FLL_REF_DIV_SFT 10 -#define NAU8540_FLL_REF_DIV_MASK (0x3 << NAU8540_FLL_REF_DIV_SFT) +#define NAU8540_FLL_REF_DIV_SFT 10 +#define NAU8540_FLL_REF_DIV_MASK (0x3 << NAU8540_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ -#define NAU8540_FLL_PDB_DAC_EN (0x1 << 15) -#define NAU8540_FLL_LOOP_FTR_EN (0x1 << 14) -#define NAU8540_FLL_CLK_SW_MASK (0x1 << 13) -#define NAU8540_FLL_CLK_SW_N2 (0x1 << 13) -#define NAU8540_FLL_CLK_SW_REF (0x0 << 13) -#define NAU8540_FLL_FTR_SW_MASK (0x1 << 12) -#define NAU8540_FLL_FTR_SW_ACCU (0x1 << 12) -#define NAU8540_FLL_FTR_SW_FILTER (0x0 << 12) +#define NAU8540_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8540_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8540_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8540_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8540_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8540_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8540_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8540_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8540_DCO_EN (0x1 << 15) -#define NAU8540_SDM_EN (0x1 << 14) +#define NAU8540_DCO_EN (0x1 << 15) +#define NAU8540_SDM_EN (0x1 << 14) /* PCM_CTRL0 (0x10) */ -#define NAU8540_I2S_BP_SFT 7 -#define NAU8540_I2S_BP_INV (0x1 << NAU8540_I2S_BP_SFT) -#define NAU8540_I2S_PCMB_SFT 6 -#define NAU8540_I2S_PCMB_EN (0x1 << NAU8540_I2S_PCMB_SFT) -#define NAU8540_I2S_DL_SFT 2 -#define NAU8540_I2S_DL_MASK (0x3 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_16 (0 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_20 (0x1 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_24 (0x2 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_32 (0x3 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DF_MASK 0x3 -#define NAU8540_I2S_DF_RIGTH 0 -#define NAU8540_I2S_DF_LEFT 0x1 -#define NAU8540_I2S_DF_I2S 0x2 -#define NAU8540_I2S_DF_PCM_AB 0x3 +#define NAU8540_I2S_BP_SFT 7 +#define NAU8540_I2S_BP_INV (0x1 << NAU8540_I2S_BP_SFT) +#define NAU8540_I2S_PCMB_SFT 6 +#define NAU8540_I2S_PCMB_EN (0x1 << NAU8540_I2S_PCMB_SFT) +#define NAU8540_I2S_DL_SFT 2 +#define NAU8540_I2S_DL_MASK (0x3 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_16 (0 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_20 (0x1 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_24 (0x2 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_32 (0x3 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DF_MASK 0x3 +#define NAU8540_I2S_DF_RIGTH 0 +#define NAU8540_I2S_DF_LEFT 0x1 +#define NAU8540_I2S_DF_I2S 0x2 +#define NAU8540_I2S_DF_PCM_AB 0x3 /* PCM_CTRL1 (0x11) */ -#define NAU8540_I2S_LRC_DIV_SFT 12 -#define NAU8540_I2S_LRC_DIV_MASK (0x3 << NAU8540_I2S_LRC_DIV_SFT) -#define NAU8540_I2S_DO12_OE (0x1 << 4) -#define NAU8540_I2S_MS_SFT 3 -#define NAU8540_I2S_MS_MASK (0x1 << NAU8540_I2S_MS_SFT) -#define NAU8540_I2S_MS_MASTER (0x1 << NAU8540_I2S_MS_SFT) -#define NAU8540_I2S_MS_SLAVE (0x0 << NAU8540_I2S_MS_SFT) -#define NAU8540_I2S_BLK_DIV_MASK 0x7 +#define NAU8540_I2S_LRC_DIV_SFT 12 +#define NAU8540_I2S_LRC_DIV_MASK (0x3 << NAU8540_I2S_LRC_DIV_SFT) +#define NAU8540_I2S_DO12_OE (0x1 << 4) +#define NAU8540_I2S_MS_SFT 3 +#define NAU8540_I2S_MS_MASK (0x1 << NAU8540_I2S_MS_SFT) +#define NAU8540_I2S_MS_MASTER (0x1 << NAU8540_I2S_MS_SFT) +#define NAU8540_I2S_MS_SLAVE (0x0 << NAU8540_I2S_MS_SFT) +#define NAU8540_I2S_BLK_DIV_MASK 0x7 /* PCM_CTRL1 (0x12) */ -#define NAU8540_I2S_DO34_OE (0x1 << 11) -#define NAU8540_I2S_TSLOT_L_MASK 0x3ff +#define NAU8540_I2S_DO34_OE (0x1 << 11) +#define NAU8540_I2S_TSLOT_L_MASK 0x3ff /* PCM_CTRL4 (0x14) */ -#define NAU8540_TDM_MODE (0x1 << 15) -#define NAU8540_TDM_OFFSET_EN (0x1 << 14) -#define NAU8540_TDM_TX_MASK 0xf +#define NAU8540_TDM_MODE (0x1 << 15) +#define NAU8540_TDM_OFFSET_EN (0x1 << 14) +#define NAU8540_TDM_TX_MASK 0xf /* ADC_SAMPLE_RATE (0x3A) */ -#define NAU8540_ADC_OSR_MASK 0x3 -#define NAU8540_ADC_OSR_256 0x3 -#define NAU8540_ADC_OSR_128 0x2 -#define NAU8540_ADC_OSR_64 0x1 -#define NAU8540_ADC_OSR_32 0x0 +#define NAU8540_ADC_OSR_MASK 0x3 +#define NAU8540_ADC_OSR_256 0x3 +#define NAU8540_ADC_OSR_128 0x2 +#define NAU8540_ADC_OSR_64 0x1 +#define NAU8540_ADC_OSR_32 0x0 /* VMID_CTRL (0x60) */ -#define NAU8540_VMID_EN (1 << 6) -#define NAU8540_VMID_SEL_SFT 4 -#define NAU8540_VMID_SEL_MASK (0x3 << NAU8540_VMID_SEL_SFT) +#define NAU8540_VMID_EN (1 << 6) +#define NAU8540_VMID_SEL_SFT 4 +#define NAU8540_VMID_SEL_MASK (0x3 << NAU8540_VMID_SEL_SFT) /* MIC_BIAS (0x67) */ -#define NAU8540_PU_PRE (0x1 << 8) +#define NAU8540_PU_PRE (0x1 << 8) /* REFERENCE (0x68) */ -#define NAU8540_PRECHARGE_DIS (0x1 << 13) -#define NAU8540_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8540_PRECHARGE_DIS (0x1 << 13) +#define NAU8540_GLOBAL_BIAS_EN (0x1 << 12) /* System Clock Source */ enum { - NAU8540_CLK_DIS, - NAU8540_CLK_MCLK, - NAU8540_CLK_INTERNAL, - NAU8540_CLK_FLL_MCLK, - NAU8540_CLK_FLL_BLK, - NAU8540_CLK_FLL_FS, + NAU8540_CLK_DIS, + NAU8540_CLK_MCLK, + NAU8540_CLK_INTERNAL, + NAU8540_CLK_FLL_MCLK, + NAU8540_CLK_FLL_BLK, + NAU8540_CLK_FLL_FS, }; struct nau8540 { - struct device *dev; - struct regmap *regmap; + struct device *dev; + struct regmap *regmap; }; struct nau8540_fll { - int mclk_src; - int ratio; - int fll_frac; - int fll_int; - int clk_ref_div; + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; }; struct nau8540_fll_attr { - unsigned int param; - unsigned int val; + unsigned int param; + unsigned int val; }; /* over sampling rate */ struct nau8540_osr_attr { - unsigned int osr; - unsigned int clk_src; + unsigned int osr; + unsigned int clk_src; }; -#endif /* __NAU8540_H__ */ +#endif /* __NAU8540_H__ */ -- 2.11.0

1 0

[alsa-devel] [PATCH 0/2] Lenovo Thinkpad quirks
by Pierre-Louis Bossart 18 Apr '17

18 Apr '17

Minor update to support new skews and need to manage quirks in codec driver for jack detection to work. Nicole Faerber (1): ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk Pierre-Louis Bossart (1): ASoC: Intel: Atom: update Thinkpad 10 quirk sound/soc/codecs/rt5670.c | 21 +++++++++++++++++++++ sound/soc/intel/atom/sst/sst_acpi.c | 16 +++++++++++++++- 2 files changed, 36 insertions(+), 1 deletion(-) -- 2.9.3

3 6

[alsa-devel] [PATCH] ASoC: topology: use j for internal loop counter
by Colin King 18 Apr '17

18 Apr '17

From: Colin Ian King <colin.king(a)canonical.com> Currently variable i is being for 2 nested for loops. Fix this by using integer loop counter j for the inside for loop. Fixes: 1a7dd6e2f1929 ("ASoC: topology: Allow a widget to have multiple enum controls") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 058bc99c6c34..002772e3ba2c 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -495,12 +495,13 @@ static void remove_widget(struct snd_soc_component *comp, struct snd_kcontrol *kcontrol = w->kcontrols[i]; struct soc_enum *se = (struct soc_enum *)kcontrol->private_value; + int j; snd_ctl_remove(card, kcontrol); kfree(se->dobj.control.dvalues); - for (i = 0; i < se->items; i++) - kfree(se->dobj.control.dtexts[i]); + for (j = 0; j < se->items; j++) + kfree(se->dobj.control.dtexts[j]); kfree(se); } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: cs35l35: Correct some register defaults
by Charles Keepax 18 Apr '17

18 Apr '17

Correct some minor errors in the register defaults. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 88c48e2..f8aef58 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -50,7 +50,7 @@ static const struct reg_default cs35l35_reg[] = { {CS35L35_PWRCTL2, 0x11}, {CS35L35_PWRCTL3, 0x00}, {CS35L35_CLK_CTL1, 0x04}, - {CS35L35_CLK_CTL2, 0x10}, + {CS35L35_CLK_CTL2, 0x12}, {CS35L35_CLK_CTL3, 0xCF}, {CS35L35_SP_FMT_CTL1, 0x20}, {CS35L35_SP_FMT_CTL2, 0x00}, @@ -70,7 +70,7 @@ static const struct reg_default cs35l35_reg[] = { {CS35L35_BST_RAMP_CTL, 0x85}, {CS35L35_BST_CONV_COEF_1, 0x24}, {CS35L35_BST_CONV_COEF_2, 0x24}, - {CS35L35_BST_CONV_SLOPE_COMP, 0x47}, + {CS35L35_BST_CONV_SLOPE_COMP, 0x4E}, {CS35L35_BST_CONV_SW_FREQ, 0x04}, {CS35L35_CLASS_H_CTL, 0x0B}, {CS35L35_CLASS_H_HEADRM_CTL, 0x0B}, @@ -78,9 +78,9 @@ static const struct reg_default cs35l35_reg[] = { {CS35L35_CLASS_H_FET_DRIVE_CTL, 0x41}, {CS35L35_CLASS_H_VP_CTL, 0xC5}, {CS35L35_VPBR_CTL, 0x0A}, - {CS35L35_VPBR_VOL_CTL, 0x09}, + {CS35L35_VPBR_VOL_CTL, 0x90}, {CS35L35_VPBR_TIMING_CTL, 0x6A}, - {CS35L35_VPBR_MODE_VOL_CTL, 0x40}, + {CS35L35_VPBR_MODE_VOL_CTL, 0x00}, {CS35L35_SPKR_MON_CTL, 0xC0}, {CS35L35_IMON_SCALE_CTL, 0x30}, {CS35L35_AUDIN_RXLOC_CTL, 0x00}, -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: nau8540: fix tab conversion problem
by John Hsu 18 Apr '17

18 Apr '17

Fix the tab converting to space problem. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8540.c | 1224 ++++++++++++++++++++++---------------------- sound/soc/codecs/nau8540.h | 310 +++++------ 2 files changed, 767 insertions(+), 767 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 9e8f0f4..c8bcb1d 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -39,147 +39,147 @@ /* scaling for mclk from sysclk_src output */ static const struct nau8540_fll_attr mclk_src_scaling[] = { - { 1, 0x0 }, - { 2, 0x2 }, - { 4, 0x3 }, - { 8, 0x4 }, - { 16, 0x5 }, - { 32, 0x6 }, - { 3, 0x7 }, - { 6, 0xa }, - { 12, 0xb }, - { 24, 0xc }, + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, }; /* ratio for input clk freq */ static const struct nau8540_fll_attr fll_ratio[] = { - { 512000, 0x01 }, - { 256000, 0x02 }, - { 128000, 0x04 }, - { 64000, 0x08 }, - { 32000, 0x10 }, - { 8000, 0x20 }, - { 4000, 0x40 }, + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, }; static const struct nau8540_fll_attr fll_pre_scalar[] = { - { 1, 0x0 }, - { 2, 0x1 }, - { 4, 0x2 }, - { 8, 0x3 }, + { 1, 0x0 }, + { 2, 0x1 }, + { 4, 0x2 }, + { 8, 0x3 }, }; /* over sampling rate */ static const struct nau8540_osr_attr osr_adc_sel[] = { - { 32, 3 }, /* OSR 32, SRC 1/8 */ - { 64, 2 }, /* OSR 64, SRC 1/4 */ - { 128, 1 }, /* OSR 128, SRC 1/2 */ - { 256, 0 }, /* OSR 256, SRC 1 */ + { 32, 3 }, /* OSR 32, SRC 1/8 */ + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 256, 0 }, /* OSR 256, SRC 1 */ }; static const struct reg_default nau8540_reg_defaults[] = { - {NAU8540_REG_POWER_MANAGEMENT, 0x0000}, - {NAU8540_REG_CLOCK_CTRL, 0x0000}, - {NAU8540_REG_CLOCK_SRC, 0x0000}, - {NAU8540_REG_FLL1, 0x0001}, - {NAU8540_REG_FLL2, 0x3126}, - {NAU8540_REG_FLL3, 0x0008}, - {NAU8540_REG_FLL4, 0x0010}, - {NAU8540_REG_FLL5, 0xC000}, - {NAU8540_REG_FLL6, 0x6000}, - {NAU8540_REG_FLL_VCO_RSV, 0xF13C}, - {NAU8540_REG_PCM_CTRL0, 0x000B}, - {NAU8540_REG_PCM_CTRL1, 0x3010}, - {NAU8540_REG_PCM_CTRL2, 0x0800}, - {NAU8540_REG_PCM_CTRL3, 0x0000}, - {NAU8540_REG_PCM_CTRL4, 0x000F}, - {NAU8540_REG_ALC_CONTROL_1, 0x0000}, - {NAU8540_REG_ALC_CONTROL_2, 0x700B}, - {NAU8540_REG_ALC_CONTROL_3, 0x0022}, - {NAU8540_REG_ALC_CONTROL_4, 0x1010}, - {NAU8540_REG_ALC_CONTROL_5, 0x1010}, - {NAU8540_REG_NOTCH_FIL1_CH1, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH1, 0x0000}, - {NAU8540_REG_NOTCH_FIL1_CH2, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH2, 0x0000}, - {NAU8540_REG_NOTCH_FIL1_CH3, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH3, 0x0000}, - {NAU8540_REG_NOTCH_FIL1_CH4, 0x0000}, - {NAU8540_REG_NOTCH_FIL2_CH4, 0x0000}, - {NAU8540_REG_HPF_FILTER_CH12, 0x0000}, - {NAU8540_REG_HPF_FILTER_CH34, 0x0000}, - {NAU8540_REG_ADC_SAMPLE_RATE, 0x0002}, - {NAU8540_REG_DIGITAL_GAIN_CH1, 0x0400}, - {NAU8540_REG_DIGITAL_GAIN_CH2, 0x0400}, - {NAU8540_REG_DIGITAL_GAIN_CH3, 0x0400}, - {NAU8540_REG_DIGITAL_GAIN_CH4, 0x0400}, - {NAU8540_REG_DIGITAL_MUX, 0x00E4}, - {NAU8540_REG_GPIO_CTRL, 0x0000}, - {NAU8540_REG_MISC_CTRL, 0x0000}, - {NAU8540_REG_I2C_CTRL, 0xEFFF}, - {NAU8540_REG_VMID_CTRL, 0x0000}, - {NAU8540_REG_MUTE, 0x0000}, - {NAU8540_REG_ANALOG_ADC1, 0x0011}, - {NAU8540_REG_ANALOG_ADC2, 0x0020}, - {NAU8540_REG_ANALOG_PWR, 0x0000}, - {NAU8540_REG_MIC_BIAS, 0x0004}, - {NAU8540_REG_REFERENCE, 0x0000}, - {NAU8540_REG_FEPGA1, 0x0000}, - {NAU8540_REG_FEPGA2, 0x0000}, - {NAU8540_REG_FEPGA3, 0x0101}, - {NAU8540_REG_FEPGA4, 0x0101}, - {NAU8540_REG_PWR, 0x0000}, + {NAU8540_REG_POWER_MANAGEMENT, 0x0000}, + {NAU8540_REG_CLOCK_CTRL, 0x0000}, + {NAU8540_REG_CLOCK_SRC, 0x0000}, + {NAU8540_REG_FLL1, 0x0001}, + {NAU8540_REG_FLL2, 0x3126}, + {NAU8540_REG_FLL3, 0x0008}, + {NAU8540_REG_FLL4, 0x0010}, + {NAU8540_REG_FLL5, 0xC000}, + {NAU8540_REG_FLL6, 0x6000}, + {NAU8540_REG_FLL_VCO_RSV, 0xF13C}, + {NAU8540_REG_PCM_CTRL0, 0x000B}, + {NAU8540_REG_PCM_CTRL1, 0x3010}, + {NAU8540_REG_PCM_CTRL2, 0x0800}, + {NAU8540_REG_PCM_CTRL3, 0x0000}, + {NAU8540_REG_PCM_CTRL4, 0x000F}, + {NAU8540_REG_ALC_CONTROL_1, 0x0000}, + {NAU8540_REG_ALC_CONTROL_2, 0x700B}, + {NAU8540_REG_ALC_CONTROL_3, 0x0022}, + {NAU8540_REG_ALC_CONTROL_4, 0x1010}, + {NAU8540_REG_ALC_CONTROL_5, 0x1010}, + {NAU8540_REG_NOTCH_FIL1_CH1, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH1, 0x0000}, + {NAU8540_REG_NOTCH_FIL1_CH2, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH2, 0x0000}, + {NAU8540_REG_NOTCH_FIL1_CH3, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH3, 0x0000}, + {NAU8540_REG_NOTCH_FIL1_CH4, 0x0000}, + {NAU8540_REG_NOTCH_FIL2_CH4, 0x0000}, + {NAU8540_REG_HPF_FILTER_CH12, 0x0000}, + {NAU8540_REG_HPF_FILTER_CH34, 0x0000}, + {NAU8540_REG_ADC_SAMPLE_RATE, 0x0002}, + {NAU8540_REG_DIGITAL_GAIN_CH1, 0x0400}, + {NAU8540_REG_DIGITAL_GAIN_CH2, 0x0400}, + {NAU8540_REG_DIGITAL_GAIN_CH3, 0x0400}, + {NAU8540_REG_DIGITAL_GAIN_CH4, 0x0400}, + {NAU8540_REG_DIGITAL_MUX, 0x00E4}, + {NAU8540_REG_GPIO_CTRL, 0x0000}, + {NAU8540_REG_MISC_CTRL, 0x0000}, + {NAU8540_REG_I2C_CTRL, 0xEFFF}, + {NAU8540_REG_VMID_CTRL, 0x0000}, + {NAU8540_REG_MUTE, 0x0000}, + {NAU8540_REG_ANALOG_ADC1, 0x0011}, + {NAU8540_REG_ANALOG_ADC2, 0x0020}, + {NAU8540_REG_ANALOG_PWR, 0x0000}, + {NAU8540_REG_MIC_BIAS, 0x0004}, + {NAU8540_REG_REFERENCE, 0x0000}, + {NAU8540_REG_FEPGA1, 0x0000}, + {NAU8540_REG_FEPGA2, 0x0000}, + {NAU8540_REG_FEPGA3, 0x0101}, + {NAU8540_REG_FEPGA4, 0x0101}, + {NAU8540_REG_PWR, 0x0000}, }; static bool nau8540_readable_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case NAU8540_REG_POWER_MANAGEMENT ... NAU8540_REG_FLL_VCO_RSV: - case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: - case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: - case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ADC_SAMPLE_RATE: - case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: - case NAU8540_REG_P2P_CH1 ... NAU8540_REG_I2C_CTRL: - case NAU8540_REG_I2C_DEVICE_ID: - case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: - case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: - return true; - default: - return false; - } + switch (reg) { + case NAU8540_REG_POWER_MANAGEMENT ... NAU8540_REG_FLL_VCO_RSV: + case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: + case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: + case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ADC_SAMPLE_RATE: + case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: + case NAU8540_REG_P2P_CH1 ... NAU8540_REG_I2C_CTRL: + case NAU8540_REG_I2C_DEVICE_ID: + case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: + case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: + return true; + default: + return false; + } } static bool nau8540_writeable_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case NAU8540_REG_SW_RESET ... NAU8540_REG_FLL_VCO_RSV: - case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: - case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: - case NAU8540_REG_NOTCH_FIL1_CH1 ... NAU8540_REG_ADC_SAMPLE_RATE: - case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: - case NAU8540_REG_GPIO_CTRL ... NAU8540_REG_I2C_CTRL: - case NAU8540_REG_RST: - case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: - case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: - return true; - default: - return false; - } + switch (reg) { + case NAU8540_REG_SW_RESET ... NAU8540_REG_FLL_VCO_RSV: + case NAU8540_REG_PCM_CTRL0 ... NAU8540_REG_PCM_CTRL4: + case NAU8540_REG_ALC_CONTROL_1 ... NAU8540_REG_ALC_CONTROL_5: + case NAU8540_REG_NOTCH_FIL1_CH1 ... NAU8540_REG_ADC_SAMPLE_RATE: + case NAU8540_REG_DIGITAL_GAIN_CH1 ... NAU8540_REG_DIGITAL_MUX: + case NAU8540_REG_GPIO_CTRL ... NAU8540_REG_I2C_CTRL: + case NAU8540_REG_RST: + case NAU8540_REG_VMID_CTRL ... NAU8540_REG_MUTE: + case NAU8540_REG_ANALOG_ADC1 ... NAU8540_REG_PWR: + return true; + default: + return false; + } } static bool nau8540_volatile_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case NAU8540_REG_SW_RESET: - case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ALC_STATUS: - case NAU8540_REG_P2P_CH1 ... NAU8540_REG_PEAK_CH4: - case NAU8540_REG_I2C_DEVICE_ID: - case NAU8540_REG_RST: - return true; - default: - return false; - } + switch (reg) { + case NAU8540_REG_SW_RESET: + case NAU8540_REG_ALC_GAIN_CH12 ... NAU8540_REG_ALC_STATUS: + case NAU8540_REG_P2P_CH1 ... NAU8540_REG_PEAK_CH4: + case NAU8540_REG_I2C_DEVICE_ID: + case NAU8540_REG_RST: + return true; + default: + return false; + } } @@ -187,255 +187,255 @@ static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -12800, 3600); static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); static const struct snd_kcontrol_new nau8540_snd_controls[] = { - SOC_SINGLE_TLV("Mic1 Volume", NAU8540_REG_DIGITAL_GAIN_CH1, - 0, 0x520, 0, adc_vol_tlv), - SOC_SINGLE_TLV("Mic2 Volume", NAU8540_REG_DIGITAL_GAIN_CH2, - 0, 0x520, 0, adc_vol_tlv), - SOC_SINGLE_TLV("Mic3 Volume", NAU8540_REG_DIGITAL_GAIN_CH3, - 0, 0x520, 0, adc_vol_tlv), - SOC_SINGLE_TLV("Mic4 Volume", NAU8540_REG_DIGITAL_GAIN_CH4, - 0, 0x520, 0, adc_vol_tlv), - - SOC_SINGLE_TLV("Frontend PGA1 Volume", NAU8540_REG_FEPGA3, - 0, 0x25, 0, fepga_gain_tlv), - SOC_SINGLE_TLV("Frontend PGA2 Volume", NAU8540_REG_FEPGA3, - 8, 0x25, 0, fepga_gain_tlv), - SOC_SINGLE_TLV("Frontend PGA3 Volume", NAU8540_REG_FEPGA4, - 0, 0x25, 0, fepga_gain_tlv), - SOC_SINGLE_TLV("Frontend PGA4 Volume", NAU8540_REG_FEPGA4, - 8, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Mic1 Volume", NAU8540_REG_DIGITAL_GAIN_CH1, + 0, 0x520, 0, adc_vol_tlv), + SOC_SINGLE_TLV("Mic2 Volume", NAU8540_REG_DIGITAL_GAIN_CH2, + 0, 0x520, 0, adc_vol_tlv), + SOC_SINGLE_TLV("Mic3 Volume", NAU8540_REG_DIGITAL_GAIN_CH3, + 0, 0x520, 0, adc_vol_tlv), + SOC_SINGLE_TLV("Mic4 Volume", NAU8540_REG_DIGITAL_GAIN_CH4, + 0, 0x520, 0, adc_vol_tlv), + + SOC_SINGLE_TLV("Frontend PGA1 Volume", NAU8540_REG_FEPGA3, + 0, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Frontend PGA2 Volume", NAU8540_REG_FEPGA3, + 8, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Frontend PGA3 Volume", NAU8540_REG_FEPGA4, + 0, 0x25, 0, fepga_gain_tlv), + SOC_SINGLE_TLV("Frontend PGA4 Volume", NAU8540_REG_FEPGA4, + 8, 0x25, 0, fepga_gain_tlv), }; static const char * const adc_channel[] = { - "ADC channel 1", "ADC channel 2", "ADC channel 3", "ADC channel 4" + "ADC channel 1", "ADC channel 2", "ADC channel 3", "ADC channel 4" }; static SOC_ENUM_SINGLE_DECL( - digital_ch4_enum, NAU8540_REG_DIGITAL_MUX, 6, adc_channel); + digital_ch4_enum, NAU8540_REG_DIGITAL_MUX, 6, adc_channel); static const struct snd_kcontrol_new digital_ch4_mux = - SOC_DAPM_ENUM("Digital CH4 Select", digital_ch4_enum); + SOC_DAPM_ENUM("Digital CH4 Select", digital_ch4_enum); static SOC_ENUM_SINGLE_DECL( - digital_ch3_enum, NAU8540_REG_DIGITAL_MUX, 4, adc_channel); + digital_ch3_enum, NAU8540_REG_DIGITAL_MUX, 4, adc_channel); static const struct snd_kcontrol_new digital_ch3_mux = - SOC_DAPM_ENUM("Digital CH3 Select", digital_ch3_enum); + SOC_DAPM_ENUM("Digital CH3 Select", digital_ch3_enum); static SOC_ENUM_SINGLE_DECL( - digital_ch2_enum, NAU8540_REG_DIGITAL_MUX, 2, adc_channel); + digital_ch2_enum, NAU8540_REG_DIGITAL_MUX, 2, adc_channel); static const struct snd_kcontrol_new digital_ch2_mux = - SOC_DAPM_ENUM("Digital CH2 Select", digital_ch2_enum); + SOC_DAPM_ENUM("Digital CH2 Select", digital_ch2_enum); static SOC_ENUM_SINGLE_DECL( - digital_ch1_enum, NAU8540_REG_DIGITAL_MUX, 0, adc_channel); + digital_ch1_enum, NAU8540_REG_DIGITAL_MUX, 0, adc_channel); static const struct snd_kcontrol_new digital_ch1_mux = - SOC_DAPM_ENUM("Digital CH1 Select", digital_ch1_enum); + SOC_DAPM_ENUM("Digital CH1 Select", digital_ch1_enum); static const struct snd_soc_dapm_widget nau8540_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("MICBIAS2", NAU8540_REG_MIC_BIAS, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MICBIAS1", NAU8540_REG_MIC_BIAS, 10, 0, NULL, 0), - - SND_SOC_DAPM_INPUT("MIC1"), - SND_SOC_DAPM_INPUT("MIC2"), - SND_SOC_DAPM_INPUT("MIC3"), - SND_SOC_DAPM_INPUT("MIC4"), - - SND_SOC_DAPM_PGA("Frontend PGA1", NAU8540_REG_PWR, 12, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA2", NAU8540_REG_PWR, 13, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA3", NAU8540_REG_PWR, 14, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA4", NAU8540_REG_PWR, 15, 0, NULL, 0), - - SND_SOC_DAPM_ADC("ADC1", NULL, - NAU8540_REG_POWER_MANAGEMENT, 0, 0), - SND_SOC_DAPM_ADC("ADC2", NULL, - NAU8540_REG_POWER_MANAGEMENT, 1, 0), - SND_SOC_DAPM_ADC("ADC3", NULL, - NAU8540_REG_POWER_MANAGEMENT, 2, 0), - SND_SOC_DAPM_ADC("ADC4", NULL, - NAU8540_REG_POWER_MANAGEMENT, 3, 0), - - SND_SOC_DAPM_PGA("ADC CH1", NAU8540_REG_ANALOG_PWR, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH2", NAU8540_REG_ANALOG_PWR, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH3", NAU8540_REG_ANALOG_PWR, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH4", NAU8540_REG_ANALOG_PWR, 3, 0, NULL, 0), - - SND_SOC_DAPM_MUX("Digital CH4 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch4_mux), - SND_SOC_DAPM_MUX("Digital CH3 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch3_mux), - SND_SOC_DAPM_MUX("Digital CH2 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch2_mux), - SND_SOC_DAPM_MUX("Digital CH1 Mux", - SND_SOC_NOPM, 0, 0, &digital_ch1_mux), - - SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", NAU8540_REG_MIC_BIAS, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", NAU8540_REG_MIC_BIAS, 10, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_PGA("Frontend PGA1", NAU8540_REG_PWR, 12, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA2", NAU8540_REG_PWR, 13, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA3", NAU8540_REG_PWR, 14, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA4", NAU8540_REG_PWR, 15, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC1", NULL, + NAU8540_REG_POWER_MANAGEMENT, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, + NAU8540_REG_POWER_MANAGEMENT, 1, 0), + SND_SOC_DAPM_ADC("ADC3", NULL, + NAU8540_REG_POWER_MANAGEMENT, 2, 0), + SND_SOC_DAPM_ADC("ADC4", NULL, + NAU8540_REG_POWER_MANAGEMENT, 3, 0), + + SND_SOC_DAPM_PGA("ADC CH1", NAU8540_REG_ANALOG_PWR, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC CH2", NAU8540_REG_ANALOG_PWR, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC CH3", NAU8540_REG_ANALOG_PWR, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC CH4", NAU8540_REG_ANALOG_PWR, 3, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Digital CH4 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch4_mux), + SND_SOC_DAPM_MUX("Digital CH3 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch3_mux), + SND_SOC_DAPM_MUX("Digital CH2 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch2_mux), + SND_SOC_DAPM_MUX("Digital CH1 Mux", + SND_SOC_NOPM, 0, 0, &digital_ch1_mux), + + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { - {"Frontend PGA1", NULL, "MIC1"}, - {"Frontend PGA2", NULL, "MIC2"}, - {"Frontend PGA3", NULL, "MIC3"}, - {"Frontend PGA4", NULL, "MIC4"}, - - {"ADC1", NULL, "Frontend PGA1"}, - {"ADC2", NULL, "Frontend PGA2"}, - {"ADC3", NULL, "Frontend PGA3"}, - {"ADC4", NULL, "Frontend PGA4"}, - - {"ADC CH1", NULL, "ADC1"}, - {"ADC CH2", NULL, "ADC2"}, - {"ADC CH3", NULL, "ADC3"}, - {"ADC CH4", NULL, "ADC4"}, - - {"ADC1", NULL, "MICBIAS1"}, - {"ADC2", NULL, "MICBIAS1"}, - {"ADC3", NULL, "MICBIAS2"}, - {"ADC4", NULL, "MICBIAS2"}, - - {"Digital CH1 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH1 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH1 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH1 Mux", "ADC channel 4", "ADC CH4"}, - - {"Digital CH2 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH2 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH2 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH2 Mux", "ADC channel 4", "ADC CH4"}, - - {"Digital CH3 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH3 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH3 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH3 Mux", "ADC channel 4", "ADC CH4"}, - - {"Digital CH4 Mux", "ADC channel 1", "ADC CH1"}, - {"Digital CH4 Mux", "ADC channel 2", "ADC CH2"}, - {"Digital CH4 Mux", "ADC channel 3", "ADC CH3"}, - {"Digital CH4 Mux", "ADC channel 4", "ADC CH4"}, - - {"AIFTX", NULL, "Digital CH1 Mux"}, - {"AIFTX", NULL, "Digital CH2 Mux"}, - {"AIFTX", NULL, "Digital CH3 Mux"}, - {"AIFTX", NULL, "Digital CH4 Mux"}, + {"Frontend PGA1", NULL, "MIC1"}, + {"Frontend PGA2", NULL, "MIC2"}, + {"Frontend PGA3", NULL, "MIC3"}, + {"Frontend PGA4", NULL, "MIC4"}, + + {"ADC1", NULL, "Frontend PGA1"}, + {"ADC2", NULL, "Frontend PGA2"}, + {"ADC3", NULL, "Frontend PGA3"}, + {"ADC4", NULL, "Frontend PGA4"}, + + {"ADC CH1", NULL, "ADC1"}, + {"ADC CH2", NULL, "ADC2"}, + {"ADC CH3", NULL, "ADC3"}, + {"ADC CH4", NULL, "ADC4"}, + + {"ADC1", NULL, "MICBIAS1"}, + {"ADC2", NULL, "MICBIAS1"}, + {"ADC3", NULL, "MICBIAS2"}, + {"ADC4", NULL, "MICBIAS2"}, + + {"Digital CH1 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH1 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH1 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH1 Mux", "ADC channel 4", "ADC CH4"}, + + {"Digital CH2 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH2 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH2 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH2 Mux", "ADC channel 4", "ADC CH4"}, + + {"Digital CH3 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH3 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH3 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH3 Mux", "ADC channel 4", "ADC CH4"}, + + {"Digital CH4 Mux", "ADC channel 1", "ADC CH1"}, + {"Digital CH4 Mux", "ADC channel 2", "ADC CH2"}, + {"Digital CH4 Mux", "ADC channel 3", "ADC CH3"}, + {"Digital CH4 Mux", "ADC channel 4", "ADC CH4"}, + + {"AIFTX", NULL, "Digital CH1 Mux"}, + {"AIFTX", NULL, "Digital CH2 Mux"}, + {"AIFTX", NULL, "Digital CH3 Mux"}, + {"AIFTX", NULL, "Digital CH4 Mux"}, }; static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) { - int osrate; + int osrate; - if (osr >= ARRAY_SIZE(osr_adc_sel)) - return -EINVAL; - osrate = osr_adc_sel[osr].osr; + if (osr >= ARRAY_SIZE(osr_adc_sel)) + return -EINVAL; + osrate = osr_adc_sel[osr].osr; - if (rate * osr > CLK_ADC_MAX) { - dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); - return -EINVAL; - } + if (rate * osr > CLK_ADC_MAX) { + dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); + return -EINVAL; + } - return 0; + return 0; } static int nau8540_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, osr; - - /* CLK_ADC = OSR * FS - * ADC clock frequency is defined as Over Sampling Rate (OSR) - * multiplied by the audio sample rate (Fs). Note that the OSR and Fs - * values must be selected such that the maximum frequency is less - * than 6.144 MHz. - */ - regmap_read(nau8540->regmap, NAU8540_REG_ADC_SAMPLE_RATE, &osr); - osr &= NAU8540_ADC_OSR_MASK; - if (nau8540_clock_check(nau8540, params_rate(params), osr)) - return -EINVAL; - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_ADC_SRC_MASK, - osr_adc_sel[osr].clk_src << NAU8540_CLK_ADC_SRC_SFT); - - switch (params_width(params)) { - case 16: - val_len |= NAU8540_I2S_DL_16; - break; - case 20: - val_len |= NAU8540_I2S_DL_20; - break; - case 24: - val_len |= NAU8540_I2S_DL_24; - break; - case 32: - val_len |= NAU8540_I2S_DL_32; - break; - default: - return -EINVAL; - } - - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, - NAU8540_I2S_DL_MASK, val_len); - - return 0; + struct snd_soc_codec *codec = dai->codec; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, osr; + + /* CLK_ADC = OSR * FS + * ADC clock frequency is defined as Over Sampling Rate (OSR) + * multiplied by the audio sample rate (Fs). Note that the OSR and Fs + * values must be selected such that the maximum frequency is less + * than 6.144 MHz. + */ + regmap_read(nau8540->regmap, NAU8540_REG_ADC_SAMPLE_RATE, &osr); + osr &= NAU8540_ADC_OSR_MASK; + if (nau8540_clock_check(nau8540, params_rate(params), osr)) + return -EINVAL; + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_ADC_SRC_MASK, + osr_adc_sel[osr].clk_src << NAU8540_CLK_ADC_SRC_SFT); + + switch (params_width(params)) { + case 16: + val_len |= NAU8540_I2S_DL_16; + break; + case 20: + val_len |= NAU8540_I2S_DL_20; + break; + case 24: + val_len |= NAU8540_I2S_DL_24; + break; + case 32: + val_len |= NAU8540_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, + NAU8540_I2S_DL_MASK, val_len); + + return 0; } static int nau8540_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - struct snd_soc_codec *codec = dai->codec; - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - unsigned int ctrl1_val = 0, ctrl2_val = 0; - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - ctrl2_val |= NAU8540_I2S_MS_MASTER; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_IB_NF: - ctrl1_val |= NAU8540_I2S_BP_INV; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - ctrl1_val |= NAU8540_I2S_DF_I2S; - break; - case SND_SOC_DAIFMT_LEFT_J: - ctrl1_val |= NAU8540_I2S_DF_LEFT; - break; - case SND_SOC_DAIFMT_RIGHT_J: - ctrl1_val |= NAU8540_I2S_DF_RIGTH; - break; - case SND_SOC_DAIFMT_DSP_A: - ctrl1_val |= NAU8540_I2S_DF_PCM_AB; - break; - case SND_SOC_DAIFMT_DSP_B: - ctrl1_val |= NAU8540_I2S_DF_PCM_AB; - ctrl1_val |= NAU8540_I2S_PCMB_EN; - break; - default: - return -EINVAL; - } - - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, - NAU8540_I2S_DL_MASK | NAU8540_I2S_DF_MASK | - NAU8540_I2S_BP_INV | NAU8540_I2S_PCMB_EN, ctrl1_val); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, - NAU8540_I2S_MS_MASK | NAU8540_I2S_DO12_OE, ctrl2_val); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, - NAU8540_I2S_DO34_OE, 0); - - return 0; + struct snd_soc_codec *codec = dai->codec; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8540_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8540_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8540_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8540_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8540_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8540_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8540_I2S_DF_PCM_AB; + ctrl1_val |= NAU8540_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL0, + NAU8540_I2S_DL_MASK | NAU8540_I2S_DF_MASK | + NAU8540_I2S_BP_INV | NAU8540_I2S_PCMB_EN, ctrl1_val); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, + NAU8540_I2S_MS_MASK | NAU8540_I2S_DO12_OE, ctrl2_val); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, + NAU8540_I2S_DO34_OE, 0); + + return 0; } /** @@ -451,55 +451,55 @@ static int nau8540_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) * Configures a DAI for TDM operation. Only support 4 slots TDM. */ static int nau8540_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct snd_soc_codec *codec = dai->codec; - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - unsigned int ctrl2_val = 0, ctrl4_val = 0; - - if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf))) - return -EINVAL; - - ctrl4_val |= (NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN); - if (tx_mask & 0xf0) { - ctrl2_val = 4 * slot_width; - ctrl4_val |= (tx_mask >> 4); - } else { - ctrl4_val |= tx_mask; - } - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL4, - NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN | - NAU8540_TDM_TX_MASK, ctrl4_val); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, - NAU8540_I2S_DO12_OE, NAU8540_I2S_DO12_OE); - regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, - NAU8540_I2S_DO34_OE | NAU8540_I2S_TSLOT_L_MASK, - NAU8540_I2S_DO34_OE | ctrl2_val); - - return 0; + struct snd_soc_codec *codec = dai->codec; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl2_val = 0, ctrl4_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf))) + return -EINVAL; + + ctrl4_val |= (NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + ctrl2_val = 4 * slot_width; + ctrl4_val |= (tx_mask >> 4); + } else { + ctrl4_val |= tx_mask; + } + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL4, + NAU8540_TDM_MODE | NAU8540_TDM_OFFSET_EN | + NAU8540_TDM_TX_MASK, ctrl4_val); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, + NAU8540_I2S_DO12_OE, NAU8540_I2S_DO12_OE); + regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, + NAU8540_I2S_DO34_OE | NAU8540_I2S_TSLOT_L_MASK, + NAU8540_I2S_DO34_OE | ctrl2_val); + + return 0; } static const struct snd_soc_dai_ops nau8540_dai_ops = { - .hw_params = nau8540_hw_params, - .set_fmt = nau8540_set_fmt, - .set_tdm_slot = nau8540_set_tdm_slot, + .hw_params = nau8540_hw_params, + .set_fmt = nau8540_set_fmt, + .set_tdm_slot = nau8540_set_tdm_slot, }; #define NAU8540_RATES SNDRV_PCM_RATE_8000_48000 #define NAU8540_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ - | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver nau8540_dai = { - .name = "nau8540-hifi", - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 4, - .rates = NAU8540_RATES, - .formats = NAU8540_FORMATS, - }, - .ops = &nau8540_dai_ops, + .name = "nau8540-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = NAU8540_RATES, + .formats = NAU8540_FORMATS, + }, + .ops = &nau8540_dai_ops, }; /** @@ -513,320 +513,320 @@ static struct snd_soc_dai_driver nau8540_dai = { * Returns 0 for success or negative error code. */ static int nau8540_calc_fll_param(unsigned int fll_in, - unsigned int fs, struct nau8540_fll *fll_param) + unsigned int fs, struct nau8540_fll *fll_param) { - u64 fvco, fvco_max; - unsigned int fref, i, fvco_sel; - - /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing - * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. - * FREF = freq_in / NAU8540_FLL_REF_DIV_MASK - */ - for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { - fref = fll_in / fll_pre_scalar[i].param; - if (fref <= NAU_FREF_MAX) - break; - } - if (i == ARRAY_SIZE(fll_pre_scalar)) - return -EINVAL; - fll_param->clk_ref_div = fll_pre_scalar[i].val; - - /* Choose the FLL ratio based on FREF */ - for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { - if (fref >= fll_ratio[i].param) - break; - } - if (i == ARRAY_SIZE(fll_ratio)) - return -EINVAL; - fll_param->ratio = fll_ratio[i].val; - - /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 124MHz or the FFL cannot be - * guaranteed across the full range of operation. - * FDCO = freq_out * 2 * mclk_src_scaling - */ - fvco_max = 0; - fvco_sel = ARRAY_SIZE(mclk_src_scaling); - for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && - fvco_max < fvco) { - fvco_max = fvco; - fvco_sel = i; - } - } - if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) - return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; - - /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional - * input based on FDCO, FREF and FLL ratio. - */ - fvco = div_u64(fvco_max << 16, fref * fll_param->ratio); - fll_param->fll_int = (fvco >> 16) & 0x3FF; - fll_param->fll_frac = fvco & 0xFFFF; - return 0; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing + * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8540_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in / fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } + } + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; + + /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco_max << 16, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 16) & 0x3FF; + fll_param->fll_frac = fvco & 0xFFFF; + return 0; } static void nau8540_fll_apply(struct regmap *regmap, - struct nau8540_fll *fll_param) + struct nau8540_fll *fll_param) { - regmap_update_bits(regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK | NAU8540_CLK_MCLK_SRC_MASK, - NAU8540_CLK_SRC_MCLK | fll_param->mclk_src); - regmap_update_bits(regmap, NAU8540_REG_FLL1, - NAU8540_FLL_RATIO_MASK, fll_param->ratio); - /* FLL 16-bit fractional input */ - regmap_write(regmap, NAU8540_REG_FLL2, fll_param->fll_frac); - /* FLL 10-bit integer input */ - regmap_update_bits(regmap, NAU8540_REG_FLL3, - NAU8540_FLL_INTEGER_MASK, fll_param->fll_int); - /* FLL pre-scaler */ - regmap_update_bits(regmap, NAU8540_REG_FLL4, - NAU8540_FLL_REF_DIV_MASK, - fll_param->clk_ref_div << NAU8540_FLL_REF_DIV_SFT); - regmap_update_bits(regmap, NAU8540_REG_FLL5, - NAU8540_FLL_CLK_SW_MASK, NAU8540_FLL_CLK_SW_REF); - regmap_update_bits(regmap, - NAU8540_REG_FLL6, NAU8540_DCO_EN, 0); - if (fll_param->fll_frac) { - regmap_update_bits(regmap, NAU8540_REG_FLL5, - NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | - NAU8540_FLL_FTR_SW_MASK, - NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | - NAU8540_FLL_FTR_SW_FILTER); - regmap_update_bits(regmap, NAU8540_REG_FLL6, - NAU8540_SDM_EN, NAU8540_SDM_EN); - } else { - regmap_update_bits(regmap, NAU8540_REG_FLL5, - NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | - NAU8540_FLL_FTR_SW_MASK, NAU8540_FLL_FTR_SW_ACCU); - regmap_update_bits(regmap, - NAU8540_REG_FLL6, NAU8540_SDM_EN, 0); - } + regmap_update_bits(regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK | NAU8540_CLK_MCLK_SRC_MASK, + NAU8540_CLK_SRC_MCLK | fll_param->mclk_src); + regmap_update_bits(regmap, NAU8540_REG_FLL1, + NAU8540_FLL_RATIO_MASK, fll_param->ratio); + /* FLL 16-bit fractional input */ + regmap_write(regmap, NAU8540_REG_FLL2, fll_param->fll_frac); + /* FLL 10-bit integer input */ + regmap_update_bits(regmap, NAU8540_REG_FLL3, + NAU8540_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(regmap, NAU8540_REG_FLL4, + NAU8540_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8540_FLL_REF_DIV_SFT); + regmap_update_bits(regmap, NAU8540_REG_FLL5, + NAU8540_FLL_CLK_SW_MASK, NAU8540_FLL_CLK_SW_REF); + regmap_update_bits(regmap, + NAU8540_REG_FLL6, NAU8540_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(regmap, NAU8540_REG_FLL5, + NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | + NAU8540_FLL_FTR_SW_MASK, + NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | + NAU8540_FLL_FTR_SW_FILTER); + regmap_update_bits(regmap, NAU8540_REG_FLL6, + NAU8540_SDM_EN, NAU8540_SDM_EN); + } else { + regmap_update_bits(regmap, NAU8540_REG_FLL5, + NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | + NAU8540_FLL_FTR_SW_MASK, NAU8540_FLL_FTR_SW_ACCU); + regmap_update_bits(regmap, + NAU8540_REG_FLL6, NAU8540_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ static int nau8540_set_pll(struct snd_soc_codec *codec, int pll_id, int source, - unsigned int freq_in, unsigned int freq_out) + unsigned int freq_in, unsigned int freq_out) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - struct nau8540_fll fll_param; - int ret, fs; - - switch (pll_id) { - case NAU8540_CLK_FLL_MCLK: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_MCLK); - break; - - case NAU8540_CLK_FLL_BLK: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_BLK); - break; - - case NAU8540_CLK_FLL_FS: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_FS); - break; - - default: - dev_err(nau8540->dev, "Invalid clock id (%d)\n", pll_id); - return -EINVAL; - } - dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", - freq_out, pll_id); - - fs = freq_out / 256; - ret = nau8540_calc_fll_param(freq_in, fs, &fll_param); - if (ret < 0) { - dev_err(nau8540->dev, "Unsupported input clock %d\n", freq_in); - return ret; - } - dev_dbg(nau8540->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", - fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, - fll_param.fll_int, fll_param.clk_ref_div); - - nau8540_fll_apply(nau8540->regmap, &fll_param); - mdelay(2); - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); - - return 0; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + struct nau8540_fll fll_param; + int ret, fs; + + switch (pll_id) { + case NAU8540_CLK_FLL_MCLK: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, + NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_MCLK); + break; + + case NAU8540_CLK_FLL_BLK: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, + NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_BLK); + break; + + case NAU8540_CLK_FLL_FS: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, + NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_FS); + break; + + default: + dev_err(nau8540->dev, "Invalid clock id (%d)\n", pll_id); + return -EINVAL; + } + dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", + freq_out, pll_id); + + fs = freq_out / 256; + ret = nau8540_calc_fll_param(freq_in, fs, &fll_param); + if (ret < 0) { + dev_err(nau8540->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_dbg(nau8540->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, + fll_param.fll_int, fll_param.clk_ref_div); + + nau8540_fll_apply(nau8540->regmap, &fll_param); + mdelay(2); + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); + + return 0; } static int nau8540_set_sysclk(struct snd_soc_codec *codec, - int clk_id, int source, unsigned int freq, int dir) + int clk_id, int source, unsigned int freq, int dir) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - - switch (clk_id) { - case NAU8540_CLK_DIS: - case NAU8540_CLK_MCLK: - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_MCLK); - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, - NAU8540_DCO_EN, 0); - break; - - case NAU8540_CLK_INTERNAL: - regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, - NAU8540_DCO_EN, NAU8540_DCO_EN); - regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, - NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); - break; - - default: - dev_err(nau8540->dev, "Invalid clock id (%d)\n", clk_id); - return -EINVAL; - } - - dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", - freq, clk_id); - - return 0; + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case NAU8540_CLK_DIS: + case NAU8540_CLK_MCLK: + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_MCLK); + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, + NAU8540_DCO_EN, 0); + break; + + case NAU8540_CLK_INTERNAL: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL6, + NAU8540_DCO_EN, NAU8540_DCO_EN); + regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, + NAU8540_CLK_SRC_MASK, NAU8540_CLK_SRC_VCO); + break; + + default: + dev_err(nau8540->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8540->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; } static void nau8540_reset_chip(struct regmap *regmap) { - regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); - regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); + regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); + regmap_write(regmap, NAU8540_REG_SW_RESET, 0x00); } static void nau8540_init_regs(struct nau8540 *nau8540) { - struct regmap *regmap = nau8540->regmap; - - /* Enable Bias/VMID/VMID Tieoff */ - regmap_update_bits(regmap, NAU8540_REG_VMID_CTRL, - NAU8540_VMID_EN | NAU8540_VMID_SEL_MASK, - NAU8540_VMID_EN | (0x2 << NAU8540_VMID_SEL_SFT)); - regmap_update_bits(regmap, NAU8540_REG_REFERENCE, - NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN, - NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN); - mdelay(2); - regmap_update_bits(regmap, NAU8540_REG_MIC_BIAS, - NAU8540_PU_PRE, NAU8540_PU_PRE); - regmap_update_bits(regmap, NAU8540_REG_CLOCK_CTRL, - NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN, - NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN); - /* ADC OSR selection, CLK_ADC = Fs * OSR */ - regmap_update_bits(regmap, NAU8540_REG_ADC_SAMPLE_RATE, - NAU8540_ADC_OSR_MASK, NAU8540_ADC_OSR_64); + struct regmap *regmap = nau8540->regmap; + + /* Enable Bias/VMID/VMID Tieoff */ + regmap_update_bits(regmap, NAU8540_REG_VMID_CTRL, + NAU8540_VMID_EN | NAU8540_VMID_SEL_MASK, + NAU8540_VMID_EN | (0x2 << NAU8540_VMID_SEL_SFT)); + regmap_update_bits(regmap, NAU8540_REG_REFERENCE, + NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN, + NAU8540_PRECHARGE_DIS | NAU8540_GLOBAL_BIAS_EN); + mdelay(2); + regmap_update_bits(regmap, NAU8540_REG_MIC_BIAS, + NAU8540_PU_PRE, NAU8540_PU_PRE); + regmap_update_bits(regmap, NAU8540_REG_CLOCK_CTRL, + NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN, + NAU8540_CLK_ADC_EN | NAU8540_CLK_I2S_EN); + /* ADC OSR selection, CLK_ADC = Fs * OSR */ + regmap_update_bits(regmap, NAU8540_REG_ADC_SAMPLE_RATE, + NAU8540_ADC_OSR_MASK, NAU8540_ADC_OSR_64); } static int __maybe_unused nau8540_suspend(struct snd_soc_codec *codec) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(nau8540->regmap, true); - regcache_mark_dirty(nau8540->regmap); + regcache_cache_only(nau8540->regmap, true); + regcache_mark_dirty(nau8540->regmap); - return 0; + return 0; } static int __maybe_unused nau8540_resume(struct snd_soc_codec *codec) { - struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); + struct nau8540 *nau8540 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(nau8540->regmap, false); - regcache_sync(nau8540->regmap); + regcache_cache_only(nau8540->regmap, false); + regcache_sync(nau8540->regmap); - return 0; + return 0; } static struct snd_soc_codec_driver nau8540_codec_driver = { - .set_sysclk = nau8540_set_sysclk, - .set_pll = nau8540_set_pll, - .suspend = nau8540_suspend, - .resume = nau8540_resume, - .suspend_bias_off = true, - - .component_driver = { - .controls = nau8540_snd_controls, - .num_controls = ARRAY_SIZE(nau8540_snd_controls), - .dapm_widgets = nau8540_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(nau8540_dapm_widgets), - .dapm_routes = nau8540_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(nau8540_dapm_routes), - }, + .set_sysclk = nau8540_set_sysclk, + .set_pll = nau8540_set_pll, + .suspend = nau8540_suspend, + .resume = nau8540_resume, + .suspend_bias_off = true, + + .component_driver = { + .controls = nau8540_snd_controls, + .num_controls = ARRAY_SIZE(nau8540_snd_controls), + .dapm_widgets = nau8540_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8540_dapm_widgets), + .dapm_routes = nau8540_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8540_dapm_routes), + }, }; static const struct regmap_config nau8540_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = 16, + .reg_bits = 16, - .max_register = NAU8540_REG_MAX, - .readable_reg = nau8540_readable_reg, - .writeable_reg = nau8540_writeable_reg, - .volatile_reg = nau8540_volatile_reg, + .max_register = NAU8540_REG_MAX, + .readable_reg = nau8540_readable_reg, + .writeable_reg = nau8540_writeable_reg, + .volatile_reg = nau8540_volatile_reg, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = nau8540_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(nau8540_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8540_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8540_reg_defaults), }; static int nau8540_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { - struct device *dev = &i2c->dev; - struct nau8540 *nau8540 = dev_get_platdata(dev); - int ret, value; - - if (!nau8540) { - nau8540 = devm_kzalloc(dev, sizeof(*nau8540), GFP_KERNEL); - if (!nau8540) - return -ENOMEM; - } - i2c_set_clientdata(i2c, nau8540); - - nau8540->regmap = devm_regmap_init_i2c(i2c, &nau8540_regmap_config); - if (IS_ERR(nau8540->regmap)) - return PTR_ERR(nau8540->regmap); - ret = regmap_read(nau8540->regmap, NAU8540_REG_I2C_DEVICE_ID, &value); - if (ret < 0) { - dev_err(dev, "Failed to read device id from the NAU85L40: %d\n", - ret); - return ret; - } - - nau8540->dev = dev; - nau8540_reset_chip(nau8540->regmap); - nau8540_init_regs(nau8540); - - return snd_soc_register_codec(dev, - &nau8540_codec_driver, &nau8540_dai, 1); + struct device *dev = &i2c->dev; + struct nau8540 *nau8540 = dev_get_platdata(dev); + int ret, value; + + if (!nau8540) { + nau8540 = devm_kzalloc(dev, sizeof(*nau8540), GFP_KERNEL); + if (!nau8540) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8540); + + nau8540->regmap = devm_regmap_init_i2c(i2c, &nau8540_regmap_config); + if (IS_ERR(nau8540->regmap)) + return PTR_ERR(nau8540->regmap); + ret = regmap_read(nau8540->regmap, NAU8540_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU85L40: %d\n", + ret); + return ret; + } + + nau8540->dev = dev; + nau8540_reset_chip(nau8540->regmap); + nau8540_init_regs(nau8540); + + return snd_soc_register_codec(dev, + &nau8540_codec_driver, &nau8540_dai, 1); } static int nau8540_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); - return 0; + snd_soc_unregister_codec(&client->dev); + return 0; } static const struct i2c_device_id nau8540_i2c_ids[] = { - { "nau8540", 0 }, - { } + { "nau8540", 0 }, + { } }; MODULE_DEVICE_TABLE(i2c, nau8540_i2c_ids); #ifdef CONFIG_OF static const struct of_device_id nau8540_of_ids[] = { - { .compatible = "nuvoton,nau8540", }, - {} + { .compatible = "nuvoton,nau8540", }, + {} }; MODULE_DEVICE_TABLE(of, nau8540_of_ids); #endif static struct i2c_driver nau8540_i2c_driver = { - .driver = { - .name = "nau8540", - .of_match_table = of_match_ptr(nau8540_of_ids), - }, - .probe = nau8540_i2c_probe, - .remove = nau8540_i2c_remove, - .id_table = nau8540_i2c_ids, + .driver = { + .name = "nau8540", + .of_match_table = of_match_ptr(nau8540_of_ids), + }, + .probe = nau8540_i2c_probe, + .remove = nau8540_i2c_remove, + .id_table = nau8540_i2c_ids, }; module_i2c_driver(nau8540_i2c_driver); diff --git a/sound/soc/codecs/nau8540.h b/sound/soc/codecs/nau8540.h index d06e651..5db5b22 100644 --- a/sound/soc/codecs/nau8540.h +++ b/sound/soc/codecs/nau8540.h @@ -12,211 +12,211 @@ #ifndef __NAU8540_H__ #define __NAU8540_H__ -#define NAU8540_REG_SW_RESET 0x00 -#define NAU8540_REG_POWER_MANAGEMENT 0x01 -#define NAU8540_REG_CLOCK_CTRL 0x02 -#define NAU8540_REG_CLOCK_SRC 0x03 -#define NAU8540_REG_FLL1 0x04 -#define NAU8540_REG_FLL2 0x05 -#define NAU8540_REG_FLL3 0x06 -#define NAU8540_REG_FLL4 0x07 -#define NAU8540_REG_FLL5 0x08 -#define NAU8540_REG_FLL6 0x09 -#define NAU8540_REG_FLL_VCO_RSV 0x0A -#define NAU8540_REG_PCM_CTRL0 0x10 -#define NAU8540_REG_PCM_CTRL1 0x11 -#define NAU8540_REG_PCM_CTRL2 0x12 -#define NAU8540_REG_PCM_CTRL3 0x13 -#define NAU8540_REG_PCM_CTRL4 0x14 -#define NAU8540_REG_ALC_CONTROL_1 0x20 -#define NAU8540_REG_ALC_CONTROL_2 0x21 -#define NAU8540_REG_ALC_CONTROL_3 0x22 -#define NAU8540_REG_ALC_CONTROL_4 0x23 -#define NAU8540_REG_ALC_CONTROL_5 0x24 -#define NAU8540_REG_ALC_GAIN_CH12 0x2D -#define NAU8540_REG_ALC_GAIN_CH34 0x2E -#define NAU8540_REG_ALC_STATUS 0x2F -#define NAU8540_REG_NOTCH_FIL1_CH1 0x30 -#define NAU8540_REG_NOTCH_FIL2_CH1 0x31 -#define NAU8540_REG_NOTCH_FIL1_CH2 0x32 -#define NAU8540_REG_NOTCH_FIL2_CH2 0x33 -#define NAU8540_REG_NOTCH_FIL1_CH3 0x34 -#define NAU8540_REG_NOTCH_FIL2_CH3 0x35 -#define NAU8540_REG_NOTCH_FIL1_CH4 0x36 -#define NAU8540_REG_NOTCH_FIL2_CH4 0x37 -#define NAU8540_REG_HPF_FILTER_CH12 0x38 -#define NAU8540_REG_HPF_FILTER_CH34 0x39 -#define NAU8540_REG_ADC_SAMPLE_RATE 0x3A -#define NAU8540_REG_DIGITAL_GAIN_CH1 0x40 -#define NAU8540_REG_DIGITAL_GAIN_CH2 0x41 -#define NAU8540_REG_DIGITAL_GAIN_CH3 0x42 -#define NAU8540_REG_DIGITAL_GAIN_CH4 0x43 -#define NAU8540_REG_DIGITAL_MUX 0x44 -#define NAU8540_REG_P2P_CH1 0x48 -#define NAU8540_REG_P2P_CH2 0x49 -#define NAU8540_REG_P2P_CH3 0x4A -#define NAU8540_REG_P2P_CH4 0x4B -#define NAU8540_REG_PEAK_CH1 0x4C -#define NAU8540_REG_PEAK_CH2 0x4D -#define NAU8540_REG_PEAK_CH3 0x4E -#define NAU8540_REG_PEAK_CH4 0x4F -#define NAU8540_REG_GPIO_CTRL 0x50 -#define NAU8540_REG_MISC_CTRL 0x51 -#define NAU8540_REG_I2C_CTRL 0x52 -#define NAU8540_REG_I2C_DEVICE_ID 0x58 -#define NAU8540_REG_RST 0x5A -#define NAU8540_REG_VMID_CTRL 0x60 -#define NAU8540_REG_MUTE 0x61 -#define NAU8540_REG_ANALOG_ADC1 0x64 -#define NAU8540_REG_ANALOG_ADC2 0x65 -#define NAU8540_REG_ANALOG_PWR 0x66 -#define NAU8540_REG_MIC_BIAS 0x67 -#define NAU8540_REG_REFERENCE 0x68 -#define NAU8540_REG_FEPGA1 0x69 -#define NAU8540_REG_FEPGA2 0x6A -#define NAU8540_REG_FEPGA3 0x6B -#define NAU8540_REG_FEPGA4 0x6C -#define NAU8540_REG_PWR 0x6D -#define NAU8540_REG_MAX NAU8540_REG_PWR +#define NAU8540_REG_SW_RESET 0x00 +#define NAU8540_REG_POWER_MANAGEMENT 0x01 +#define NAU8540_REG_CLOCK_CTRL 0x02 +#define NAU8540_REG_CLOCK_SRC 0x03 +#define NAU8540_REG_FLL1 0x04 +#define NAU8540_REG_FLL2 0x05 +#define NAU8540_REG_FLL3 0x06 +#define NAU8540_REG_FLL4 0x07 +#define NAU8540_REG_FLL5 0x08 +#define NAU8540_REG_FLL6 0x09 +#define NAU8540_REG_FLL_VCO_RSV 0x0A +#define NAU8540_REG_PCM_CTRL0 0x10 +#define NAU8540_REG_PCM_CTRL1 0x11 +#define NAU8540_REG_PCM_CTRL2 0x12 +#define NAU8540_REG_PCM_CTRL3 0x13 +#define NAU8540_REG_PCM_CTRL4 0x14 +#define NAU8540_REG_ALC_CONTROL_1 0x20 +#define NAU8540_REG_ALC_CONTROL_2 0x21 +#define NAU8540_REG_ALC_CONTROL_3 0x22 +#define NAU8540_REG_ALC_CONTROL_4 0x23 +#define NAU8540_REG_ALC_CONTROL_5 0x24 +#define NAU8540_REG_ALC_GAIN_CH12 0x2D +#define NAU8540_REG_ALC_GAIN_CH34 0x2E +#define NAU8540_REG_ALC_STATUS 0x2F +#define NAU8540_REG_NOTCH_FIL1_CH1 0x30 +#define NAU8540_REG_NOTCH_FIL2_CH1 0x31 +#define NAU8540_REG_NOTCH_FIL1_CH2 0x32 +#define NAU8540_REG_NOTCH_FIL2_CH2 0x33 +#define NAU8540_REG_NOTCH_FIL1_CH3 0x34 +#define NAU8540_REG_NOTCH_FIL2_CH3 0x35 +#define NAU8540_REG_NOTCH_FIL1_CH4 0x36 +#define NAU8540_REG_NOTCH_FIL2_CH4 0x37 +#define NAU8540_REG_HPF_FILTER_CH12 0x38 +#define NAU8540_REG_HPF_FILTER_CH34 0x39 +#define NAU8540_REG_ADC_SAMPLE_RATE 0x3A +#define NAU8540_REG_DIGITAL_GAIN_CH1 0x40 +#define NAU8540_REG_DIGITAL_GAIN_CH2 0x41 +#define NAU8540_REG_DIGITAL_GAIN_CH3 0x42 +#define NAU8540_REG_DIGITAL_GAIN_CH4 0x43 +#define NAU8540_REG_DIGITAL_MUX 0x44 +#define NAU8540_REG_P2P_CH1 0x48 +#define NAU8540_REG_P2P_CH2 0x49 +#define NAU8540_REG_P2P_CH3 0x4A +#define NAU8540_REG_P2P_CH4 0x4B +#define NAU8540_REG_PEAK_CH1 0x4C +#define NAU8540_REG_PEAK_CH2 0x4D +#define NAU8540_REG_PEAK_CH3 0x4E +#define NAU8540_REG_PEAK_CH4 0x4F +#define NAU8540_REG_GPIO_CTRL 0x50 +#define NAU8540_REG_MISC_CTRL 0x51 +#define NAU8540_REG_I2C_CTRL 0x52 +#define NAU8540_REG_I2C_DEVICE_ID 0x58 +#define NAU8540_REG_RST 0x5A +#define NAU8540_REG_VMID_CTRL 0x60 +#define NAU8540_REG_MUTE 0x61 +#define NAU8540_REG_ANALOG_ADC1 0x64 +#define NAU8540_REG_ANALOG_ADC2 0x65 +#define NAU8540_REG_ANALOG_PWR 0x66 +#define NAU8540_REG_MIC_BIAS 0x67 +#define NAU8540_REG_REFERENCE 0x68 +#define NAU8540_REG_FEPGA1 0x69 +#define NAU8540_REG_FEPGA2 0x6A +#define NAU8540_REG_FEPGA3 0x6B +#define NAU8540_REG_FEPGA4 0x6C +#define NAU8540_REG_PWR 0x6D +#define NAU8540_REG_MAX NAU8540_REG_PWR /* POWER_MANAGEMENT (0x01) */ -#define NAU8540_ADC4_EN (0x1 << 3) -#define NAU8540_ADC3_EN (0x1 << 2) -#define NAU8540_ADC2_EN (0x1 << 1) -#define NAU8540_ADC1_EN 0x1 +#define NAU8540_ADC4_EN (0x1 << 3) +#define NAU8540_ADC3_EN (0x1 << 2) +#define NAU8540_ADC2_EN (0x1 << 1) +#define NAU8540_ADC1_EN 0x1 /* CLOCK_CTRL (0x02) */ -#define NAU8540_CLK_ADC_EN (0x1 << 15) -#define NAU8540_CLK_I2S_EN (0x1 << 1) +#define NAU8540_CLK_ADC_EN (0x1 << 15) +#define NAU8540_CLK_I2S_EN (0x1 << 1) /* CLOCK_SRC (0x03) */ -#define NAU8540_CLK_SRC_SFT 15 -#define NAU8540_CLK_SRC_MASK (1 << NAU8540_CLK_SRC_SFT) -#define NAU8540_CLK_SRC_VCO (1 << NAU8540_CLK_SRC_SFT) -#define NAU8540_CLK_SRC_MCLK (0 << NAU8540_CLK_SRC_SFT) -#define NAU8540_CLK_ADC_SRC_SFT 6 -#define NAU8540_CLK_ADC_SRC_MASK (0x3 << NAU8540_CLK_ADC_SRC_SFT) -#define NAU8540_CLK_MCLK_SRC_MASK 0xf +#define NAU8540_CLK_SRC_SFT 15 +#define NAU8540_CLK_SRC_MASK (1 << NAU8540_CLK_SRC_SFT) +#define NAU8540_CLK_SRC_VCO (1 << NAU8540_CLK_SRC_SFT) +#define NAU8540_CLK_SRC_MCLK (0 << NAU8540_CLK_SRC_SFT) +#define NAU8540_CLK_ADC_SRC_SFT 6 +#define NAU8540_CLK_ADC_SRC_MASK (0x3 << NAU8540_CLK_ADC_SRC_SFT) +#define NAU8540_CLK_MCLK_SRC_MASK 0xf /* FLL1 (0x04) */ -#define NAU8540_FLL_RATIO_MASK 0x7f +#define NAU8540_FLL_RATIO_MASK 0x7f /* FLL3 (0x06) */ -#define NAU8540_FLL_CLK_SRC_SFT 10 -#define NAU8540_FLL_CLK_SRC_MASK (0x3 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_CLK_SRC_MCLK (0 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_CLK_SRC_BLK (0x2 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_CLK_SRC_FS (0x3 << NAU8540_FLL_CLK_SRC_SFT) -#define NAU8540_FLL_INTEGER_MASK 0x3ff +#define NAU8540_FLL_CLK_SRC_SFT 10 +#define NAU8540_FLL_CLK_SRC_MASK (0x3 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_CLK_SRC_MCLK (0 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_CLK_SRC_BLK (0x2 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_CLK_SRC_FS (0x3 << NAU8540_FLL_CLK_SRC_SFT) +#define NAU8540_FLL_INTEGER_MASK 0x3ff /* FLL4 (0x07) */ -#define NAU8540_FLL_REF_DIV_SFT 10 -#define NAU8540_FLL_REF_DIV_MASK (0x3 << NAU8540_FLL_REF_DIV_SFT) +#define NAU8540_FLL_REF_DIV_SFT 10 +#define NAU8540_FLL_REF_DIV_MASK (0x3 << NAU8540_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ -#define NAU8540_FLL_PDB_DAC_EN (0x1 << 15) -#define NAU8540_FLL_LOOP_FTR_EN (0x1 << 14) -#define NAU8540_FLL_CLK_SW_MASK (0x1 << 13) -#define NAU8540_FLL_CLK_SW_N2 (0x1 << 13) -#define NAU8540_FLL_CLK_SW_REF (0x0 << 13) -#define NAU8540_FLL_FTR_SW_MASK (0x1 << 12) -#define NAU8540_FLL_FTR_SW_ACCU (0x1 << 12) -#define NAU8540_FLL_FTR_SW_FILTER (0x0 << 12) +#define NAU8540_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8540_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8540_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8540_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8540_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8540_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8540_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8540_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8540_DCO_EN (0x1 << 15) -#define NAU8540_SDM_EN (0x1 << 14) +#define NAU8540_DCO_EN (0x1 << 15) +#define NAU8540_SDM_EN (0x1 << 14) /* PCM_CTRL0 (0x10) */ -#define NAU8540_I2S_BP_SFT 7 -#define NAU8540_I2S_BP_INV (0x1 << NAU8540_I2S_BP_SFT) -#define NAU8540_I2S_PCMB_SFT 6 -#define NAU8540_I2S_PCMB_EN (0x1 << NAU8540_I2S_PCMB_SFT) -#define NAU8540_I2S_DL_SFT 2 -#define NAU8540_I2S_DL_MASK (0x3 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_16 (0 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_20 (0x1 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_24 (0x2 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DL_32 (0x3 << NAU8540_I2S_DL_SFT) -#define NAU8540_I2S_DF_MASK 0x3 -#define NAU8540_I2S_DF_RIGTH 0 -#define NAU8540_I2S_DF_LEFT 0x1 -#define NAU8540_I2S_DF_I2S 0x2 -#define NAU8540_I2S_DF_PCM_AB 0x3 +#define NAU8540_I2S_BP_SFT 7 +#define NAU8540_I2S_BP_INV (0x1 << NAU8540_I2S_BP_SFT) +#define NAU8540_I2S_PCMB_SFT 6 +#define NAU8540_I2S_PCMB_EN (0x1 << NAU8540_I2S_PCMB_SFT) +#define NAU8540_I2S_DL_SFT 2 +#define NAU8540_I2S_DL_MASK (0x3 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_16 (0 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_20 (0x1 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_24 (0x2 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DL_32 (0x3 << NAU8540_I2S_DL_SFT) +#define NAU8540_I2S_DF_MASK 0x3 +#define NAU8540_I2S_DF_RIGTH 0 +#define NAU8540_I2S_DF_LEFT 0x1 +#define NAU8540_I2S_DF_I2S 0x2 +#define NAU8540_I2S_DF_PCM_AB 0x3 /* PCM_CTRL1 (0x11) */ -#define NAU8540_I2S_LRC_DIV_SFT 12 -#define NAU8540_I2S_LRC_DIV_MASK (0x3 << NAU8540_I2S_LRC_DIV_SFT) -#define NAU8540_I2S_DO12_OE (0x1 << 4) -#define NAU8540_I2S_MS_SFT 3 -#define NAU8540_I2S_MS_MASK (0x1 << NAU8540_I2S_MS_SFT) -#define NAU8540_I2S_MS_MASTER (0x1 << NAU8540_I2S_MS_SFT) -#define NAU8540_I2S_MS_SLAVE (0x0 << NAU8540_I2S_MS_SFT) -#define NAU8540_I2S_BLK_DIV_MASK 0x7 +#define NAU8540_I2S_LRC_DIV_SFT 12 +#define NAU8540_I2S_LRC_DIV_MASK (0x3 << NAU8540_I2S_LRC_DIV_SFT) +#define NAU8540_I2S_DO12_OE (0x1 << 4) +#define NAU8540_I2S_MS_SFT 3 +#define NAU8540_I2S_MS_MASK (0x1 << NAU8540_I2S_MS_SFT) +#define NAU8540_I2S_MS_MASTER (0x1 << NAU8540_I2S_MS_SFT) +#define NAU8540_I2S_MS_SLAVE (0x0 << NAU8540_I2S_MS_SFT) +#define NAU8540_I2S_BLK_DIV_MASK 0x7 /* PCM_CTRL1 (0x12) */ -#define NAU8540_I2S_DO34_OE (0x1 << 11) -#define NAU8540_I2S_TSLOT_L_MASK 0x3ff +#define NAU8540_I2S_DO34_OE (0x1 << 11) +#define NAU8540_I2S_TSLOT_L_MASK 0x3ff /* PCM_CTRL4 (0x14) */ -#define NAU8540_TDM_MODE (0x1 << 15) -#define NAU8540_TDM_OFFSET_EN (0x1 << 14) -#define NAU8540_TDM_TX_MASK 0xf +#define NAU8540_TDM_MODE (0x1 << 15) +#define NAU8540_TDM_OFFSET_EN (0x1 << 14) +#define NAU8540_TDM_TX_MASK 0xf /* ADC_SAMPLE_RATE (0x3A) */ -#define NAU8540_ADC_OSR_MASK 0x3 -#define NAU8540_ADC_OSR_256 0x3 -#define NAU8540_ADC_OSR_128 0x2 -#define NAU8540_ADC_OSR_64 0x1 -#define NAU8540_ADC_OSR_32 0x0 +#define NAU8540_ADC_OSR_MASK 0x3 +#define NAU8540_ADC_OSR_256 0x3 +#define NAU8540_ADC_OSR_128 0x2 +#define NAU8540_ADC_OSR_64 0x1 +#define NAU8540_ADC_OSR_32 0x0 /* VMID_CTRL (0x60) */ -#define NAU8540_VMID_EN (1 << 6) -#define NAU8540_VMID_SEL_SFT 4 -#define NAU8540_VMID_SEL_MASK (0x3 << NAU8540_VMID_SEL_SFT) +#define NAU8540_VMID_EN (1 << 6) +#define NAU8540_VMID_SEL_SFT 4 +#define NAU8540_VMID_SEL_MASK (0x3 << NAU8540_VMID_SEL_SFT) /* MIC_BIAS (0x67) */ -#define NAU8540_PU_PRE (0x1 << 8) +#define NAU8540_PU_PRE (0x1 << 8) /* REFERENCE (0x68) */ -#define NAU8540_PRECHARGE_DIS (0x1 << 13) -#define NAU8540_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8540_PRECHARGE_DIS (0x1 << 13) +#define NAU8540_GLOBAL_BIAS_EN (0x1 << 12) /* System Clock Source */ enum { - NAU8540_CLK_DIS, - NAU8540_CLK_MCLK, - NAU8540_CLK_INTERNAL, - NAU8540_CLK_FLL_MCLK, - NAU8540_CLK_FLL_BLK, - NAU8540_CLK_FLL_FS, + NAU8540_CLK_DIS, + NAU8540_CLK_MCLK, + NAU8540_CLK_INTERNAL, + NAU8540_CLK_FLL_MCLK, + NAU8540_CLK_FLL_BLK, + NAU8540_CLK_FLL_FS, }; struct nau8540 { - struct device *dev; - struct regmap *regmap; + struct device *dev; + struct regmap *regmap; }; struct nau8540_fll { - int mclk_src; - int ratio; - int fll_frac; - int fll_int; - int clk_ref_div; + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; }; struct nau8540_fll_attr { - unsigned int param; - unsigned int val; + unsigned int param; + unsigned int val; }; /* over sampling rate */ struct nau8540_osr_attr { - unsigned int osr; - unsigned int clk_src; + unsigned int osr; + unsigned int clk_src; }; -#endif /* __NAU8540_H__ */ +#endif /* __NAU8540_H__ */ -- 2.6.4

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[alsa-devel] [resend][PATCH v5 00/10] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 18 Apr '17

18 Apr '17

Hi Rob, Mark +2 weeks passed. I will re-post these patches. 1) - 4): OF-graph base patches (= Rot) 5) - 10): ALSA SoC patches (= Mark) Kuninori Morimoto (10): 1) of_graph: export symbol of_phandle_iterator_init/next 2) of_graph: add of_graph_get_remote_endpoint() 3) of_graph: add of_graph_get_port_parent() 4) of_graph: add of_graph_get_endpoint_count() 5) ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name 6) ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() 7) ASoC: add snd_soc_get_dai_id() 8) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 9) ASoC: add audio-graph-card document 10) ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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Re: [alsa-devel] [PATCH] ASoC: rt286: fix headphone click/crack noise on Dell XPS 9343 I2S mode
by Garrett Thornburg 18 Apr '17

18 Apr '17

This is my first time posting in any type of mailing list, so hopefully I'm doing this right. I recently encountered the audio trouble with I2S mode via the headphone jack on my Dell XPS 9343 as well and found your patch. I applied the patch to the latest stable kernel (4.10) on arch linux mostly using their tools and the repeated clicking sound is fixed when no sound is playing. However, sometimes when sound is playing, the clicking returns but it's much more mild. If I pause/ resume music, for example, it might "click.. click.. click.." very quietly, but not always. It's audible when I turn the volume level down low, but overall it's much better. I do still get somewhat loud pops when resuming music after the sound card has been idle for a while, or a pop when the card switches to an idle state. This patch has really improved my situation though, so thank you :).

1 0

[alsa-devel] [PATCH v4 6/6] sound: replace /proc/bus/usb by /dev/bus/usb
by Mauro Carvalho Chehab 17 Apr '17

17 Apr '17

The /proc/bus/usb devices don't exist anymore, since when we got rid of usbfs. Those devices are now seen at /dev/bus/usb. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> --- sound/usb/usx2y/us122l.c | 2 +- sound/usb/usx2y/usX2Yhwdep.c | 2 +- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index cf45bf1f7ee0..e118bdca983d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -472,7 +472,7 @@ static int usb_stream_hwdep_new(struct snd_card *card) hw->ops.mmap = usb_stream_hwdep_mmap; hw->ops.poll = usb_stream_hwdep_poll; - sprintf(hw->name, "/proc/bus/usb/%03d/%03d/hwdeppcm", + sprintf(hw->name, "/dev/bus/usb/%03d/%03d/hwdeppcm", dev->bus->busnum, dev->devnum); return 0; } diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 605e1047c01d..f4b3cda412fc 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -258,7 +258,7 @@ int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device) hw->ops.mmap = snd_us428ctls_mmap; hw->ops.poll = snd_us428ctls_poll; hw->exclusive = 1; - sprintf(hw->name, "/proc/bus/usb/%03d/%03d", device->bus->busnum, device->devnum); + sprintf(hw->name, "/dev/bus/usb/%03d/%03d", device->bus->busnum, device->devnum); return 0; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f95164b91152..d51c7fd7835b 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -723,7 +723,7 @@ int usX2Y_hwdep_pcm_new(struct snd_card *card) hw->ops.release = snd_usX2Y_hwdep_pcm_release; hw->ops.mmap = snd_usX2Y_hwdep_pcm_mmap; hw->exclusive = 1; - sprintf(hw->name, "/proc/bus/usb/%03d/%03d/hwdeppcm", dev->bus->busnum, dev->devnum); + sprintf(hw->name, "/dev/bus/usb/%03d/%03d/hwdeppcm", dev->bus->busnum, dev->devnum); err = snd_pcm_new(card, NAME_ALLCAPS" hwdep Audio", 2, 1, 1, &pcm); if (err < 0) { -- 2.9.3

2 1

Re: [alsa-devel] [PATCH 2/3] docs-rst/usb: usb/devices is now at /sys/kernel/debug/
by Mauro Carvalho Chehab 16 Apr '17

16 Apr '17

Em Sun, 16 Apr 2017 10:39:58 -0400 (EDT) Alan Stern <stern(a)rowland.harvard.edu> escreveu: > On Sun, 16 Apr 2017, Mauro Carvalho Chehab wrote: > > > The /proc/bus/usb/devices got moved to sysfs. It is now > > sitting at: > > /sys/kernel/debug/usb/devices > > > > Fix the old references at the USB documentation > > Many of the things you changed are still out of date. > > > Signed-off-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> > > --- > > Documentation/driver-api/usb/usb.rst | 14 +++++++------- > > Documentation/usb/acm.txt | 2 +- > > Documentation/usb/gadget_serial.txt | 4 ++-- > > Documentation/usb/proc_usb_info.txt | 20 ++++++++++---------- > > 4 files changed, 20 insertions(+), 20 deletions(-) > > > > diff --git a/Documentation/driver-api/usb/usb.rst b/Documentation/driver-api/usb/usb.rst > > index 6824089ef4c8..036e77038d42 100644 > > --- a/Documentation/driver-api/usb/usb.rst > > +++ b/Documentation/driver-api/usb/usb.rst > > @@ -233,7 +233,7 @@ What files are in "usbfs"? > > > > Conventionally mounted at ``/proc/bus/usb``, usbfs features include: > > This whole section is highly misleading. usbfs no longer exists as a > filesystem. Instead, the files which used to be part of usbfs have > either moved to debugfs (/sys/kernel/debug/usb/devices) or are > character device nodes managed by systemctl/udev and typically created > under /dev/bus/usb. > > > @@ -299,23 +299,23 @@ invocations of *mount*, are: > > 0) > > > > *listgid*\ =NNNNN > > - Controls the GID used for the /proc/bus/usb/devices and drivers > > + Controls the GID used for the /sys/kernel/debug/usb/devices and drivers > > files. (Default: 0) > > > > *listmode*\ =MMM > > - Controls the file mode used for the /proc/bus/usb/devices and > > + Controls the file mode used for the /sys/kernel/debug/usb/devices and > > drivers files. (Default: 0444) > > > > *listuid*\ =NNNNN > > - Controls the UID used for the /proc/bus/usb/devices and drivers > > + Controls the UID used for the /sys/kernel/debug/usb/devices and drivers > > files. (Default: 0) > > > > Note that many Linux distributions hard-wire the mount options for usbfs > > in their init scripts, such as ``/etc/rc.d/rc.sysinit``, rather than > > making it easy to set this per-system policy in ``/etc/fstab``. > > This is all totally out of date. usbfs is no longer a filesystem that > can be mounted. True. Sent already a new patchset covering it, and other places where /proc/bus/bus are mentioned. Besides the stuff already fixed, there are still a few other bits to address: Documentation/usb/proc_usb_info.txt:/proc/bus/usb filesystem output Documentation/usb/proc_usb_info.txt:/proc/bus/usb. It provides the /sys/kernel/debug/usb/devices file, as well as Documentation/usb/proc_usb_info.txt:the /proc/bus/usb/BBB/DDD files. Documentation/usb/proc_usb_info.txt:**NOTE**: If /proc/bus/usb appears empty, and a host controller Documentation/usb/proc_usb_info.txt: mount -t usbfs none /proc/bus/usb Documentation/usb/proc_usb_info.txt: none /proc/bus/usb usbfs defaults 0 0 Documentation/usb/proc_usb_info.txt:THE /proc/bus/usb/BBB/DDD FILES: Documentation/usb/proc_usb_info.txt:devices. You would open the /proc/bus/usb/BBB/DDD file read/write, This should likely be merged at usb.rst file or fully reworked. include/uapi/linux/capability.h:/* Allow sending USB messages to any device via /proc/bus/usb */ include/uapi/linux/usb/ch9.h: * not other descriptors) are read from usbfs (i.e. /proc/bus/usb/BBB/DDD); I'll address this on a version 4. sound/usb/usx2y/us122l.c: sprintf(hw->name, "/proc/bus/usb/%03d/%03d/hwdeppcm", sound/usb/usx2y/usX2Yhwdep.c: sprintf(hw->name, "/proc/bus/usb/%03d/%03d", device->bus->busnum, device->devnum); sound/usb/usx2y/usx2yhwdeppcm.c: sprintf(hw->name, "/proc/bus/usb/%03d/%03d/hwdeppcm", dev->bus->busnum, dev->devnum); Not sure about the above. Regards, Mauro

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Re: [alsa-devel] Internal Speaker problem
by Laszlo Papp 15 Apr '17

15 Apr '17

Guys, seriously, no one willing to help with a completely broken setup?! It works fine on Windows, so I would really like to get it working on Linux, too. On Mon, Mar 27, 2017 at 8:29 AM, Laszlo Papp <lpapp(a)kde.org> wrote: > I am sorry about flooding with my emails. I just thought that I would > amend some information that I forgot to mention in my original email. > > The same laptop and internal speaker work ok on Windows 10. Also, the > setup used to work about 1-2 months ago. I cannot remember what exactly > broke, perhaps a system upgrade. If it matters, I am using Archlinux. > > On Mon, Mar 27, 2017 at 8:27 AM, Laszlo Papp <lpapp(a)kde.org> wrote: > >> Dear Alsa Developers, >> >> My internal speaker in the Lenovo Thinkpad T510 laptop stopped working. >> Normally, it would provide either no sound or just for a couple of seconds >> and then it would go off. >> >> I do not have auto-mute enabled. I checked it with alsamixer that it is >> disabled. >> >> My ALSA information is located at http://www.alsa-project.org/db >> /?f=87547de5ff55e44e360aa2382d4e5d3b7bbed091 >> >> Do you know how I could fix this? >> >> Ys, L. >> >> >

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[alsa-devel] [PATCH 1/3] ASoC: rt5514: Mark rt5514_i2c_driver as static
by Douglas Anderson 14 Apr '17

14 Apr '17

There's no reason for rt5514_i2c_driver to be non-static. Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index b281a46d769d..481e77763fe4 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1149,7 +1149,7 @@ static int rt5514_i2c_remove(struct i2c_client *i2c) return 0; } -struct i2c_driver rt5514_i2c_driver = { +static struct i2c_driver rt5514_i2c_driver = { .driver = { .name = "rt5514", .of_match_table = of_match_ptr(rt5514_of_match), -- 2.12.2.762.g0e3151a226-goog

2 5

[alsa-devel] [PATCH] ALSA: fireface: obsolete usage of ktime_set() for zero assignment
by Takashi Sakamoto 14 Apr '17

14 Apr '17

In development period for Linux v4.10, ktime_t became an alias of s64, instead of union. I forgot it. We can just assign zero, instead of usage of ktime_set(0, 0). Fixes: 19174295788 ("ALSA: fireface: add transaction support") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-transaction.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index d1b098f..dd6c8e8 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -21,7 +21,7 @@ static void finish_transmit_midi_msg(struct snd_ff *ff, unsigned int port, if (rcode != RCODE_COMPLETE) { /* Transfer the message again, immediately. */ - ff->next_ktime[port] = ktime_set(0, 0); + ff->next_ktime[port] = 0; schedule_work(&ff->rx_midi_work[port]); return; } -- 2.9.3

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[alsa-devel] [PATCH v2 0/8] ALSA: firewire-lib/firewire-tascam: localize async midi port
by Takashi Sakamoto 14 Apr '17

14 Apr '17

Hi, This patchset is a revised version of my previous one: [alsa-devel] [PATCH 0/8] ALSA: firewire-lib/firewire-tascam: localize async midi port http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119748.html A bug of this feature was reported: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119775.html And I posted a fix: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119782.html The fix brought conflict to the previous patchset and this is a revised version. Takashi Sakamoto (8): ALSA: firewire-lib/firewire-tascam: localize async midi port ALSA: firewire-tascam: remove callback function from async midi port ALSA: firewire-tascam: send fixed-length transaction for async midi port ALSA: firewire-tascam: use the same address for asynchronous transaction for MIDI message ALSA: firewire-tascam: use fixed-length array for message cache to async midi port ALSA: firewire-tascam: initialize parameters at open of rawmidi character devices ALSA: firewire-tascam: move message parameters for async midi port ALSA: firewire-tascam: support drain callback for MIDI playback substream sound/firewire/lib.c | 141 ---------------------------- sound/firewire/lib.h | 54 ----------- sound/firewire/tascam/tascam-midi.c | 13 ++- sound/firewire/tascam/tascam-transaction.c | 142 ++++++++++++++++++++++++----- sound/firewire/tascam/tascam.h | 39 +++++++- 5 files changed, 162 insertions(+), 227 deletions(-) -- 2.9.3

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[alsa-devel] usx2y: usx2yhwdeppcm.c gives "should not be here with counts=40" (linux 4.9.0)
by Jaime T 14 Apr '17

14 Apr '17

Hi all. I'm running a fresh and cleanly-installed debian testing (stretch) with my tascam us-122. After about 6 hours of perfect, joyous sound, the kernel (4.9.0-2-686) emits: should not be here with counts=40 and my jack server stops working. I'm aware that I should be running a vanilla kernel before logging a bug against the driver - is that my next step? (it might take a very long time!) (FWIW, I've run alsa-info.sh and the results are available at http://www.alsa-project.org/db/?f=f5e0f8f654ef467cedaaed7ae129b02267c46a95) Many thanks, Jaime

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[alsa-devel] [PATCH 0/8] ALSA: firewire-lib/firewire-tascam: localize async midi port
by Takashi Sakamoto 14 Apr '17

14 Apr '17

Hi, At a development period for Linux v4.4, firewire-lib module got an abstraction of asynchronous transaction for transmission of MIDI message. I did this with an assumption that this is required for Digi00x series, as well as TASCAM FireWire series. However, my recent work reveal it's actually not[0]. This patchset localizes the abstraction just for TASCAM FireWire series. [0] [alsa-devel] [PATCH 0/4] ALSA: firewire-digi00x: integration to support console models http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119397.html Takashi Sakamoto (8): firewire-lib/firewire-tascam: localize async midi port firewire-tascam: remove callback function from async midi port firewire-tascam: send fixed-length transaction for async midi port firewire-tascam: use the same address for asynchronous transaction for MIDI message firewire-tascam: use fixed-length array for message cache to async midi port firewire-tascam: initialize parameters at open of rawmidi character devices firewire-tascam: move message parameters for async midi port firewire-tascam: support drain callback for MIDI playback substream sound/firewire/lib.c | 141 ---------------------------- sound/firewire/lib.h | 54 ----------- sound/firewire/tascam/tascam-midi.c | 13 ++- sound/firewire/tascam/tascam-transaction.c | 142 ++++++++++++++++++++++++----- sound/firewire/tascam/tascam.h | 39 +++++++- 5 files changed, 162 insertions(+), 227 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] ALSA: usb-line6: constify snd_kcontrol_new strucutre array
by Takashi Sakamoto 14 Apr '17

14 Apr '17

In kernel APIs of ALSA control interface, drivers can create a control element set by a call of snd_ctl_new1() with a template. This template is known to have const qualifier in general cases. This commit adds the qualifier to template array, for safer program and runtime. Application of this change moves the symbol from .data section to .rodata section. Cc: Bhumika Goyal <bhumirks(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/usb/line6/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index fab53f5..b3854f8 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -430,7 +430,7 @@ static int snd_line6_control_playback_put(struct snd_kcontrol *kcontrol, } /* control definition */ -static struct snd_kcontrol_new line6_controls[] = { +static const struct snd_kcontrol_new line6_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", -- 2.9.3

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[alsa-devel] [PATCH] ALSA: firewire-lib: fix inappropriate assignment between signed/unsigned type
by Takashi Sakamoto 14 Apr '17

14 Apr '17

An abstraction of asynchronous transaction for transmission of MIDI messages was introduced in Linux v4.4. Each driver can utilize this abstraction to transfer MIDI messages via fixed-length payload of transaction to a certain unit address. Filling payload of the transaction is done by callback. In this callback, each driver can return negative error code, however current implementation assigns the return value to unsigned variable. This commit changes type of the variable to fix the bug. Reported-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Cc: <stable(a)vger.kernel.org> # 4.4+ Fixes: 585d7cba5e1f ("ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/lib.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index f676931..c3768cd 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -45,7 +45,7 @@ struct snd_fw_async_midi_port { struct snd_rawmidi_substream *substream; snd_fw_async_midi_port_fill fill; - unsigned int consume_bytes; + int consume_bytes; }; int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port, -- 2.9.3

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Re: [alsa-devel] [PATCH 2/8] firewire-tascam: remove callback function from async midi port (fwd)
by Julia Lawall 14 Apr '17

14 Apr '17

Hello, I haven't checked on the type, but it may be worth looking into (line 201). julia ---------- Forwarded message ---------- Date: Thu, 13 Apr 2017 18:20:17 +0800 From: kbuild test robot <fengguang.wu(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: Re: [alsa-devel] [PATCH 2/8] firewire-tascam: remove callback function from async midi port In-Reply-To: <20170413051527.21396-3-o-takashi(a)sakamocchi.jp> Hi Takashi, [auto build test WARNING on sound/for-next] [also build test WARNING on v4.11-rc6 next-20170413] [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] url: https://github.com/0day-ci/linux/commits/Takashi-Sakamoto/ALSA-firewire-lib… base: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next :::::: branch date: 3 hours ago :::::: commit date: 3 hours ago >> sound/firewire/tascam/tascam-transaction.c:201:5-24: WARNING: Unsigned expression compared with zero: port -> consume_bytes <= 0 git remote add linux-review https://github.com/0day-ci/linux git remote update linux-review git checkout b866eaa9a02893e6383eb5177c11078a16af4e80 vim +201 sound/firewire/tascam/tascam-transaction.c 741f7c77a Takashi Sakamoto 2017-04-13 185 /* Nothing to do. */ 741f7c77a Takashi Sakamoto 2017-04-13 186 if (substream == NULL || snd_rawmidi_transmit_empty(substream)) 741f7c77a Takashi Sakamoto 2017-04-13 187 return; 741f7c77a Takashi Sakamoto 2017-04-13 188 741f7c77a Takashi Sakamoto 2017-04-13 189 /* Do it in next chance. */ 741f7c77a Takashi Sakamoto 2017-04-13 190 if (ktime_after(port->next_ktime, ktime_get())) { 741f7c77a Takashi Sakamoto 2017-04-13 191 schedule_work(&port->work); 741f7c77a Takashi Sakamoto 2017-04-13 192 return; 741f7c77a Takashi Sakamoto 2017-04-13 193 } 741f7c77a Takashi Sakamoto 2017-04-13 194 741f7c77a Takashi Sakamoto 2017-04-13 195 /* 741f7c77a Takashi Sakamoto 2017-04-13 196 * Fill the buffer. The callee must use snd_rawmidi_transmit_peek(). 741f7c77a Takashi Sakamoto 2017-04-13 197 * Later, snd_rawmidi_transmit_ack() is called. 741f7c77a Takashi Sakamoto 2017-04-13 198 */ 741f7c77a Takashi Sakamoto 2017-04-13 199 memset(port->buf, 0, port->len); b866eaa9a Takashi Sakamoto 2017-04-13 200 port->consume_bytes = fill_message(substream, port->buf); 741f7c77a Takashi Sakamoto 2017-04-13 @201 if (port->consume_bytes <= 0) { 741f7c77a Takashi Sakamoto 2017-04-13 202 /* Do it in next chance, immediately. */ 741f7c77a Takashi Sakamoto 2017-04-13 203 if (port->consume_bytes == 0) { 741f7c77a Takashi Sakamoto 2017-04-13 204 port->next_ktime = 0; 741f7c77a Takashi Sakamoto 2017-04-13 205 schedule_work(&port->work); 741f7c77a Takashi Sakamoto 2017-04-13 206 } else { 741f7c77a Takashi Sakamoto 2017-04-13 207 /* Fatal error. */ 741f7c77a Takashi Sakamoto 2017-04-13 208 port->error = true; 741f7c77a Takashi Sakamoto 2017-04-13 209 } :::::: The code at line 201 was first introduced by commit :::::: 741f7c77a1a15d04ebafcce4e079cfe1cfb612ff firewire-lib/firewire-tascam: localize async midi port :::::: TO: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> :::::: CC: 0day robot <fengguang.wu(a)intel.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v4 0/2] Add STM32 SAI support
by olivier moysan 13 Apr '17

13 Apr '17

This patch-set handles the Serial Audio Interface (SAI) IP on STM32 platforms. The SAI IP is composed of two Sub-block A and B. (see the figure below) Each sub-block makes use of individual and shared resources. Shared resources: - Reset line. Reset common and sub-block registers. - Bus interface clocks. This clock is not exposed in device as it it managed by clock driver, according to kernel clocks gating. - Common register. SAI IP exhibits a common configuration register to manage synchronization modes. NB: These modes are not yet implemented in this version of the SAI driver. - Interrupt. Sub-blocks have their own interrupt status registers but they share the same interrupt line. Individual resources: - Register set - DMA request line. - Communication interface. Each sub-block has its own GPIOs and associated bus lines. - Kernel clock. Each sub-block has its own dedicated clock for its communication interface. To reflect this architecture Sub-block A and B are handled by 2 child devices. Sub-block A and B devices can be configured independently either as transmitter or receiver. A PCM device is associated to each sub-block. A sub-block has to select the appropriated parent clock at runtime, depending on the audio stream sampling rate to be played or captured. Two parent clocks must be provided to support sampling rates multiples of 8 kHz or 11.025kHz. Interface clock +--------------------------------+ +---------------> | SAI IP | Reset | +-----------------+ | +---------------> | |common registers | | | +-----------------+ | | | | +----------------------+ | | | Sub-block A | | Kernel clock A | | | | Bus A +--------------------> | +----------------+ +----------------> | | |A registers | | | | | +----------------+ | | | +----------------------+ | | | | +----------------------+ | | | Sub-block B | | Kernel clock B | | | | Bus B +--------------------> | +----------------+ +-----------------> | | |B registers | | | | | +----------------+ | | | +----------------------+ | | | +--------------------------------+ v3 -> v4 Apply minor changes from Peter review olivier moysan (2): ASoC: stm32: add bindings for SAI ASoC: stm32: add SAI driver .../devicetree/bindings/sound/st,stm32-sai.txt | 89 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 6 + sound/soc/stm/stm32_sai.c | 115 +++ sound/soc/stm/stm32_sai.h | 199 +++++ sound/soc/stm/stm32_sai_sub.c | 881 +++++++++++++++++++++ 8 files changed, 1300 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_sai.c create mode 100644 sound/soc/stm/stm32_sai.h create mode 100644 sound/soc/stm/stm32_sai_sub.c -- 1.9.1

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[alsa-devel] [PATCH] ALSA: seq: Don't break snd_use_lock_sync() loop by timeout
by Takashi Iwai 13 Apr '17

13 Apr '17

The snd_use_lock_sync() (thus its implementation snd_use_lock_sync_helper()) has the 5 seconds timeout to break out of the sync loop. It was introduced from the beginning, just to be "safer", in terms of avoiding the stupid bugs. However, as Ben Hutchings suggested, this timeout rather introduces a potential leak or use-after-free that was apparently fixed by the commit 2d7d54002e39 ("ALSA: seq: Fix race during FIFO resize"): for example, snd_seq_fifo_event_in() -> snd_seq_event_dup() -> copy_from_user() could block for a long time, and snd_use_lock_sync() goes timeout and still leaves the cell at releasing the pool. For fixing such a problem, we remove the break by the timeout while still keeping the warning. Suggested-by: Ben Hutchings <ben.hutchings(a)codethink.co.uk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_lock.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c index 3b693e924db7..12ba83367b1b 100644 --- a/sound/core/seq/seq_lock.c +++ b/sound/core/seq/seq_lock.c @@ -28,19 +28,16 @@ /* wait until all locks are released */ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line) { - int max_count = 5 * HZ; + int warn_count = 5 * HZ; if (atomic_read(lockp) < 0) { pr_warn("ALSA: seq_lock: lock trouble [counter = %d] in %s:%d\n", atomic_read(lockp), file, line); return; } while (atomic_read(lockp) > 0) { - if (max_count == 0) { - pr_warn("ALSA: seq_lock: timeout [%d left] in %s:%d\n", atomic_read(lockp), file, line); - break; - } + if (warn_count-- == 0) + pr_warn("ALSA: seq_lock: waiting [%d left] in %s:%d\n", atomic_read(lockp), file, line); schedule_timeout_uninterruptible(1); - max_count--; } } -- 2.11.1

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[alsa-devel] [PATCH v2 1/2] ALSA: hda: Move common haswell init to a helper
by Subhransu S. Prusty 13 Apr '17

13 Apr '17

From: Ander Conselvan De Oliveira <ander.conselvan.de.oliveira(a)intel.com> Geminilake vendor nid is different from other Skylake variants, but rest of the initialization code is same. So a variable is added in hdmi_spec to store the platform specific vendor nid and move the initialization code to a helper function to be used by both platform specific init. Fixes: 126cfa2f5e15 ("ALSA: hda: Add Geminilake HDMI codec ID") Signed-off-by: Ander Conselvan De Oliveira <ander.conselvan.de.oliveira(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi(a)intel.com> Cc: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 37f11560186a..90e4ff87445e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -177,6 +177,7 @@ struct hdmi_spec { bool i915_bound; /* was i915 bound in this driver? */ struct hdac_chmap chmap; + hda_nid_t vendor_nid; }; #ifdef CONFIG_SND_HDA_I915 @@ -2372,6 +2373,7 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, } #define INTEL_VENDOR_NID 0x08 +#define INTEL_GLK_VENDOR_NID 0x0B #define INTEL_GET_VENDOR_VERB 0xf81 #define INTEL_SET_VENDOR_VERB 0x781 #define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ @@ -2381,14 +2383,15 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, bool update_tree) { unsigned int vendor_param; + struct hdmi_spec *spec = codec->spec; - vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) return; vendor_param |= INTEL_EN_ALL_PIN_CVTS; - vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0, INTEL_SET_VENDOR_VERB, vendor_param); if (vendor_param == -1) return; @@ -2400,8 +2403,9 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) { unsigned int vendor_param; + struct hdmi_spec *spec = codec->spec; - vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_DP12) return; @@ -2409,7 +2413,7 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) /* enable DP1.2 mode */ vendor_param |= INTEL_EN_DP12; snd_hdac_regmap_add_vendor_verb(&codec->core, INTEL_SET_VENDOR_VERB); - snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0, + snd_hda_codec_write_cache(codec, spec->vendor_nid, 0, INTEL_SET_VENDOR_VERB, vendor_param); } @@ -2503,7 +2507,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, } /* Intel Haswell and onwards; audio component with eld notifier */ -static int patch_i915_hsw_hdmi(struct hda_codec *codec) +static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid) { struct hdmi_spec *spec; int err; @@ -2520,6 +2524,7 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec) spec = codec->spec; codec->dp_mst = true; spec->dyn_pcm_assign = true; + spec->vendor_nid = vendor_nid; intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); @@ -2548,6 +2553,16 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec) return 0; } +static int patch_i915_hsw_hdmi(struct hda_codec *codec) +{ + return intel_hsw_common_init(codec, INTEL_VENDOR_NID); +} + +static int patch_i915_glk_hdmi(struct hda_codec *codec) +{ + return intel_hsw_common_init(codec, INTEL_GLK_VENDOR_NID); +} + /* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { @@ -3800,7 +3815,7 @@ static int patch_via_hdmi(struct hda_codec *codec) HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), -HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 1.9.1

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[alsa-devel] [PATCH] topology: Allow a data section to contain multiple tuples objects
by fuweix.tang＠intel.com 13 Apr '17

13 Apr '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> It's easy to use a vendor tuples object to define a C structure instance. And a data section can include multiple vendor tuples objects. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Fuwei Tang <fuweix.tang(a)intel.com> --- include/topology.h | 14 ++++++++++ src/topology/data.c | 78 ++++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 74 insertions(+), 18 deletions(-) diff --git a/include/topology.h b/include/topology.h index 1bebe1e..ccb3a00 100644 --- a/include/topology.h +++ b/include/topology.h @@ -214,6 +214,20 @@ extern "C" { * The keyword tuples is to define vendor specific tuples. Please refer to * section Vendor Tokens and Vendor tuples. * + * It's easy to use a vendor tuples object to define a C structure instance. + * And a data section can include multiple vendor tuples objects: + * + * <pre> + * SectionData."data element name" { + * index "1" #Index number + * tuples [ + * "id of the 1st vendor tuples section" + * "id of the 2nd vendor tuples section" + * ... + * ] + * }; + * </pre> + * * <h5>How to define an element with private data</h5> * An element can refer to a single data section or multiple data * sections. diff --git a/src/topology/data.c b/src/topology/data.c index 61882b8..bd90df9 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -408,7 +408,7 @@ static unsigned int get_tuple_size(int type) } } -/* fill a data element's private buffer with its tuples */ +/* Add a tuples object to the private buffer of its parent data element */ static int copy_tuples(struct tplg_elem *elem, struct tplg_vendor_tuples *tuples, struct tplg_vendor_tokens *tokens) { @@ -423,12 +423,9 @@ static int copy_tuples(struct tplg_elem *elem, unsigned int i, j; int token_val; - if (priv) { - SNDERR("error: %s has more data than tuples\n", elem->id); - return -EINVAL; - } + size = priv ? priv->size : 0; /* original private data size */ - size = 0; + /* scan each tuples set (one set per type) */ for (i = 0; i < tuples->num_sets ; i++) { tuple_set = tuples->set[i]; set_size = sizeof(struct snd_soc_tplg_vendor_array) @@ -448,7 +445,7 @@ static int copy_tuples(struct tplg_elem *elem, return -ENOMEM; off = priv->size; - priv->size = size; + priv->size = size; /* update private data size */ array = (struct snd_soc_tplg_vendor_array *)(priv->data + off); array->size = set_size; @@ -495,6 +492,7 @@ static int build_tuples(snd_tplg_t *tplg, struct tplg_elem *elem) struct tplg_ref *ref; struct list_head *base, *pos; struct tplg_elem *tuples, *tokens; + int err; base = &elem->ref_list; list_for_each(pos, base) { @@ -504,23 +502,27 @@ static int build_tuples(snd_tplg_t *tplg, struct tplg_elem *elem) if (!ref->id || ref->type != SND_TPLG_TYPE_TUPLE) continue; - tplg_dbg("look up tuples %s\n", ref->id); + tplg_dbg("tuples '%s' used by data '%s'\n", ref->id, elem->id); if (!ref->elem) ref->elem = tplg_elem_lookup(&tplg->tuple_list, ref->id, SND_TPLG_TYPE_TUPLE, elem->index); tuples = ref->elem; - if (!tuples) + if (!tuples) { + SNDERR("error: cannot find tuples %s\n", ref->id); return -EINVAL; + } - tplg_dbg("found tuples %s\n", tuples->id); tokens = get_tokens(tplg, tuples); - if (!tokens) + if (!tokens) { + SNDERR("error: cannot find token for %s\n", ref->id); return -EINVAL; + } - tplg_dbg("found tokens %s\n", tokens->id); - /* a data object can only have one tuples object */ - return copy_tuples(elem, tuples->tuples, tokens->tokens); + /* a data object can have multiple tuples objects */ + err = copy_tuples(elem, tuples->tuples, tokens->tokens); + if (err < 0) + return err; } return 0; @@ -689,6 +691,47 @@ static int parse_tuple_sets(snd_config_t *cfg, return 0; } +/* Parse tuples references for a data element, either a single tuples section + * or a list of tuples sections. + */ +static int parse_tuples_refs(snd_config_t *cfg, + struct tplg_elem *elem) +{ + snd_config_type_t type; + snd_config_iterator_t i, next; + snd_config_t *n; + const char *val = NULL; + + type = snd_config_get_type(cfg); + + /* refer to a single tuples section */ + if (type == SND_CONFIG_TYPE_STRING) { + if (snd_config_get_string(cfg, &val) < 0) + return -EINVAL; + tplg_dbg("\ttuples: %s\n", val); + return tplg_ref_add(elem, SND_TPLG_TYPE_TUPLE, val); + } + + if (type != SND_CONFIG_TYPE_COMPOUND) { + SNDERR("error: compound type expected for %s", elem->id); + return -EINVAL; + } + + /* refer to a list of data sections */ + snd_config_for_each(i, next, cfg) { + const char *val; + + n = snd_config_iterator_entry(i); + if (snd_config_get_string(n, &val) < 0) + continue; + + tplg_dbg("\ttuples: %s\n", val); + tplg_ref_add(elem, SND_TPLG_TYPE_TUPLE, val); + } + + return 0; +} + /* Parse private data references for the element, either a single data section * or a list of data sections. */ @@ -1002,10 +1045,9 @@ int tplg_parse_data(snd_tplg_t *tplg, snd_config_t *cfg, } if (strcmp(id, "tuples") == 0) { - if (snd_config_get_string(n, &val) < 0) - return -EINVAL; - tplg_dbg(" Data: %s\n", val); - tplg_ref_add(elem, SND_TPLG_TYPE_TUPLE, val); + err = parse_tuples_refs(n, elem); + if (err < 0) + return err; continue; } -- 2.7.4

1 0

[alsa-devel] [PATCH] topology: Added supply widget support
by fuweix.tang＠intel.com 13 Apr '17

13 Apr '17

From: "Tewani, Pradeep D" <pradeep.d.tewani(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Fuwei Tang <fuweix.tang(a)intel.com> --- include/sound/asoc.h | 3 ++- src/topology/dapm.c | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/asoc.h b/include/sound/asoc.h index 082c542..9bef6eb 100644 --- a/include/sound/asoc.h +++ b/include/sound/asoc.h @@ -70,7 +70,8 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_SUPPLY 16 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_SUPPLY /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 13aa1c4..b7e244e 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -38,6 +38,7 @@ static const struct map_elem widget_map[] = { {"dai_in", SND_SOC_TPLG_DAPM_DAI_IN}, {"dai_out", SND_SOC_TPLG_DAPM_DAI_OUT}, {"dai_link", SND_SOC_TPLG_DAPM_DAI_LINK}, + {"supply", SND_SOC_TPLG_DAPM_SUPPLY}, }; static int lookup_widget(const char *w) -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: line6: constify snd_kcontrol_new structures
by Bhumika Goyal 13 Apr '17

13 Apr '17

Declare snd_kcontrol_new strcutures as const as they are only passed as an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const too. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/usb/line6/pod.c | 2 +- sound/usb/line6/toneport.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index 17aa616..358224c 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -380,7 +380,7 @@ static int snd_pod_control_monitor_put(struct snd_kcontrol *kcontrol, } /* control definition */ -static struct snd_kcontrol_new pod_control_monitor = { +static const struct snd_kcontrol_new pod_control_monitor = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitor Playback Volume", .index = 0, diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index 8e22f43..ba7975c 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -250,7 +250,7 @@ static void toneport_start_pcm(unsigned long arg) } /* control definition */ -static struct snd_kcontrol_new toneport_control_monitor = { +static const struct snd_kcontrol_new toneport_control_monitor = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitor Playback Volume", .index = 0, @@ -261,7 +261,7 @@ static struct snd_kcontrol_new toneport_control_monitor = { }; /* source selector definition */ -static struct snd_kcontrol_new toneport_control_source = { +static const struct snd_kcontrol_new toneport_control_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Capture Source", .index = 0, -- 2.7.4

3 3

[alsa-devel] [PATCH v2] ASoC: fsl ssi doc: Move optional properties to the correct section
by Fabio Estevam 12 Apr '17

12 Apr '17

Properties 'fsl,ssi-asynchronous', 'fsl,playback-dma' and 'fsl,capture-dma' are optional, so move them under the optional section of the document. While at it mention that 'fsl,playback-dma' and 'fsl,capture-dma' only apply to Power Architecture. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Combine the two previous changes into a single patch - Fix identation .../devicetree/bindings/sound/fsl,ssi.txt | 34 ++++++++++++---------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index 5b76be4..d415888 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -20,24 +20,8 @@ Required properties: have. - interrupt-parent: The phandle for the interrupt controller that services interrupts for this device. -- fsl,playback-dma: Phandle to a node for the DMA channel to use for - playback of audio. This is typically dictated by SOC - design. See the notes below. -- fsl,capture-dma: Phandle to a node for the DMA channel to use for - capture (recording) of audio. This is typically dictated - by SOC design. See the notes below. - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. This number is the maximum allowed value for SFCSR[TFWM0]. -- fsl,ssi-asynchronous: - If specified, the SSI is to be programmed in asynchronous - mode. In this mode, pins SRCK, STCK, SRFS, and STFS must - all be connected to valid signals. In synchronous mode, - SRCK and SRFS are ignored. Asynchronous mode allows - playback and capture to use different sample sizes and - sample rates. Some drivers may require that SRCK and STCK - be connected together, and SRFS and STFS be connected - together. This would still allow different sample sizes, - but not different sample rates. - clocks: "ipg" - Required clock for the SSI unit "baud" - Required clock for SSI master mode. Otherwise this clock is not used @@ -61,6 +45,24 @@ Optional properties: - fsl,mode: The operating mode for the AC97 interface only. "ac97-slave" - AC97 mode, SSI is clock slave "ac97-master" - AC97 mode, SSI is clock master +- fsl,ssi-asynchronous: + If specified, the SSI is to be programmed in asynchronous + mode. In this mode, pins SRCK, STCK, SRFS, and STFS must + all be connected to valid signals. In synchronous mode, + SRCK and SRFS are ignored. Asynchronous mode allows + playback and capture to use different sample sizes and + sample rates. Some drivers may require that SRCK and STCK + be connected together, and SRFS and STFS be connected + together. This would still allow different sample sizes, + but not different sample rates. +- fsl,playback-dma: Phandle to a node for the DMA channel to use for + playback of audio. This is typically dictated by SOC + design. See the notes below. + Only used on Power Architecture. +- fsl,capture-dma: Phandle to a node for the DMA channel to use for + capture (recording) of audio. This is typically dictated + by SOC design. See the notes below. + Only used on Power Architecture. Child 'codec' node required properties: - compatible: Compatible list, contains the name of the codec -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: fsl_esai: Remove unneeded definition
by Fabio Estevam 12 Apr '17

12 Apr '17

There is no need for defining FSL_ESAI_RATES locally as the standard SNDRV_PCM_RATE_8000_192000 definition can be used instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 38bfd46..809a069 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -19,7 +19,6 @@ #include "fsl_esai.h" #include "imx-pcm.h" -#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 #define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ @@ -647,14 +646,14 @@ static struct snd_soc_dai_driver fsl_esai_dai = { .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 12, - .rates = FSL_ESAI_RATES, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = FSL_ESAI_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 8, - .rates = FSL_ESAI_RATES, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = FSL_ESAI_FORMATS, }, .ops = &fsl_esai_dai_ops, -- 2.7.4

2 1

[alsa-devel] [PATCH 00/11] ASoC: Intel: Skylake: More driver updates
by jeeja.kp＠intel.com 12 Apr '17

12 Apr '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides updates on the following: o Fix LLCH register read. o Add 16-bit constraint to FE bxt_rt298 machine. o Fix DMA position reporting for capture stream. o Use the sig_bits to define dai bps caps. o Update sig_bits based on converter capability. o Rearrangement of code to cleanup SKL SST library. o Add support for deferred DSP module bind. B, Jayachandran (1): ALSA: hda: Fix LLCH register read G Kranthi (1): ASoC: Intel: Skylake: Add 16-bit constraint to FE bxt_rt298 machine Hardik T Shah (1): ASoC: Intel: Skylake: Fix DMA position reporting for capture stream Jeeja KP (5): ASoC: Intel: Skylake: Use the sig_bits to define dai bps capability ASoC: hdac_hdmi: Update sig_bits based on converter capability ASoC: Intel: Skylake: Rearrangement of code to cleanup SKL SST library ASoC: Intel: Skylake: Fix module state after unbind and delete ASoC: Intel: Skylake: Add support for deferred DSP module bind Vinod Koul (3): ASoC: Intel: Skylake: Don't unload module when in use ASoC: Intel: Skylake: Remove redundant vmixer handler ASoC: Intel: Skylake: remove hard coded ACPI path sound/hda/hdac_controller.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 3 + sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-nhlt.c | 7 +- sound/soc/intel/skylake/skl-pcm.c | 94 ++++++++++++++++--- sound/soc/intel/skylake/skl-sst-dsp.h | 24 +++-- sound/soc/intel/skylake/skl-sst-ipc.h | 8 ++ sound/soc/intel/skylake/skl-sst-utils.c | 60 ++---------- sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 159 +++++++++++++++++++++++++------- sound/soc/intel/skylake/skl-topology.h | 17 ++-- sound/soc/intel/skylake/skl.h | 1 + 13 files changed, 262 insertions(+), 124 deletions(-) -- 2.5.0

5 25

[alsa-devel] [PATCH v2 0/3] ucm-conf: Add intel UCM conf files to alsa-ucm-conf repo
by fuweix.tang＠intel.com 12 Apr '17

12 Apr '17

From: Fuwei Tang <fuweix.tang(a)intel.com> Add the intel UCM configs to a dedicated UCM conf repo and release them using the BSD license. The other non intel UCM files can be moved over when there is agreement with the file authors, but in the mean time they will stay in alsa-lib. The configs are moved from alsa-lib repo. The original authorship and commit message of all config files will be reserved. Fuwei Tang (1): ucm-conf: Set up for installation Vinod Koul (2): ucm-conf: broxton: add broxton-rt298 conf files ucm-conf: skylake: add skylake-rt286 conf files .gitignore | 12 ++++ ChangeLog | 1 + INSTALL | 1 + Makefile.am | 1 + broxton-rt298/Hdmi1 | 22 +++++++ broxton-rt298/Hdmi2 | 14 +++++ broxton-rt298/HiFi | 129 +++++++++++++++++++++++++++++++++++++++ broxton-rt298/Makefile.am | 4 ++ broxton-rt298/broxton-rt298.conf | 58 ++++++++++++++++++ configure.ac | 24 ++++++++ gitcompile | 5 ++ skylake-rt286/Hdmi1 | 20 ++++++ skylake-rt286/Hdmi2 | 20 ++++++ skylake-rt286/HiFi | 129 +++++++++++++++++++++++++++++++++++++++ skylake-rt286/Makefile.am | 4 ++ skylake-rt286/skylake-rt286.conf | 60 ++++++++++++++++++ 16 files changed, 504 insertions(+) create mode 100644 .gitignore create mode 100644 ChangeLog create mode 120000 INSTALL create mode 100644 Makefile.am create mode 100644 broxton-rt298/Hdmi1 create mode 100644 broxton-rt298/Hdmi2 create mode 100644 broxton-rt298/HiFi create mode 100644 broxton-rt298/Makefile.am create mode 100644 broxton-rt298/broxton-rt298.conf create mode 100644 configure.ac create mode 100755 gitcompile create mode 100644 skylake-rt286/Hdmi1 create mode 100644 skylake-rt286/Hdmi2 create mode 100644 skylake-rt286/HiFi create mode 100644 skylake-rt286/Makefile.am create mode 100644 skylake-rt286/skylake-rt286.conf -- 2.7.4

4 7

[alsa-devel] [PATCH] ALSA: usb-audio: constify snd_kcontrol_new structures
by Bhumika Goyal 12 Apr '17

12 Apr '17

Declare snd_kcontrol_new strcutures as const as they are only passed as an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const too. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/usb/midi.c | 2 +- sound/usb/mixer.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 6e763bc..a35f414 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1922,7 +1922,7 @@ static int roland_load_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new roland_load_ctl = { +static const struct snd_kcontrol_new roland_load_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "MIDI Input Mode", .info = roland_load_info, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4703cae..082736c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1172,7 +1172,7 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = { }; /* the read-only variant */ -static struct snd_kcontrol_new usb_feature_unit_ctl_ro = { +static const struct snd_kcontrol_new usb_feature_unit_ctl_ro = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later manually */ .info = mixer_ctl_feature_info, @@ -1745,7 +1745,7 @@ static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, } /* alsa control interface for processing/extension unit */ -static struct snd_kcontrol_new mixer_procunit_ctl = { +static const struct snd_kcontrol_new mixer_procunit_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later */ .info = mixer_ctl_feature_info, @@ -2033,7 +2033,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, } /* alsa control interface for selector unit */ -static struct snd_kcontrol_new mixer_selectunit_ctl = { +static const struct snd_kcontrol_new mixer_selectunit_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later */ .info = mixer_ctl_selector_info, -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: cache maximum length of payload to reduce function calls
by Takashi Sakamoto 12 Apr '17

12 Apr '17

During packet streaming, maximum length of payload for isochronous packet is invariable, therefore no need to recalculate. Current ALSA IEC 61883-1/6 engine calls a function to calculate it 8,000 or more times per second for incoming packet processing. This commit adds a member to have maximum length of payload into 'struct amdtp_stream', to reduces the function calls. At first callback from isochronous context, the length is calculated and stored for later processing. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 9 +++++---- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f6af8e6..9e6f54f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -412,8 +412,7 @@ static inline int queue_out_packet(struct amdtp_stream *s, static inline int queue_in_packet(struct amdtp_stream *s) { - return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s)); + return queue_packet(s, IN_PACKET_HEADER_SIZE, s->max_payload_length); } static int handle_out_packet(struct amdtp_stream *s, @@ -713,12 +712,12 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, cycle = decrement_cycle_count(cycle, packets); /* For buffer-over-run prevention. */ - max_payload_length = amdtp_stream_get_max_payload(s); + max_payload_length = s->max_payload_length; for (i = 0; i < packets; i++) { cycle = increment_cycle_count(cycle, 1); - /* The number of quadlets in this packet */ + /* The number of bytes in this packet */ payload_length = (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT); if (payload_length > max_payload_length) { @@ -751,6 +750,8 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, u32 cycle; unsigned int packets; + s->max_payload_length = amdtp_stream_get_max_payload(s); + /* * For in-stream, first packet has come. * For out-stream, prepared to transmit first packet diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 2bd4de4..7e88317 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -110,6 +110,7 @@ struct amdtp_stream { int (*handle_packet)(struct amdtp_stream *s, unsigned int payload_quadlets, unsigned int cycle, unsigned int index); + unsigned int max_payload_length; /* For CIP headers. */ unsigned int source_node_id_field; -- 2.9.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl ssi doc: Move 'fsl, ssi-asynchronous' to the optional section
by Fabio Estevam 12 Apr '17

12 Apr '17

Property 'fsl,ssi-asynchronous' is optional, so move it to the optional section of the document. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,ssi.txt | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index 5b76be4..1d292d7 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -28,16 +28,6 @@ Required properties: by SOC design. See the notes below. - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. This number is the maximum allowed value for SFCSR[TFWM0]. -- fsl,ssi-asynchronous: - If specified, the SSI is to be programmed in asynchronous - mode. In this mode, pins SRCK, STCK, SRFS, and STFS must - all be connected to valid signals. In synchronous mode, - SRCK and SRFS are ignored. Asynchronous mode allows - playback and capture to use different sample sizes and - sample rates. Some drivers may require that SRCK and STCK - be connected together, and SRFS and STFS be connected - together. This would still allow different sample sizes, - but not different sample rates. - clocks: "ipg" - Required clock for the SSI unit "baud" - Required clock for SSI master mode. Otherwise this clock is not used @@ -61,6 +51,16 @@ Optional properties: - fsl,mode: The operating mode for the AC97 interface only. "ac97-slave" - AC97 mode, SSI is clock slave "ac97-master" - AC97 mode, SSI is clock master +- fsl,ssi-asynchronous: + If specified, the SSI is to be programmed in asynchronous + mode. In this mode, pins SRCK, STCK, SRFS, and STFS must + all be connected to valid signals. In synchronous mode, + SRCK and SRFS are ignored. Asynchronous mode allows + playback and capture to use different sample sizes and + sample rates. Some drivers may require that SRCK and STCK + be connected together, and SRFS and STFS be connected + together. This would still allow different sample sizes, + but not different sample rates. Child 'codec' node required properties: - compatible: Compatible list, contains the name of the codec -- 2.7.4

2 2

[alsa-devel] [PATCH 1/3] ALSA: hda: Move common haswell init to a helper
by Subhransu S. Prusty 12 Apr '17

12 Apr '17

Geminilake vendor nid is different from other Skylake variants, but rest of the initialization code is same. So a variable is added in hdmi_spec to store the platform specific vendor nid and move the initialization code to a helper function to be used by both platform specific init. Fixes: 126cfa2f5e15 ("ALSA: hda: Add Geminilake HDMI codec ID") Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi(a)intel.com> Cc: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 48 ++++++++++++++++++++++++++++++---------------- 1 file changed, 31 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 37f11560186a..7413fff06d70 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -177,6 +177,7 @@ struct hdmi_spec { bool i915_bound; /* was i915 bound in this driver? */ struct hdac_chmap chmap; + hda_nid_t vendor_nid; }; #ifdef CONFIG_SND_HDA_I915 @@ -2381,14 +2382,15 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, bool update_tree) { unsigned int vendor_param; + struct hdmi_spec *spec = codec->spec; - vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) return; vendor_param |= INTEL_EN_ALL_PIN_CVTS; - vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0, INTEL_SET_VENDOR_VERB, vendor_param); if (vendor_param == -1) return; @@ -2400,8 +2402,9 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) { unsigned int vendor_param; + struct hdmi_spec *spec = codec->spec; - vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_DP12) return; @@ -2409,7 +2412,7 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) /* enable DP1.2 mode */ vendor_param |= INTEL_EN_DP12; snd_hdac_regmap_add_vendor_verb(&codec->core, INTEL_SET_VENDOR_VERB); - snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0, + snd_hda_codec_write_cache(codec, spec->vendor_nid, 0, INTEL_SET_VENDOR_VERB, vendor_param); } @@ -2502,22 +2505,11 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, } } -/* Intel Haswell and onwards; audio component with eld notifier */ -static int patch_i915_hsw_hdmi(struct hda_codec *codec) +static int intel_haswell_common_init(struct hda_codec *codec) { - struct hdmi_spec *spec; + struct hdmi_spec *spec = codec->spec; int err; - /* HSW+ requires i915 binding */ - if (!codec->bus->core.audio_component) { - codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); - return -ENODEV; - } - - err = alloc_generic_hdmi(codec); - if (err < 0) - return err; - spec = codec->spec; codec->dp_mst = true; spec->dyn_pcm_assign = true; @@ -2548,6 +2540,28 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec) return 0; } +/* Intel Haswell and onwards; audio component with eld notifier */ +static int patch_i915_hsw_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* HSW+ requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; + } + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + + spec = codec->spec; + spec->vendor_nid = INTEL_VENDOR_NID; + + return intel_haswell_common_init(codec); +} + /* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { -- 1.9.1

2 3

[alsa-devel] [PATCH v5 0/2] ASoC: Add support for Conexant CX2092X DSP
by simon.ho.cnxt＠gmail.com 12 Apr '17

12 Apr '17

From: Simon Ho <simon.ho(a)conexant.com> Thanks to Charles' review, this fixed things were pointed by him. Changes with v4: - Changed lable name to lower case. - Uses non-blocking sleep function msleep() instead of mdelay(). - Fixed incorrect return code when sendcmd operation fails. Simon Ho (2): doc: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add support for Conexant CX2092X DSP .../devicetree/bindings/sound/cnxt,cx2092x.txt | 25 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2092x.c | 440 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 19 + 5 files changed, 492 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cnxt,cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

4 6

[alsa-devel] Applied "ASoC: hisilicon: Add error returns even for cases that shouldn't happen." to the asoc tree
by Mark Brown 11 Apr '17

11 Apr '17

The patch ASoC: hisilicon: Add error returns even for cases that shouldn't happen. 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 da13d7462b013ab58129fe20bfb3acb3aa73e07e Mon Sep 17 00:00:00 2001 From: John Stultz <john.stultz(a)linaro.org> Date: Tue, 11 Apr 2017 12:15:16 -0700 Subject: [PATCH] ASoC: hisilicon: Add error returns even for cases that shouldn't happen. This patch addresses feedback from Mark Brown, adding a few extra error returns in cases that shouldn't happen Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/hisilicon/hi6210-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 45691b70060c..1909bfc6950a 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -395,6 +395,7 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, break; default: WARN_ONCE(1, "Invalid i2s->fmt MASTER_MASK. This shouldn't happen\n"); + return -EINVAL; } switch (i2s->format & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -409,6 +410,7 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, break; default: WARN_ONCE(1, "Invalid i2s->fmt FORMAT_MASK. This shouldn't happen\n"); + return -EINVAL; } val = hi6210_read_reg(i2s, HII2S_I2S_CFG); -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: hisilicon: Address style nit to use break in final default of switch statement" to the asoc tree
by Mark Brown 11 Apr '17

11 Apr '17

The patch ASoC: hisilicon: Address style nit to use break in final default of switch statement 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 16c1c089b23aea681600d504102db950c1bd6ee4 Mon Sep 17 00:00:00 2001 From: John Stultz <john.stultz(a)linaro.org> Date: Tue, 11 Apr 2017 12:15:17 -0700 Subject: [PATCH] ASoC: hisilicon: Address style nit to use break in final default of switch statement Mark Brown suggested a style change to use break in the final default of a switch statement, so this patch addresses that. Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/hisilicon/hi6210-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 1909bfc6950a..27553dca7cc6 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -324,6 +324,7 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, default: i2s->bits = 16; dma_data->addr_width = 2; + break; } i2s->rate = params_rate(params); i2s->channels = params_channels(params); @@ -442,6 +443,7 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, val = hi6210_read_reg(i2s, HII2S_I2S_CFG); val &= ~HII2S_I2S_CFG__S2_FRAME_MODE; hi6210_write_reg(i2s, HII2S_I2S_CFG, val); + break; } /* clear loopback, set signed type and word length */ -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: hisilicon: Use devm_snd_soc_register_component" to the asoc tree
by Mark Brown 11 Apr '17

11 Apr '17

The patch ASoC: hisilicon: Use devm_snd_soc_register_component 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 e6d56d21bdcfd8968c1859bc1892608acad3e6d8 Mon Sep 17 00:00:00 2001 From: John Stultz <john.stultz(a)linaro.org> Date: Tue, 11 Apr 2017 12:15:18 -0700 Subject: [PATCH] ASoC: hisilicon: Use devm_snd_soc_register_component Per feedback from Mark Brown, this patch updates the hi6210-i2s driver to use devm_snd_soc_register_component which simplifies the logic a bit. Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/hisilicon/hi6210-i2s.c | 18 ++---------------- 1 file changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 27553dca7cc6..45163e5202f5 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -591,22 +591,9 @@ static int hi6210_i2s_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &hi6210_i2s_i2s_comp, + ret = devm_snd_soc_register_component(&pdev->dev, &hi6210_i2s_i2s_comp, &i2s->dai, 1); - if (ret) { - dev_err(&pdev->dev, "Failed to register dai\n"); - return ret; - } - - return 0; -} - -static int hi6210_i2s_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - return 0; + return ret; } static const struct of_device_id hi6210_i2s_dt_ids[] = { @@ -618,7 +605,6 @@ MODULE_DEVICE_TABLE(of, hi6210_i2s_dt_ids); static struct platform_driver hi6210_i2s_driver = { .probe = hi6210_i2s_probe, - .remove = hi6210_i2s_remove, .driver = { .name = "hi6210_i2s", .of_match_table = hi6210_i2s_dt_ids, -- 2.11.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: tas2552: Describe the possible I2C addresses
by Fabio Estevam 11 Apr '17

11 Apr '17

According to the TAS2552 datasheet the possible I2C addresses are: 0x40 - when ADDR pin is 0 0x41 - when ADDR pin is 1 List the possible values for better clarification. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/tas2552.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt index 54c1c87..ea8abf0 100644 --- a/Documentation/devicetree/bindings/sound/tas2552.txt +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -5,7 +5,8 @@ The tas2552 serial control bus communicates through I2C protocols Required properties: - compatible - One of: "ti,tas2552" - TAS2552 - - reg - I2C slave address + - reg - I2C slave address: it can be 0x40 if ADDR pin is 0 + or 0x41 if ADDR pin is 1. - supply-*: Required supply regulators are: "vbat" battery voltage "iovdd" I/O Voltage -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: rt5665: move rt5665_set_jack_detect to .set_jack
by Bard Liao 11 Apr '17

11 Apr '17

Now, we can use .set_jack callback function on codec level. So we don't need export rt5665_set_jack_detect. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 6 +++--- sound/soc/codecs/rt5665.h | 2 -- 2 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 42bb5fd..8cd2230 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1260,8 +1260,8 @@ static void rt5665_jd_check_handler(struct work_struct *work) } } -int rt5665_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hs_jack) +static int rt5665_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack, void *data) { struct rt5665_priv *rt5665 = snd_soc_codec_get_drvdata(codec); @@ -1288,7 +1288,6 @@ int rt5665_set_jack_detect(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(rt5665_set_jack_detect); static void rt5665_jack_detect_handler(struct work_struct *work) { @@ -4567,6 +4566,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5665 = { }, .set_sysclk = rt5665_set_codec_sysclk, .set_pll = rt5665_set_codec_pll, + .set_jack = rt5665_set_jack_detect, }; diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index a30f5e6..1db5c6a 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1984,7 +1984,5 @@ enum { int rt5665_sel_asrc_clk_src(struct snd_soc_codec *codec, unsigned int filter_mask, unsigned int clk_src); -int rt5665_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hs_jack); #endif /* __RT5665_H__ */ -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: Improve hi6210-i2s DT bindings" to the asoc tree
by Mark Brown 11 Apr '17

11 Apr '17

The patch ASoC: Improve hi6210-i2s DT bindings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 86666c083a4eacb798754f2485bbecfedc9fc7d7 Mon Sep 17 00:00:00 2001 From: John Stultz <john.stultz(a)linaro.org> Date: Mon, 10 Apr 2017 12:35:11 -0700 Subject: [PATCH] ASoC: Improve hi6210-i2s DT bindings This patch improves the previously submitted hi6210-i2s DT binding, adding extra details to how the multi-dai index value maps to the potential interfaces. (Currently just index 0 -> the S2 interface, as there is only one supported, but in the future other interfaces may be enabled.) Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt index 680bb03577ef..7a296784eb37 100644 --- a/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt +++ b/Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt @@ -17,6 +17,10 @@ Required properties: - dma-names: should be "tx" and "rx" - hisilicon,sysctrl-syscon: phandle to sysctrl syscon - #sound-dai-cells: Should be set to 1 (for multi-dai) + - The dai cell indexes reference the following interfaces: + 0: S2 interface + (Currently that is the only one available, but more may be + supported in the future) Example for the hi6210 i2s controller: @@ -32,3 +36,7 @@ i2s0: i2s@f7118000{ hisilicon,sysctrl-syscon = <&sys_ctrl>; #sound-dai-cells = <1>; }; + +Then when referencing the i2s controller: + sound-dai = <&i2s0 0>; /* index 0 => S2 interface */ + -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: topology: Fix to store enum text values
by Subhransu S. Prusty 11 Apr '17

11 Apr '17

From: Mousumi Jana <mousumix.jana(a)intel.com> Add missing enum texts store in soc_enum. Signed-off-by: Mousumi Jana <mousumix.jana(a)intel.com> Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada(a)intel.com> Signed-off-by: Kranthikumar, GudishaX <gudishax.kranthikumar(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3e9b1c0bb1ce..058bc99c6c34 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -933,6 +933,7 @@ static int soc_tplg_denum_create_texts(struct soc_enum *se, } } + se->texts = (const char * const *)se->dobj.control.dtexts; return 0; err: -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tas2552: Return the real error code
by Fabio Estevam 11 Apr '17

11 Apr '17

In the case of error in tas2552_codec_probe() we should better propagate the real error code instead of always returning '-EIO'. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index baf455e..fd5251e 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -611,7 +611,7 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), tas2552->supplies); - return -EIO; + return ret; } static int tas2552_codec_remove(struct snd_soc_codec *codec) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: firewire-lib/firewire-motu: add tracepoints for IEC 61883-1/6 variants
by Takashi Sakamoto 11 Apr '17

11 Apr '17

Hi, ALSA IEC 61883-1/6 engine already supports some tracepoints to probe events for a part of parameters about incoming/outgoing packets. The main purpose of the tracepoints to assists users/developers for further work to the engine. For example, users can gather the parameters and developers use it to investigate quirks of actual units. Actually, as I reported [0], some drivers in ALSA firewire stack seems to have timestamping issue which have never resolved by any Free Software [1], and further investigation is required for users/developers to improves the engine. My recent works add support for some variants of IEC 61883-1/6. Existent tracepoints are unavailable for a part of the variants. Furthermore, in a part of the variants, presentation timestamp is transferred by a different way against IEC 61883-1/6. It's better to trace packet parameters for the variants, as well as for compliant protocols to IEC 61883-1/6. This patchset adds some tracepoints for the purpose, mainly for RME Fireface series and MOTU FireWire series. Below code/command is a sample to work with perf(1) for tracepoints on ALSA driver for MOTU FireWire series, added at third patch. The final command outputs contents of data chunks for message from units. Please install or build perf command with python 2 support, in advance. $ cat ./motu.py import os import sys sys.path.append(os.environ['PERF_EXEC_PATH'] + \ '/scripts/python/Perf-Trace-Util/lib/Perf/Trace') from perf_trace_context import * from Core import * # For a case that several MOTU FireWire units on the same bus. cycles = {} def snd_firewire_lib__in_packet(name, context, cpu, secs, nsecs, pid, comm, callchain, second, cycle, channel, src, dest, cip_header0, cip_header1, payload_quadlets, packet_index, irq, index): global cycles if src not in cycles: cycles[src] = {} cycles[src]['sec'] = second cycles[src]['cycle'] = cycle def snd_firewire_motu__in_data_block_message(name, context, cpu, secs, nsecs, pid, comm, callchain, src, dst, data_blocks, messages): global cycles if src not in cycles: return print('{0} sec {1} cycle'.format(cycles[src]['sec'], cycles[src]['cycle'])) for i in range(data_blocks): offset = i * 8 msg = messages[offset:offset + 8] print(' {0}: {1[5]:02x}{1[4]:02x}{1[3]:02x}{1[2]:02x}{1[1]:02x}{1[0]:02x}'.format(i, msg)) $ ~/linux.git/tools/perf/perf record -a -e snd_firewire_motu:in_data_block_message -e snd_firewire_lib:in_packet ^C $ PERF_EXEC_PATH=~/linux.git/tools/perf/ ~/linux.git/tools/perf/perf script -s ./motu.py ... 7 sec 7999 cycle 0: 000000805854 1: 0000008050fc 2: 0000808050a4 3: 00003f80504c 4: 0000008048f4 5: 00000080489c 6: 000080804844 7: 00003f8040ec 0 sec 0 cycle 0 sec 1 cycle 0: 000000804094 1: 0000008050fc 2: 0000808050a4 3: 00003f80504c 4: 0000008048f4 5: 00000080489c 6: 000080804844 7: 00003f8058ac 0 sec 2 cycle 0: 000000805854 1: 0000008050fc 2: 0000808050a4 3: 00003f80504c 4: 0000008048f4 5: 00000080605c 6: 000080806004 7: 00003f8058ac ... [0] [alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6 http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html [1] This is not resolved in FFADO, user space library to drive the units, too. Some workarounds are applied to the library, but it can not work well depending on units and protocols. Takashi Sakamoto (3): ALSA: firewire_lib: add tracepoints for packets without CIP headers ALSA: firewire-motu: add tracepoints for SPH in IEC 61883-1 fashion ALSA: firewire-motu: add tracepoints for messages for unique protocol sound/firewire/amdtp-stream-trace.h | 88 +++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 8 +++ sound/firewire/motu/Makefile | 2 + sound/firewire/motu/amdtp-motu-trace.h | 123 +++++++++++++++++++++++++++++++++ sound/firewire/motu/amdtp-motu.c | 37 ++++++++++ 5 files changed, 258 insertions(+) create mode 100644 sound/firewire/motu/amdtp-motu-trace.h -- 2.9.3

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[alsa-devel] [PATCH] ASoC: jack: add snd_soc_codec_set_jack
by Bard Liao 10 Apr '17

10 Apr '17

There are many codecs with the capability of jack detection. Usually, we create a jack on machine driver but there is no common function for machine driver to deliver the jack pointer to codec driver. snd_soc_codec_set_jack can be used for delivering the jack pointer to codec driver and enable the jack detection function. To make it work, codec driver need to define a callback function to receive the jack pointer and do all necessary procedure for enabling jack detection. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 4 ++++ sound/soc/soc-jack.c | 18 ++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 77ef074..864597d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -434,6 +434,8 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_codec_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, void *data); int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); @@ -913,6 +915,8 @@ struct snd_soc_codec_driver { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, void *data); /* codec IO */ struct regmap *(*get_regmap)(struct device *); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index a03dcbb9..7daf21f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -23,6 +23,24 @@ #include <trace/events/asoc.h> /** + * snd_soc_codec_set_jack - configure codec jack. + * @codec: CODEC + * @jack: structure to use for the jack + * @data: can be used if codec driver need extra data for configuring jack + * + * Configures and enables jack detection function. + */ +int snd_soc_codec_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, void *data) +{ + if (codec->driver->set_jack) + return codec->driver->set_jack(codec, jack, data); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); + +/** * snd_soc_card_jack_new - Create a new jack * @card: ASoC card * @id: an identifying string for this jack -- 2.7.4

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