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[alsa-devel] [PATCH] ASoC: meson: fix do_div warning in spdifin
by Jerome Brunet 13 Dec '18

13 Dec '18

Even if this spdif input driver is only supposed to be used on 64bits platform, there is possible problem with 32bits and do_div, as reported by the kbuild robot. Just fix it. Fixes: 5ce5658375e6 ("ASoC: meson: add axg spdif input") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi Mark, This fixes the problem reported by kbuild robot [0]. Sorry for not catching this earlier. Regards Jerome [0]: https://lkml.kernel.org/r/201812132245.lBYydcs3%fengguang.wu@intel.com sound/soc/meson/axg-spdifin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index 09f385a2bbba..01b2035fa841 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -146,8 +146,8 @@ static void axg_spdifin_write_mode_param(struct regmap *map, int mode, unsigned int base_reg, unsigned int width) { - unsigned int offset = mode, rem; - unsigned int reg, shift; + uint64_t offset = mode; + unsigned int reg, shift, rem; rem = do_div(offset, num_per_reg); -- 2.19.2

2 1

[alsa-devel] [PATCH] ASoC: core: Invoke pcm_new() for all DAI-link
by Rohit kumar 13 Dec '18

13 Dec '18

Remove no_pcm check to invoke pcm_new() for backend dai-links too. This fixes crash in hdmi codec driver during hdmi_codec_startup() while accessing chmap_info struct. chmap_info struct memory is allocated in pcm_new() of hdmi codec driver which is not invoked in case of DPCM when hdmi codec driver is part of backend dai-link. Below is the crash stack: [ 61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018 .. [ 61.666696] CM = 0, WnR = 1 [ 61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000 [ 61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000 [ 61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP [ 61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21 .. [ 61.740269] PC is at hdmi_codec_startup+0x124/0x164 [ 61.745308] LR is at hdmi_codec_startup+0xe4/0x164 Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12..abdc460 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1467,7 +1467,7 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, for (i = 0; i < num_dais; ++i) { struct snd_soc_dai_driver *drv = dais[i]->driver; - if (!rtd->dai_link->no_pcm && drv->pcm_new) + if (drv->pcm_new) ret = drv->pcm_new(rtd, dais[i]); if (ret < 0) { dev_err(dais[i]->dev, -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH 0/2] Graph fixes for using multiple endpoints per port
by Tony Lindgren 13 Dec '18

13 Dec '18

Hi all, Here are two fixes that allow me to have multiple endpoints defined in the dts file for audio-graph-card. To do that, we need to fix up few issues as the graph binding Documentation/devicetree/bindings/graph.txt allows multiple endpoints per port. This allows configuring TDM codecs for I2S for example. Kuninori-san, can you please check if this makes sense to you and compare against the graph binding? Cheers, Tony Tony Lindgren (2): ASoC: simple-card-utils: revert port changes to follow graph binding ASoC: audio-graph-card: Fix parsing of multiple endpoints sound/soc/generic/audio-graph-card.c | 41 +++++++++++++++++++++++---- sound/soc/generic/simple-card-utils.c | 22 ++++++++++++-- 2 files changed, 55 insertions(+), 8 deletions(-) -- 2.19.2

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[alsa-devel] [PATCH 0/7 v2] ASoC: merge audio-graph-scu into audio-graph - final step
by Kuninori Morimoto 13 Dec '18

13 Dec '18

Hi Mark, Tony audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is v2 of final step patches. Tony, I think 1) and 3) patch will solve your issue. Can you please test these ? Kuninori Morimoto (7): 1) ASoC: simple-card-utils: care endpoint reg for asoc_simple_card_get_dai_id() 2) ASoC: audio-graph-card: merge audio-graph-scu-card on Doc 3) ASoC: audio-graph-card: merge audio-graph-scu-card 4) ASoC: audio-graph-card: tidyup mclk-fs method 5) ASoC: audio-graph-card: tidyup convert_rate/channel method 6) ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc 7) ASoC: audio-graph-scu-card: remove audio-graph-scu-card .../devicetree/bindings/sound/audio-graph-card.txt | 217 +++++++++ .../bindings/sound/audio-graph-scu-card.txt | 123 ----- sound/soc/generic/Kconfig | 10 +- sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-card.c | 405 ++++++++++++++--- sound/soc/generic/audio-graph-scu-card.c | 501 --------------------- sound/soc/generic/simple-card-utils.c | 3 + 7 files changed, 568 insertions(+), 693 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt delete mode 100644 sound/soc/generic/audio-graph-scu-card.c -- 2.7.4

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[alsa-devel] [asoc:for-4.21 167/168] sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div'
by kbuild test robot 13 Dec '18

13 Dec '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.21 head: 1b46ed96961db1a826b3e5cadac18c7c4857f054 commit: 5ce5658375e6de2468c4884f7ae474c4ed40a13f [167/168] ASoC: meson: add axg spdif input config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 5ce5658375e6de2468c4884f7ae474c4ed40a13f # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All warnings (new ones prefixed by >>): In file included from arch/arm/include/asm/div64.h:127:0, from include/linux/kernel.h:207, from include/linux/clk.h:16, from sound/soc//meson/axg-spdifin.c:7: sound/soc//meson/axg-spdifin.c: In function 'axg_spdifin_write_mode_param': include/asm-generic/div64.h:222:28: warning: comparison of distinct pointer types lacks a cast (void)(((typeof((n)) *)0) == ((uint64_t *)0)); \ ^ >> sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div' rem = do_div(offset, num_per_reg); ^~~~~~ In file included from include/linux/build_bug.h:5:0, from include/linux/bitfield.h:18, from sound/soc//meson/axg-spdifin.c:6: include/asm-generic/div64.h:235:25: warning: right shift count >= width of type [-Wshift-count-overflow] } else if (likely(((n) >> 32) == 0)) { \ ^ include/linux/compiler.h:76:40: note: in definition of macro 'likely' # define likely(x) __builtin_expect(!!(x), 1) ^ >> sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div' rem = do_div(offset, num_per_reg); ^~~~~~ In file included from arch/arm/include/asm/div64.h:127:0, from include/linux/kernel.h:207, from include/linux/clk.h:16, from sound/soc//meson/axg-spdifin.c:7: include/asm-generic/div64.h:239:22: error: passing argument 1 of '__div64_32' from incompatible pointer type [-Werror=incompatible-pointer-types] __rem = __div64_32(&(n), __base); \ ^ >> sound/soc//meson/axg-spdifin.c:152:8: note: in expansion of macro 'do_div' rem = do_div(offset, num_per_reg); ^~~~~~ In file included from include/linux/kernel.h:207:0, from include/linux/clk.h:16, from sound/soc//meson/axg-spdifin.c:7: arch/arm/include/asm/div64.h:33:24: note: expected 'uint64_t * {aka long long unsigned int *}' but argument is of type 'unsigned int *' static inline uint32_t __div64_32(uint64_t *n, uint32_t base) ^~~~~~~~~~ cc1: some warnings being treated as errors vim +/do_div +152 sound/soc//meson/axg-spdifin.c 5 6 #include <linux/bitfield.h> > 7 #include <linux/clk.h> 8 #include <linux/module.h> 9 #include <linux/of_platform.h> 10 #include <linux/regmap.h> 11 #include <sound/soc.h> 12 #include <sound/soc-dai.h> 13 #include <sound/pcm_params.h> 14 15 #define SPDIFIN_CTRL0 0x00 16 #define SPDIFIN_CTRL0_EN BIT(31) 17 #define SPDIFIN_CTRL0_RST_OUT BIT(29) 18 #define SPDIFIN_CTRL0_RST_IN BIT(28) 19 #define SPDIFIN_CTRL0_WIDTH_SEL BIT(24) 20 #define SPDIFIN_CTRL0_STATUS_CH_SHIFT 11 21 #define SPDIFIN_CTRL0_STATUS_SEL GENMASK(10, 8) 22 #define SPDIFIN_CTRL0_SRC_SEL GENMASK(5, 4) 23 #define SPDIFIN_CTRL0_CHK_VALID BIT(3) 24 #define SPDIFIN_CTRL1 0x04 25 #define SPDIFIN_CTRL1_BASE_TIMER GENMASK(19, 0) 26 #define SPDIFIN_CTRL1_IRQ_MASK GENMASK(27, 20) 27 #define SPDIFIN_CTRL2 0x08 28 #define SPDIFIN_THRES_PER_REG 3 29 #define SPDIFIN_THRES_WIDTH 10 30 #define SPDIFIN_CTRL3 0x0c 31 #define SPDIFIN_CTRL4 0x10 32 #define SPDIFIN_TIMER_PER_REG 4 33 #define SPDIFIN_TIMER_WIDTH 8 34 #define SPDIFIN_CTRL5 0x14 35 #define SPDIFIN_CTRL6 0x18 36 #define SPDIFIN_STAT0 0x1c 37 #define SPDIFIN_STAT0_MODE GENMASK(30, 28) 38 #define SPDIFIN_STAT0_MAXW GENMASK(17, 8) 39 #define SPDIFIN_STAT0_IRQ GENMASK(7, 0) 40 #define SPDIFIN_IRQ_MODE_CHANGED BIT(2) 41 #define SPDIFIN_STAT1 0x20 42 #define SPDIFIN_STAT2 0x24 43 #define SPDIFIN_MUTE_VAL 0x28 44 45 #define SPDIFIN_MODE_NUM 7 46 47 struct axg_spdifin_cfg { 48 const unsigned int *mode_rates; 49 unsigned int ref_rate; 50 }; 51 52 struct axg_spdifin { 53 const struct axg_spdifin_cfg *conf; 54 struct regmap *map; 55 struct clk *refclk; 56 struct clk *pclk; 57 }; 58 59 /* 60 * TODO: 61 * It would have been nice to check the actual rate against the sample rate 62 * requested in hw_params(). Unfortunately, I was not able to make the mode 63 * detection and IRQ work reliably: 64 * 65 * 1. IRQs are generated on mode change only, so there is no notification 66 * on transition between no signal and mode 0 (32kHz). 67 * 2. Mode detection very often has glitches, and may detects the 68 * lowest or the highest mode before zeroing in on the actual mode. 69 * 70 * This makes calling snd_pcm_stop() difficult to get right. Even notifying 71 * the kcontrol would be very unreliable at this point. 72 * Let's keep things simple until the magic spell that makes this work is 73 * found. 74 */ 75 76 static unsigned int axg_spdifin_get_rate(struct axg_spdifin *priv) 77 { 78 unsigned int stat, mode, rate = 0; 79 80 regmap_read(priv->map, SPDIFIN_STAT0, &stat); 81 mode = FIELD_GET(SPDIFIN_STAT0_MODE, stat); 82 83 /* 84 * If max width is zero, we are not capturing anything. 85 * Also Sometimes, when the capture is on but there is no data, 86 * mode is SPDIFIN_MODE_NUM, but not always ... 87 */ 88 if (FIELD_GET(SPDIFIN_STAT0_MAXW, stat) && 89 mode < SPDIFIN_MODE_NUM) 90 rate = priv->conf->mode_rates[mode]; 91 92 return rate; 93 } 94 95 static int axg_spdifin_prepare(struct snd_pcm_substream *substream, 96 struct snd_soc_dai *dai) 97 { 98 struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); 99 100 /* Apply both reset */ 101 regmap_update_bits(priv->map, SPDIFIN_CTRL0, 102 SPDIFIN_CTRL0_RST_OUT | 103 SPDIFIN_CTRL0_RST_IN, 104 0); 105 106 /* Clear out reset before in reset */ 107 regmap_update_bits(priv->map, SPDIFIN_CTRL0, 108 SPDIFIN_CTRL0_RST_OUT, SPDIFIN_CTRL0_RST_OUT); 109 regmap_update_bits(priv->map, SPDIFIN_CTRL0, 110 SPDIFIN_CTRL0_RST_IN, SPDIFIN_CTRL0_RST_IN); 111 112 return 0; 113 } 114 115 static int axg_spdifin_startup(struct snd_pcm_substream *substream, 116 struct snd_soc_dai *dai) 117 { 118 struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); 119 int ret; 120 121 ret = clk_prepare_enable(priv->refclk); 122 if (ret) { 123 dev_err(dai->dev, 124 "failed to enable spdifin reference clock\n"); 125 return ret; 126 } 127 128 regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 129 SPDIFIN_CTRL0_EN); 130 131 return 0; 132 } 133 134 static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, 135 struct snd_soc_dai *dai) 136 { 137 struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); 138 139 regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); 140 clk_disable_unprepare(priv->refclk); 141 } 142 143 static void axg_spdifin_write_mode_param(struct regmap *map, int mode, 144 unsigned int val, 145 unsigned int num_per_reg, 146 unsigned int base_reg, 147 unsigned int width) 148 { 149 unsigned int offset = mode, rem; 150 unsigned int reg, shift; 151 > 152 rem = do_div(offset, num_per_reg); 153 154 reg = offset * regmap_get_reg_stride(map) + base_reg; 155 shift = width * (num_per_reg - 1 - rem); 156 157 regmap_update_bits(map, reg, GENMASK(width - 1, 0) << shift, 158 val << shift); 159 } 160 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2 0/7] ALSA: HD-audio display power fixes
by Takashi Iwai 13 Dec '18

13 Dec '18

Hi, this patch a revised patchset for fixing possible HD-audio display power unbalance. Basically the "fix" is done by refactoring the whole relevant code. It starts from the Intel HD-audio runtime PM refactoring, followed by the display PM API change, and lots of code cleanups. Since it changes the display power API function, it hits both legacy and ASoC drivers. The patches are kept in topic/hda-pm-refactor branch of my sound git tree. This will be an immutable branch once when merged to for-next, so that it can be merged to ASoC tree if any conflicting change needs to be applied. The major change in v2 series is the complete removal of AZX_DCAPS_I915_POWERWELL flag in patch#3, and other minor fixes suggested by Pierre. Takashi === Takashi Iwai (7): ALSA: hda/intel: Refactoring PM code ALSA: hda: Refactor display power management ALSA: hda/intel: Drop superfluous AZX_DCAPS_I915_POWERWELL checks ALSA: hda/intel: Properly free the display power at error path ALSA: hda: Make snd_hdac_display_power() void function ASoC: hdac_hdmi: Add missing display power-off at driver removal ALSA: hda/hdmi: Always set display_power_control for Intel HSW+ codecs include/sound/hda_codec.h | 1 + include/sound/hda_component.h | 11 +- include/sound/hdaudio.h | 7 +- sound/hda/hdac_component.c | 39 +++--- sound/hda/hdac_device.c | 17 --- sound/pci/hda/hda_codec.c | 16 ++- sound/pci/hda/hda_controller.c | 11 -- sound/pci/hda/hda_controller.h | 6 +- sound/pci/hda/hda_intel.c | 220 +++++++++++++-------------------- sound/pci/hda/patch_hdmi.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 24 +--- sound/soc/intel/skylake/skl.c | 40 ++---- 12 files changed, 148 insertions(+), 252 deletions(-) -- 2.19.2

2 9

[alsa-devel] [PATCH] ASoC: rsnd: remove RSND_REG_ from rsnd_reg
by Kuninori Morimoto 13 Dec '18

13 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd is using RSND_REG_xxx for register naming, and using RSND_REG_##f style macro for read/write. The biggest reason why it uses this style is that we can avoid non-existing register access. But, its demerit is sequential register access code will be very ugly. Current rsnd driver is well tested, so, let's remove RSND_REG_ from rsnd_reg, and cleanup sequential register access code. 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 | 36 +----- sound/soc/sh/rcar/ctu.c | 69 +++------- sound/soc/sh/rcar/dvc.c | 10 +- sound/soc/sh/rcar/gen.c | 23 ++-- sound/soc/sh/rcar/rsnd.h | 329 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 59 ++------- 6 files changed, 212 insertions(+), 314 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66..e821ccc7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -249,28 +249,8 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, out = out << shift; mask = 0x0f1f << shift; - switch (id / 2) { - case 0: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL0, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL0, mask, out); - break; - case 1: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL1, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL1, mask, out); - break; - case 2: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL2, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL2, mask, out); - break; - case 3: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL3, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL3, mask, out); - break; - case 4: - rsnd_mod_bset(adg_mod, SRCIN_TIMSEL4, mask, in); - rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL4, mask, out); - break; - } + rsnd_mod_bset(adg_mod, SRCIN_TIMSEL(id / 2), mask, in); + rsnd_mod_bset(adg_mod, SRCOUT_TIMSEL(id / 2), mask, out); if (en) rsnd_mod_bset(adg_mod, DIV_EN, en, en); @@ -299,17 +279,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) if (id == 8) return; - switch (id / 4) { - case 0: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL0, mask, val); - break; - case 1: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL1, mask, val); - break; - case 2: - rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val); - break; - } + rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL(id / 4), mask, val); dev_dbg(dev, "AUDIO_CLK_SEL is 0x%x\n", val); } diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index ff41a5e..8cb06da 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -72,10 +72,7 @@ struct rsnd_ctu { struct rsnd_mod mod; struct rsnd_kctrl_cfg_m pass; - struct rsnd_kctrl_cfg_m sv0; - struct rsnd_kctrl_cfg_m sv1; - struct rsnd_kctrl_cfg_m sv2; - struct rsnd_kctrl_cfg_m sv3; + struct rsnd_kctrl_cfg_m sv[4]; struct rsnd_kctrl_cfg_s reset; int channels; u32 flags; @@ -120,7 +117,7 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); u32 cpmdr = 0; u32 scmdr = 0; - int i; + int i, j; for (i = 0; i < RSND_MAX_CHANNELS; i++) { u32 val = rsnd_kctrl_valm(ctu->pass, i); @@ -139,45 +136,13 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, CTU_SCMDR, scmdr); - if (scmdr > 0) { - rsnd_mod_write(mod, CTU_SV00R, rsnd_kctrl_valm(ctu->sv0, 0)); - rsnd_mod_write(mod, CTU_SV01R, rsnd_kctrl_valm(ctu->sv0, 1)); - rsnd_mod_write(mod, CTU_SV02R, rsnd_kctrl_valm(ctu->sv0, 2)); - rsnd_mod_write(mod, CTU_SV03R, rsnd_kctrl_valm(ctu->sv0, 3)); - rsnd_mod_write(mod, CTU_SV04R, rsnd_kctrl_valm(ctu->sv0, 4)); - rsnd_mod_write(mod, CTU_SV05R, rsnd_kctrl_valm(ctu->sv0, 5)); - rsnd_mod_write(mod, CTU_SV06R, rsnd_kctrl_valm(ctu->sv0, 6)); - rsnd_mod_write(mod, CTU_SV07R, rsnd_kctrl_valm(ctu->sv0, 7)); - } - if (scmdr > 1) { - rsnd_mod_write(mod, CTU_SV10R, rsnd_kctrl_valm(ctu->sv1, 0)); - rsnd_mod_write(mod, CTU_SV11R, rsnd_kctrl_valm(ctu->sv1, 1)); - rsnd_mod_write(mod, CTU_SV12R, rsnd_kctrl_valm(ctu->sv1, 2)); - rsnd_mod_write(mod, CTU_SV13R, rsnd_kctrl_valm(ctu->sv1, 3)); - rsnd_mod_write(mod, CTU_SV14R, rsnd_kctrl_valm(ctu->sv1, 4)); - rsnd_mod_write(mod, CTU_SV15R, rsnd_kctrl_valm(ctu->sv1, 5)); - rsnd_mod_write(mod, CTU_SV16R, rsnd_kctrl_valm(ctu->sv1, 6)); - rsnd_mod_write(mod, CTU_SV17R, rsnd_kctrl_valm(ctu->sv1, 7)); - } - if (scmdr > 2) { - rsnd_mod_write(mod, CTU_SV20R, rsnd_kctrl_valm(ctu->sv2, 0)); - rsnd_mod_write(mod, CTU_SV21R, rsnd_kctrl_valm(ctu->sv2, 1)); - rsnd_mod_write(mod, CTU_SV22R, rsnd_kctrl_valm(ctu->sv2, 2)); - rsnd_mod_write(mod, CTU_SV23R, rsnd_kctrl_valm(ctu->sv2, 3)); - rsnd_mod_write(mod, CTU_SV24R, rsnd_kctrl_valm(ctu->sv2, 4)); - rsnd_mod_write(mod, CTU_SV25R, rsnd_kctrl_valm(ctu->sv2, 5)); - rsnd_mod_write(mod, CTU_SV26R, rsnd_kctrl_valm(ctu->sv2, 6)); - rsnd_mod_write(mod, CTU_SV27R, rsnd_kctrl_valm(ctu->sv2, 7)); - } - if (scmdr > 3) { - rsnd_mod_write(mod, CTU_SV30R, rsnd_kctrl_valm(ctu->sv3, 0)); - rsnd_mod_write(mod, CTU_SV31R, rsnd_kctrl_valm(ctu->sv3, 1)); - rsnd_mod_write(mod, CTU_SV32R, rsnd_kctrl_valm(ctu->sv3, 2)); - rsnd_mod_write(mod, CTU_SV33R, rsnd_kctrl_valm(ctu->sv3, 3)); - rsnd_mod_write(mod, CTU_SV34R, rsnd_kctrl_valm(ctu->sv3, 4)); - rsnd_mod_write(mod, CTU_SV35R, rsnd_kctrl_valm(ctu->sv3, 5)); - rsnd_mod_write(mod, CTU_SV36R, rsnd_kctrl_valm(ctu->sv3, 6)); - rsnd_mod_write(mod, CTU_SV37R, rsnd_kctrl_valm(ctu->sv3, 7)); + for (i = 0; i < 4; i++) { + + if (i >= scmdr) + break; + + for (j = 0; j < RSND_MAX_CHANNELS; j++) + rsnd_mod_write(mod, CTU_SVxxR(i, j), rsnd_kctrl_valm(ctu->sv[i], j)); } rsnd_mod_write(mod, CTU_CTUIR, 0); @@ -194,10 +159,10 @@ static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, for (i = 0; i < RSND_MAX_CHANNELS; i++) { rsnd_kctrl_valm(ctu->pass, i) = 0; - rsnd_kctrl_valm(ctu->sv0, i) = 0; - rsnd_kctrl_valm(ctu->sv1, i) = 0; - rsnd_kctrl_valm(ctu->sv2, i) = 0; - rsnd_kctrl_valm(ctu->sv3, i) = 0; + rsnd_kctrl_valm(ctu->sv[0], i) = 0; + rsnd_kctrl_valm(ctu->sv[1], i) = 0; + rsnd_kctrl_valm(ctu->sv[2], i) = 0; + rsnd_kctrl_valm(ctu->sv[3], i) = 0; } rsnd_kctrl_vals(ctu->reset) = 0; } @@ -247,7 +212,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv0, RSND_MAX_CHANNELS, + &ctu->sv[0], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -256,7 +221,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv1, RSND_MAX_CHANNELS, + &ctu->sv[1], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -265,7 +230,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv2, RSND_MAX_CHANNELS, + &ctu->sv[2], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; @@ -274,7 +239,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", rsnd_kctrl_accept_anytime, NULL, - &ctu->sv3, RSND_MAX_CHANNELS, + &ctu->sv[3], RSND_MAX_CHANNELS, 0x00FFFFFF); if (ret < 0) return ret; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 27c3761..8d91c0e 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -86,14 +86,8 @@ static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, val[i] = rsnd_kctrl_valm(dvc->volume, i); /* Enable Digital Volume */ - rsnd_mod_write(mod, DVC_VOL0R, val[0]); - rsnd_mod_write(mod, DVC_VOL1R, val[1]); - rsnd_mod_write(mod, DVC_VOL2R, val[2]); - rsnd_mod_write(mod, DVC_VOL3R, val[3]); - rsnd_mod_write(mod, DVC_VOL4R, val[4]); - rsnd_mod_write(mod, DVC_VOL5R, val[5]); - rsnd_mod_write(mod, DVC_VOL6R, val[6]); - rsnd_mod_write(mod, DVC_VOL7R, val[7]); + for (i = 0; i < RSND_MAX_CHANNELS; i++) + rsnd_mod_write(mod, DVC_VOLxR(i), val[i]); } static void rsnd_dvc_volume_init(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3283d67..7cda601 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -26,8 +26,8 @@ struct rsnd_gen { struct regmap *regmap[RSND_BASE_MAX]; /* RSND_REG_MAX base */ - struct regmap_field *regs[RSND_REG_MAX]; - const char *reg_name[RSND_REG_MAX]; + struct regmap_field *regs[REG_MAX]; + const char *reg_name[REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) @@ -49,11 +49,11 @@ struct rsnd_regmap_field_conf { } /* single address mapping */ #define RSND_GEN_S_REG(id, offset) \ - RSND_REG_SET(RSND_REG_##id, offset, 0, #id) + RSND_REG_SET(id, offset, 0, #id) /* multi address mapping */ #define RSND_GEN_M_REG(id, offset, _id_offset) \ - RSND_REG_SET(RSND_REG_##id, offset, _id_offset, #id) + RSND_REG_SET(id, offset, _id_offset, #id) /* * basic function @@ -79,9 +79,9 @@ static int rsnd_mod_id_cmd(struct rsnd_mod *mod) return rsnd_mod_id(mod); } -u32 rsnd_read(struct rsnd_priv *priv, - struct rsnd_mod *mod, enum rsnd_reg reg) +u32 rsnd_mod_read(struct rsnd_mod *mod, enum rsnd_reg reg) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; @@ -98,10 +98,10 @@ u32 rsnd_read(struct rsnd_priv *priv, return val; } -void rsnd_write(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data) +void rsnd_mod_write(struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); @@ -115,9 +115,10 @@ void rsnd_write(struct rsnd_priv *priv, rsnd_reg_name(gen, reg), reg, data); } -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 mask, u32 data) +void rsnd_mod_bset(struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 94d51fe..605e4b9 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -42,165 +42,175 @@ */ enum rsnd_reg { /* SCU (MIX/CTU/DVC) */ - RSND_REG_SRC_I_BUSIF_MODE, - RSND_REG_SRC_O_BUSIF_MODE, - RSND_REG_SRC_ROUTE_MODE0, - RSND_REG_SRC_SWRSR, - RSND_REG_SRC_SRCIR, - RSND_REG_SRC_ADINR, - RSND_REG_SRC_IFSCR, - RSND_REG_SRC_IFSVR, - RSND_REG_SRC_SRCCR, - RSND_REG_SRC_CTRL, - RSND_REG_SRC_BSDSR, - RSND_REG_SRC_BSISR, - RSND_REG_SRC_INT_ENABLE0, - RSND_REG_SRC_BUSIF_DALIGN, - RSND_REG_SRCIN_TIMSEL0, - RSND_REG_SRCIN_TIMSEL1, - RSND_REG_SRCIN_TIMSEL2, - RSND_REG_SRCIN_TIMSEL3, - RSND_REG_SRCIN_TIMSEL4, - RSND_REG_SRCOUT_TIMSEL0, - RSND_REG_SRCOUT_TIMSEL1, - RSND_REG_SRCOUT_TIMSEL2, - RSND_REG_SRCOUT_TIMSEL3, - RSND_REG_SRCOUT_TIMSEL4, - RSND_REG_SCU_SYS_STATUS0, - RSND_REG_SCU_SYS_STATUS1, - 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, - RSND_REG_CTU_SWRSR, - RSND_REG_CTU_CTUIR, - RSND_REG_CTU_ADINR, - RSND_REG_CTU_CPMDR, - RSND_REG_CTU_SCMDR, - RSND_REG_CTU_SV00R, - RSND_REG_CTU_SV01R, - RSND_REG_CTU_SV02R, - RSND_REG_CTU_SV03R, - RSND_REG_CTU_SV04R, - RSND_REG_CTU_SV05R, - RSND_REG_CTU_SV06R, - RSND_REG_CTU_SV07R, - RSND_REG_CTU_SV10R, - RSND_REG_CTU_SV11R, - RSND_REG_CTU_SV12R, - RSND_REG_CTU_SV13R, - RSND_REG_CTU_SV14R, - RSND_REG_CTU_SV15R, - RSND_REG_CTU_SV16R, - RSND_REG_CTU_SV17R, - RSND_REG_CTU_SV20R, - RSND_REG_CTU_SV21R, - RSND_REG_CTU_SV22R, - RSND_REG_CTU_SV23R, - RSND_REG_CTU_SV24R, - RSND_REG_CTU_SV25R, - RSND_REG_CTU_SV26R, - RSND_REG_CTU_SV27R, - RSND_REG_CTU_SV30R, - RSND_REG_CTU_SV31R, - RSND_REG_CTU_SV32R, - RSND_REG_CTU_SV33R, - RSND_REG_CTU_SV34R, - RSND_REG_CTU_SV35R, - RSND_REG_CTU_SV36R, - RSND_REG_CTU_SV37R, - RSND_REG_MIX_SWRSR, - RSND_REG_MIX_MIXIR, - RSND_REG_MIX_ADINR, - RSND_REG_MIX_MIXMR, - RSND_REG_MIX_MVPDR, - RSND_REG_MIX_MDBAR, - RSND_REG_MIX_MDBBR, - RSND_REG_MIX_MDBCR, - RSND_REG_MIX_MDBDR, - RSND_REG_MIX_MDBER, - RSND_REG_DVC_SWRSR, - RSND_REG_DVC_DVUIR, - RSND_REG_DVC_ADINR, - RSND_REG_DVC_DVUCR, - RSND_REG_DVC_ZCMCR, - RSND_REG_DVC_VOL0R, - RSND_REG_DVC_VOL1R, - RSND_REG_DVC_VOL2R, - RSND_REG_DVC_VOL3R, - RSND_REG_DVC_VOL4R, - RSND_REG_DVC_VOL5R, - RSND_REG_DVC_VOL6R, - RSND_REG_DVC_VOL7R, - RSND_REG_DVC_DVUER, - RSND_REG_DVC_VRCTR, - RSND_REG_DVC_VRPDR, - RSND_REG_DVC_VRDBR, + SRC_I_BUSIF_MODE, + SRC_O_BUSIF_MODE, + SRC_ROUTE_MODE0, + SRC_SWRSR, + SRC_SRCIR, + SRC_ADINR, + SRC_IFSCR, + SRC_IFSVR, + SRC_SRCCR, + SRC_CTRL, + SRC_BSDSR, + SRC_BSISR, + SRC_INT_ENABLE0, + SRC_BUSIF_DALIGN, + SRCIN_TIMSEL0, + SRCIN_TIMSEL1, + SRCIN_TIMSEL2, + SRCIN_TIMSEL3, + SRCIN_TIMSEL4, + SRCOUT_TIMSEL0, + SRCOUT_TIMSEL1, + SRCOUT_TIMSEL2, + SRCOUT_TIMSEL3, + SRCOUT_TIMSEL4, + SCU_SYS_STATUS0, + SCU_SYS_STATUS1, + SCU_SYS_INT_EN0, + SCU_SYS_INT_EN1, + CMD_CTRL, + CMD_BUSIF_MODE, + CMD_BUSIF_DALIGN, + CMD_ROUTE_SLCT, + CMDOUT_TIMSEL, + CTU_SWRSR, + CTU_CTUIR, + CTU_ADINR, + CTU_CPMDR, + CTU_SCMDR, + CTU_SV00R, + CTU_SV01R, + CTU_SV02R, + CTU_SV03R, + CTU_SV04R, + CTU_SV05R, + CTU_SV06R, + CTU_SV07R, + CTU_SV10R, + CTU_SV11R, + CTU_SV12R, + CTU_SV13R, + CTU_SV14R, + CTU_SV15R, + CTU_SV16R, + CTU_SV17R, + CTU_SV20R, + CTU_SV21R, + CTU_SV22R, + CTU_SV23R, + CTU_SV24R, + CTU_SV25R, + CTU_SV26R, + CTU_SV27R, + CTU_SV30R, + CTU_SV31R, + CTU_SV32R, + CTU_SV33R, + CTU_SV34R, + CTU_SV35R, + CTU_SV36R, + CTU_SV37R, + MIX_SWRSR, + MIX_MIXIR, + MIX_ADINR, + MIX_MIXMR, + MIX_MVPDR, + MIX_MDBAR, + MIX_MDBBR, + MIX_MDBCR, + MIX_MDBDR, + MIX_MDBER, + DVC_SWRSR, + DVC_DVUIR, + DVC_ADINR, + DVC_DVUCR, + DVC_ZCMCR, + DVC_VOL0R, + DVC_VOL1R, + DVC_VOL2R, + DVC_VOL3R, + DVC_VOL4R, + DVC_VOL5R, + DVC_VOL6R, + DVC_VOL7R, + DVC_DVUER, + DVC_VRCTR, + DVC_VRPDR, + DVC_VRDBR, /* ADG */ - RSND_REG_BRRA, - RSND_REG_BRRB, - RSND_REG_BRGCKR, - RSND_REG_DIV_EN, - RSND_REG_AUDIO_CLK_SEL0, - RSND_REG_AUDIO_CLK_SEL1, - RSND_REG_AUDIO_CLK_SEL2, + BRRA, + BRRB, + BRGCKR, + DIV_EN, + AUDIO_CLK_SEL0, + AUDIO_CLK_SEL1, + AUDIO_CLK_SEL2, /* SSIU */ - RSND_REG_SSI_MODE, - RSND_REG_SSI_MODE0, - RSND_REG_SSI_MODE1, - RSND_REG_SSI_MODE2, - RSND_REG_SSI_CONTROL, - RSND_REG_SSI_CTRL, - RSND_REG_SSI_BUSIF0_MODE, - RSND_REG_SSI_BUSIF0_ADINR, - RSND_REG_SSI_BUSIF0_DALIGN, - RSND_REG_SSI_BUSIF1_MODE, - RSND_REG_SSI_BUSIF1_ADINR, - RSND_REG_SSI_BUSIF1_DALIGN, - RSND_REG_SSI_BUSIF2_MODE, - RSND_REG_SSI_BUSIF2_ADINR, - RSND_REG_SSI_BUSIF2_DALIGN, - RSND_REG_SSI_BUSIF3_MODE, - RSND_REG_SSI_BUSIF3_ADINR, - RSND_REG_SSI_BUSIF3_DALIGN, - RSND_REG_SSI_BUSIF4_MODE, - RSND_REG_SSI_BUSIF4_ADINR, - RSND_REG_SSI_BUSIF4_DALIGN, - RSND_REG_SSI_BUSIF5_MODE, - RSND_REG_SSI_BUSIF5_ADINR, - RSND_REG_SSI_BUSIF5_DALIGN, - RSND_REG_SSI_BUSIF6_MODE, - RSND_REG_SSI_BUSIF6_ADINR, - RSND_REG_SSI_BUSIF6_DALIGN, - RSND_REG_SSI_BUSIF7_MODE, - RSND_REG_SSI_BUSIF7_ADINR, - RSND_REG_SSI_BUSIF7_DALIGN, - RSND_REG_SSI_INT_ENABLE, - RSND_REG_SSI_SYS_STATUS0, - RSND_REG_SSI_SYS_STATUS1, - RSND_REG_SSI_SYS_STATUS2, - RSND_REG_SSI_SYS_STATUS3, - RSND_REG_SSI_SYS_STATUS4, - RSND_REG_SSI_SYS_STATUS5, - RSND_REG_SSI_SYS_STATUS6, - RSND_REG_SSI_SYS_STATUS7, - RSND_REG_HDMI0_SEL, - RSND_REG_HDMI1_SEL, + SSI_MODE, + SSI_MODE0, + SSI_MODE1, + SSI_MODE2, + SSI_CONTROL, + SSI_CTRL, + SSI_BUSIF0_MODE, + SSI_BUSIF1_MODE, + SSI_BUSIF2_MODE, + SSI_BUSIF3_MODE, + SSI_BUSIF4_MODE, + SSI_BUSIF5_MODE, + SSI_BUSIF6_MODE, + SSI_BUSIF7_MODE, + SSI_BUSIF0_ADINR, + SSI_BUSIF1_ADINR, + SSI_BUSIF2_ADINR, + SSI_BUSIF3_ADINR, + SSI_BUSIF4_ADINR, + SSI_BUSIF5_ADINR, + SSI_BUSIF6_ADINR, + SSI_BUSIF7_ADINR, + SSI_BUSIF0_DALIGN, + SSI_BUSIF1_DALIGN, + SSI_BUSIF2_DALIGN, + SSI_BUSIF3_DALIGN, + SSI_BUSIF4_DALIGN, + SSI_BUSIF5_DALIGN, + SSI_BUSIF6_DALIGN, + SSI_BUSIF7_DALIGN, + SSI_INT_ENABLE, + SSI_SYS_STATUS0, + SSI_SYS_STATUS1, + SSI_SYS_STATUS2, + SSI_SYS_STATUS3, + SSI_SYS_STATUS4, + SSI_SYS_STATUS5, + SSI_SYS_STATUS6, + SSI_SYS_STATUS7, + HDMI0_SEL, + HDMI1_SEL, /* SSI */ - RSND_REG_SSICR, - RSND_REG_SSISR, - RSND_REG_SSITDR, - RSND_REG_SSIRDR, - RSND_REG_SSIWSR, + SSICR, + SSISR, + SSITDR, + SSIRDR, + SSIWSR, - RSND_REG_MAX, + REG_MAX, }; +#define SRCIN_TIMSEL(i) (SRCIN_TIMSEL0 + (i)) +#define SRCOUT_TIMSEL(i) (SRCOUT_TIMSEL0 + (i)) +#define CTU_SVxxR(i, j) (CTU_SV00R + (i * 8) + (j)) +#define DVC_VOLxR(i) (DVC_VOL0R + (i)) +#define AUDIO_CLK_SEL(i) (AUDIO_CLK_SEL0 + (i)) +#define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) +#define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) +#define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) + struct rsnd_priv; struct rsnd_mod; @@ -210,20 +220,9 @@ struct rsnd_dai_stream; /* * R-Car basic functions */ -#define rsnd_mod_read(m, r) \ - rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r) -#define rsnd_mod_write(m, r, d) \ - rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) -#define rsnd_mod_bset(m, r, s, d) \ - rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) - -u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data); -void rsnd_force_write(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data); -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, - u32 mask, u32 data); +u32 rsnd_mod_read(struct rsnd_mod *mod, enum rsnd_reg reg); +void rsnd_mod_write(struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); +void rsnd_mod_bset(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); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 04b1a48..c5934ad 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -65,6 +65,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, int id = rsnd_mod_id(mod); u32 mask1, val1; u32 mask2, val2; + int i; /* clear status */ switch (id) { @@ -73,16 +74,12 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, case 2: case 3: case 4: - rsnd_mod_write(mod, SSI_SYS_STATUS0, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS2, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS4, 0xf << (id * 4)); - rsnd_mod_write(mod, SSI_SYS_STATUS6, 0xf << (id * 4)); + for (i = 0; i < 4; i++) + rsnd_mod_write(mod, SSI_SYS_STATUS(i * 2), 0xf << (id * 4)); break; case 9: - rsnd_mod_write(mod, SSI_SYS_STATUS1, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS3, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS5, 0xf << 4); - rsnd_mod_write(mod, SSI_SYS_STATUS7, 0xf << 4); + for (i = 0; i < 4; i++) + rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << (id * 4)); break; } @@ -198,43 +195,15 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, id, busif); } -#define RSND_WRITE_BUSIF(i) \ - rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR, \ - rsnd_get_adinr_bit(mod, io) | \ - (rsnd_io_is_play(io) ? \ - rsnd_runtime_channel_after_ctu(io) : \ - rsnd_runtime_channel_original(io))); \ - rsnd_mod_write(mod, SSI_BUSIF##i##_MODE, \ - rsnd_get_busif_shift(io, mod) | 1); \ - rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN, \ - rsnd_get_dalign(mod, io)) - - switch (busif) { - case 0: - RSND_WRITE_BUSIF(0); - break; - case 1: - RSND_WRITE_BUSIF(1); - break; - case 2: - RSND_WRITE_BUSIF(2); - break; - case 3: - RSND_WRITE_BUSIF(3); - break; - case 4: - RSND_WRITE_BUSIF(4); - break; - case 5: - RSND_WRITE_BUSIF(5); - break; - case 6: - RSND_WRITE_BUSIF(6); - break; - case 7: - RSND_WRITE_BUSIF(7); - break; - } + rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), + rsnd_get_adinr_bit(mod, io) | + (rsnd_io_is_play(io) ? + rsnd_runtime_channel_after_ctu(io) : + rsnd_runtime_channel_original(io))); + rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), + rsnd_get_busif_shift(io, mod) | 1); + rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), + rsnd_get_dalign(mod, io)); } if (has_hdmi0 || has_hdmi1) { -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M" to the asoc tree
by Mark Brown 13 Dec '18

13 Dec '18

The patch ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From add6883619a9e3bf9658eaff1a547354131bbcd9 Mon Sep 17 00:00:00 2001 From: Fabio Estevam <festevam(a)gmail.com> Date: Thu, 13 Dec 2018 00:08:38 -0200 Subject: [PATCH] ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M eukrea-tlv320.c machine driver runs on non-DT platforms and include <asm/mach-types.h> header file in order to be able to use some machine_is_eukrea_xxx() macros. Building it for ARM64 causes the following build error: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320 driver when ARM64 is selected. This is needed in preparation for the i.MX8M support. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> Acked-by: Shawn Guo <shawnguo(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..2e75b5bc5f1d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: fsl-sai: Fix typo in "transmitter"
by Fabio Estevam 13 Dec '18

13 Dec '18

Fix the spelling of "transmitter". Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index dd9e597..2e726b9 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -35,13 +35,13 @@ Required properties: - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transimitter and the + with Rx) which means both the transmitter and the receiver will send and receive data by following receiver's bit clocks and frame sync clocks. - fsl,sai-asynchronous: This is a boolean property. If present, indicating that SAI will work in the asynchronous mode, which - means both transimitter and receiver will send and + means both transmitter and receiver will send and receive data by following their own bit clocks and frame sync clocks separately. @@ -58,8 +58,8 @@ Optional properties (for mx6ul): Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both - transimitter and receiver will send and receive data by following clocks - of transimitter. + transmitter and receiver will send and receive data by following clocks + of transmitter. - fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. Example: -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: simple-card: Dereference pointer for memcpy sizeof in asoc_simple_card_probe
by Kuninori Morimoto 13 Dec '18

13 Dec '18

From: Nathan Chancellor <natechancellor(a)gmail.com> Commit 4fb7f4df49d3 ("ASoC: simple-card: use cpu/codec pointer on simple_dai_props") updated {cpu,codec}_dai to be pointers in struct simple_dai_props but didn't update these locations to dereference the pointers. This patch fixup it for non DT simple-card use case. Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 45d3e8c..fb18e51 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -665,6 +665,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct asoc_simple_card_info *cinfo; struct snd_soc_dai_link_component *codecs; struct snd_soc_dai_link_component *platform; + int dai_idx = 0; cinfo = dev->platform_data; if (!cinfo) { @@ -681,6 +682,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } + dai_props->cpu_dai = &priv->dais[dai_idx++]; + dai_props->codec_dai = &priv->dais[dai_idx++]; + codecs = dai_link->codecs; codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; @@ -694,10 +698,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; - memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, - sizeof(priv->dai_props->cpu_dai)); - memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, - sizeof(priv->dai_props->codec_dai)); + memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, + sizeof(*priv->dai_props->cpu_dai)); + memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, + sizeof(*priv->dai_props->codec_dai)); } snd_soc_card_set_drvdata(card, priv); -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: meson: add axg spdif input support
by Jerome Brunet 13 Dec '18

13 Dec '18

This patchset adds the initial support the spdif input found on the axg SoC family. The capture itself works well but the rate detection could be enhanced in the future. Like several other drivers of this type, no check is done to verify if the rate of the input stream is consistent with the rate requested in hw_params() The automatic rate detection mechanism of this device and the related IRQ proved difficult to work with, as documented in the driver. Several approaches to work around this minor issue have been tried but none provided a complete and reliable solution so far. Hopefully more eyes on this will help figure this out. Since I authored all Amlogic ASoC drivers (and the related bugs) merged so far, I have added myself as maintainer of them in the last patch of this series. Jerome Brunet (4): ASoC: meson: axg-toddr: add support for spdifin backend ASoC: meson: add axg spdif input DT binding documentation ASoC: meson: add axg spdif input MAINTAINERS: Add Amlogic sound drivers entry .../bindings/sound/amlogic,axg-spdifin.txt | 22 + MAINTAINERS | 7 + sound/soc/meson/Kconfig | 9 +- sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-fifo.h | 3 +- sound/soc/meson/axg-spdifin.c | 521 ++++++++++++++++++ sound/soc/meson/axg-toddr.c | 15 +- 7 files changed, 567 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 sound/soc/meson/axg-spdifin.c -- 2.19.2

2 8

[alsa-devel] [PATCH] ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M
by Fabio Estevam 13 Dec '18

13 Dec '18

eukrea-tlv320.c machine driver runs on non-DT platforms and include <asm/mach-types.h> header file in order to be able to use some machine_is_eukrea_xxx() macros. Since commit 0e1532bfecf4 ("arm64: add basic Kconfig symbols for i.MX8") the ARCH_MXC symbol is visible on ARM64 too. Building it for ARM64 causes the following build error: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320 driver when ARM64 is selected. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Hi Mark, Commit commit 0e1532bfecf4 ("arm64: add basic Kconfig symbols for i.MX8") is present in Shawn's tree, so maybe this should be taken by Shawn? sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb..2e75b5b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: fsl: Add a dependency on ARM for Eukrea
by Mark Brown 13 Dec '18

13 Dec '18

The MXC architecture became buildable on arm64 which doesn't have a mach-types.h so add an explicit 32 bit dependency. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb4a934..ff2d4a8783f9 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_ARM && ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.19.0.rc2

1 0

[alsa-devel] [bunk@debian.org] Bug#916052: alsa-oss FTBFS with glibc 2.28
by Elimar Riesebieter 13 Dec '18

13 Dec '18

Control: forwarded -1 alsa-devel(a)alsa-project.org Hi, could you please investigate? It seems as alsa-oss doesn't built with glibc 2.28. Thanks in advance Elimar -- Learned men are the cisterns of knowledge, not the fountainheads ;-)

2 2

[alsa-devel] [PATCH v11 0/3] Huawei laptops
by Ayman Bagabas 13 Dec '18

13 Dec '18

This patch set is based on the new audio LED triggers in topic/leds-trigger branch from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes in v11: * Minor code changes Changes in v10: * Use ec_get_handle instead of acpi_get_handle since we are using the ec device * Switch to WMI0_EXPENSIVE_GUID and wmi_query_block to fetch keycode when WMI0 is used Changes in v9: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes in v8: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes in v7: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes in v6: * Review tags Changes in v5: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes in v4: * Code formatting Changes in v3: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 17 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 208 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 ++++ 4 files changed, 248 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

2 4

[alsa-devel] [PATCH v2] ASoC: fsl: Fix SND_SOC_EUKREA_TLV320 build error on i.MX8M
by Fabio Estevam 13 Dec '18

13 Dec '18

eukrea-tlv320.c machine driver runs on non-DT platforms and include <asm/mach-types.h> header file in order to be able to use some machine_is_eukrea_xxx() macros. Building it for ARM64 causes the following build error: sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320 driver when ARM64 is selected. This is needed in preparation for the i.MX8M support. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Changes since v1: - Drop the mention to the i.MX8M commit that is currently only in Shawn's tree. sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6ec19fb..2e75b5b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -221,7 +221,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: Dereference pointer for memcpy sizeof in asoc_simple_card_probe
by Nathan Chancellor 13 Dec '18

13 Dec '18

Clang warns: sound/soc/generic/simple-card.c:462:6: warning: argument to 'sizeof' in 'memcpy' call is the same pointer type 'struct asoc_simple_dai *' as the source; expected 'struct asoc_simple_dai' or an explicit length [-Wsizeof-pointer-memaccess] sizeof(priv->dai_props->cpu_dai)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/generic/simple-card.c:464:6: warning: argument to 'sizeof' in 'memcpy' call is the same pointer type 'struct asoc_simple_dai *' as the source; expected 'struct asoc_simple_dai' or an explicit length [-Wsizeof-pointer-memaccess] sizeof(priv->dai_props->codec_dai)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 2 warnings generated. Commit 4fb7f4df49d3 ("ASoC: simple-card: use cpu/codec pointer on simple_dai_props") updated {cpu,codec}_dai to be pointers in struct simple_dai_props but didn't update these locations to dereference the pointers to get the proper size of their contents. Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/generic/simple-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 766123485d7c..d4738d3eb2f1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -459,9 +459,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, - sizeof(priv->dai_props->cpu_dai)); + sizeof(*priv->dai_props->cpu_dai)); memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, - sizeof(priv->dai_props->codec_dai)); + sizeof(*priv->dai_props->codec_dai)); } snd_soc_card_set_drvdata(card, priv); -- 2.20.0

2 2

[alsa-devel] [PATCH] ALSA: pcm: Fix potential Spectre v1 vulnerability
by Gustavo A. R. Silva 12 Dec '18

12 Dec '18

stream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/pcm.c:140 snd_pcm_control_ioctl() warn: potential spectre issue 'pcm->streams' [r] (local cap) Fix this by sanitizing stream before using it to index pcm->streams Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index fdb9b92fc8d6..23141003a90d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -32,6 +32,8 @@ #include <sound/control.h> #include <sound/info.h> +#include <linux/nospec.h> + #include "pcm_local.h" MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); @@ -129,6 +131,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -EFAULT; if (stream < 0 || stream > 1) return -EINVAL; + stream = array_index_nospec(stream, 2); if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(&register_mutex); -- 2.17.1

3 2

[alsa-devel] [PATCH] alsa-oss: fix FTBFS with glibc v2.28
by GitHub pull_request - opened 12 Dec '18

12 Dec '18

alsa-project/alsa-oss pull request #1 was opened from takaswie: In glibc upstream, libio/libio.h had been deprecated for applications and became internal header in its v2.28 release. We can see below message in its release note[1]. * The nonstandard header files <libio.h> and <_G_config.h> are no longer installed. Software that was using either header should be updated to use standard <stdio.h> interfaces instead. This brings FTBFS with glibc v2.28 due to missing <libio.h>. This commit fixes it. [1] https://www.sourceware.org/ml/libc-alpha/2018-08/msg00003.html Reported-by: Adrian Bunk <bunk(a)debian.org> Reference: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=916052 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> ``` --- alsa/stdioemu.c | 1 - 1 file changed, 1 deletion(-) ``` Request URL : https://github.com/alsa-project/alsa-oss/pull/1 Patch URL : https://github.com/alsa-project/alsa-oss/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-oss

1 0

[alsa-devel] [PATCH] ALSA: emux_hwdep: Fix potential Spectre v1 vulnerabilities
by Gustavo A. R. Silva 12 Dec '18

12 Dec '18

info.mode and info.port are indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. These issues were detected with the help of Smatch: sound/synth/emux/emux_hwdep.c:72 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[i]->ctrls' [w] (local cap) sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs' [w] (local cap) sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[info.port]->ctrls' [w] (local cap) Fix this by sanitizing both info.mode and info.port before using them to index emu->portptrs[i]->ctrls, emu->portptrs[info.port]->ctrls and emu->portptrs. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2 Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/synth/emux/emux_hwdep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index e557946718a9..a0d5cd99572f 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -24,6 +24,7 @@ #include <linux/uaccess.h> #include "emux_voice.h" +#include <linux/nospec.h> #define TMP_CLIENT_ID 0x1001 @@ -66,13 +67,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg) return -EFAULT; if (info.mode < 0 || info.mode >= EMUX_MD_END) return -EINVAL; + info.mode = array_index_nospec(info.mode, EMUX_MD_END); if (info.port < 0) { for (i = 0; i < emu->num_ports; i++) emu->portptrs[i]->ctrls[info.mode] = info.value; } else { - if (info.port < emu->num_ports) + if (info.port < emu->num_ports) { + info.port = array_index_nospec(info.port, emu->num_ports); emu->portptrs[info.port]->ctrls[info.mode] = info.value; + } } return 0; } -- 2.17.1

3 2

[alsa-devel] [PATCH v10 0/3] Huawei laptops
by Ayman Bagabas 12 Dec '18

12 Dec '18

This patch set is based on the new audio LED triggers in topic/leds-trigger branch from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes in v10: * Use ec_get_handle instead of acpi_get_handle since we are using the ec device * Switch to WMI0_EXPENSIVE_GUID and wmi_query_block to fetch keycode when WMI0 is used Changes in v9: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes in v8: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes in v7: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes in v6: * Review tags Changes in v5: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes in v4: * Code formatting Changes in v3: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 17 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 220 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 260 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

4 10

[alsa-devel] [PATCH 0/6] ASoC: merge audio-graph-scu into audio-graph - final step
by Kuninori Morimoto 12 Dec '18

12 Dec '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is final step patches. Kuninori Morimoto (6): ASoC: audio-graph-card: merge audio-graph-scu-card on Doc ASoC: audio-graph-card: merge audio-graph-scu-card ASoC: audio-graph-card: tidyup mclk-fs method ASoC: audio-graph-card: tidyup convert_rate/channel method ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc ASoC: audio-graph-scu-card: remove audio-graph-scu-card .../devicetree/bindings/sound/audio-graph-card.txt | 222 +++++++++ .../bindings/sound/audio-graph-scu-card.txt | 123 ----- sound/soc/generic/Kconfig | 10 +- sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-card.c | 397 +++++++++++++--- sound/soc/generic/audio-graph-scu-card.c | 501 --------------------- 6 files changed, 563 insertions(+), 692 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt delete mode 100644 sound/soc/generic/audio-graph-scu-card.c -- 2.7.4

1 7

[alsa-devel] [PATCH v2] ASoC: Intel: kbl_rt5660: Add a new machine driver for kbl with rt5660
by Hui Wang 12 Dec '18

12 Dec '18

The new Dell IoT platform uses kabylake + alc3277 codec, and alc3277 shares the driver with the codec rt5660, here we generate a new machine driver based on kbl_da7219_max98357a. The audio design on this IoT platform is as below: - Intel kabylake platform - connect the codec ALC3277 via SSP0 - line-out and line-in with Micbias jacks - line-out mute control and jack detection of line-out and line-in - two HDMI ports with audio capability Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/intel/boards/Kconfig | 10 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_rt5660.c | 536 ++++++++++++++++++ .../intel/common/soc-acpi-intel-kbl-match.c | 5 + 4 files changed, 553 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_rt5660.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index b177db2a0dbb..3839d6205fcf 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,6 +293,16 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_KBL_RT5660_MACH + tristate "KBL with RT5660 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5660 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5660 I2S audio codec. + Say Y if you have such a device. + config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 5381e27df9cc..bf072ea299b7 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -20,6 +20,7 @@ snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o +snd-soc-kbl_rt5660-objs := kbl_rt5660.o snd-soc-skl_rt286-objs := skl_rt286.o snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o @@ -46,6 +47,7 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max9 obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5660_MACH) += snd-soc-kbl_rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c new file mode 100644 index 000000000000..0482e6a7fc2a --- /dev/null +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -0,0 +1,536 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018-19 Intel Corporation. + +/* + * Intel Kabylake I2S Machine Driver with RT5660 Codec + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAXIM98357a and + * DA7219 codecs + * Also referred to: + * Intel Broadwell I2S Machine driver supporting RT5677 codec + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../../codecs/hdac_hdmi.h" +#include "../../codecs/rt5660.h" + +#define KBL_RT5660_CODEC_DAI "rt5660-aif1" +#define DUAL_CHANNEL 2 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack skylake_hdmi[3]; +static struct snd_soc_jack lineout_jack; +static struct snd_soc_jack mic_jack; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct gpio_desc *gpio_lo_mute; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, +}; + +#define GPIO_LINEOUT_MUTE_INDEX 0 +#define GPIO_LINEOUT_DET_INDEX 3 +#define GPIO_LINEIN_DET_INDEX 4 + +static const struct acpi_gpio_params lineout_mute_gpio = { GPIO_LINEOUT_MUTE_INDEX, 0, true }; +static const struct acpi_gpio_params lineout_det_gpio = { GPIO_LINEOUT_DET_INDEX, 0, false }; +static const struct acpi_gpio_params mic_det_gpio = { GPIO_LINEIN_DET_INDEX, 0, false }; + + +static const struct acpi_gpio_mapping acpi_rt5660_gpios[] = { + { "lineout-mute-gpios", &lineout_mute_gpio , 1 }, + { "lineout-det-gpios", &lineout_det_gpio, 1 }, + { "mic-det-gpios", &mic_det_gpio, 1 }, + { NULL }, +}; + +static struct snd_soc_jack_pin lineout_jack_pin = { + .pin = "Line Out", + .mask = SND_JACK_LINEOUT, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Line In", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio lineout_jack_gpio = { + .name = "lineout-det", + .report = SND_JACK_LINEOUT, + .debounce_time = 200, +}; + +static struct snd_soc_jack_gpio mic_jack_gpio = { + .name = "mic-det", + .report = SND_JACK_MICROPHONE, + .debounce_time = 200, +}; + +static int kabylake_5660_event_lineout(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(dapm->card); + + gpiod_set_value_cansleep(priv->gpio_lo_mute, + !(SND_SOC_DAPM_EVENT_ON(event))); + + return 0; +} + +static const struct snd_kcontrol_new kabylake_rt5660_controls[] = { + SOC_DAPM_PIN_SWITCH("Line In"), + SOC_DAPM_PIN_SWITCH("Line Out"), +}; + +static const struct snd_soc_dapm_widget kabylake_rt5660_widgets[] = { + SND_SOC_DAPM_MIC("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", kabylake_5660_event_lineout), + SND_SOC_DAPM_SPK("DP", NULL), + SND_SOC_DAPM_SPK("HDMI", NULL), +}; + +static const struct snd_soc_dapm_route kabylake_rt5660_map[] = { + /* other jacks */ + {"IN1P", NULL, "Line In"}, + {"IN2P", NULL, "Line In"}, + {"Line Out", NULL, "LOUTR"}, + {"Line Out", NULL, "LOUTL"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, +}; + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios); + if (ret) + dev_warn(component->dev, "Failed to add driver gpios\n"); + + /* Request rt5660 GPIO for lineout mute control */ + ctx->gpio_lo_mute = devm_gpiod_get(component->dev, "lineout-mute", + GPIOD_OUT_HIGH); + if (IS_ERR(ctx->gpio_lo_mute)) { + dev_err(component->dev, "Can't find GPIO_MUTE# gpio\n"); + return PTR_ERR(ctx->gpio_lo_mute); + } + + /* Create and initialize headphone jack */ + if (!snd_soc_card_jack_new(rtd->card, "Lineout Jack", + SND_JACK_LINEOUT, &lineout_jack, + &lineout_jack_pin, 1)) { + lineout_jack_gpio.gpiod_dev = component->dev; + if (snd_soc_jack_add_gpios(&lineout_jack, 1, + &lineout_jack_gpio)) + dev_err(component->dev, "Can't add Lineout jack gpio\n"); + } else + dev_err(component->dev, "Can't create Lineout jack\n"); + + /* Create and initialize mic jack */ + if (!snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, &mic_jack, + &mic_jack_pin, 1)) { + mic_jack_gpio.gpiod_dev = component->dev; + if (snd_soc_jack_add_gpios(&mic_jack, 1, &mic_jack_gpio)) + dev_err(component->dev, "Can't add mic jack gpio\n"); + } else + dev_err(component->dev, "Can't create mic jack\n"); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "BST1"); + snd_soc_dapm_force_enable_pin(dapm, "BST2"); + + return 0; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5660_SCLK_S_PLL1, params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, + RT5660_PLL1_S_BCLK, + params_rate(params) * 50, + params_rate(params) * 512); + if (ret < 0) + dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops kabylake_rt5660_ops = { + .hw_params = kabylake_rt5660_hw_params, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_rt5660_fe_ops = { + .startup = kbl_fe_startup, +}; + +/* kabylake digital audio interface glue - connects rt5660 codec <--> CPU */ +static struct snd_soc_dai_link kabylake_rt5660_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_rt5660_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_rt5660_fe_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-10EC3277:00", + .codec_dai_name = KBL_RT5660_CODEC_DAI, + .init = kabylake_rt5660_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp0_fixup, + .ops = &kabylake_rt5660_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "iDisp1", + .id = 1, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 2, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 3, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + + +#define NAME_SIZE 32 +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &skylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &skylake_hdmi[i]); + if (err < 0) + return err; + + i++; + + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* kabylake audio machine driver for rt5660 */ +static struct snd_soc_card kabylake_audio_card_rt5660 = { + .name = "kblrt5660", + .owner = THIS_MODULE, + .dai_link = kabylake_rt5660_dais, + .num_links = ARRAY_SIZE(kabylake_rt5660_dais), + .controls = kabylake_rt5660_controls, + .num_controls = ARRAY_SIZE(kabylake_rt5660_controls), + .dapm_widgets = kabylake_rt5660_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_rt5660_widgets), + .dapm_routes = kabylake_rt5660_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_rt5660", + .driver_data = + (kernel_ulong_t)&kabylake_audio_card_rt5660, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_rt5660", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio Machine driver-RT5660 in I2S mode"); +MODULE_AUTHOR("Hui Wang <hui.wang(a)canonical.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_rt5660"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index a317b7790fce..1e41c7ded9e9 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -96,6 +96,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98927_codecs, .pdata = &skl_dmic_data }, + { + .id = "10EC3277", + .drv_name = "kbl_rt5660", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirks for Dell Optiplex 5060, 7060 for audio out while using headphones
by Girija Kumar Kasinadhuni 12 Dec '18

12 Dec '18

There are two audio ports on these machines - headset and mic. There' s no sound when hp is inserted into the headset port, though the hp is detected. Without pulseaudio, this issue is seen on both the Optiplex machines. The issue goes away with the "options snd-hda-intel model=dell-headset-dock" quirk in the /etc/modprobe.d/alsa-base.conf or /etc/modprobe.d/alsa.conf Getting this fix into the kernel instead of fixing in an application layer above, or in a .conf file. This is also helpful for audio clients using the ALSA api like cras from ChromiumOS. Signed-off-by: Girija Kumar Kasinadhuni <gkumar(a)neverware.com> --- alsa-info: Optiplex 5060 - - without fix: http://www.alsa-project.org/db/?f=2cf3c4354e82bde4aa480612a914535982fa3f91 - with fix: http://www.alsa-project.org/db/?f=7a928960400ec828b82632411284ad8a63727b92 Optiplex 7060 - without fix: http://www.alsa-project.org/db/?f=69fa554faa712e7cad1f188be4a5192353d64f2c - with fix: http://www.alsa-project.org/db/?f=80ce1fb178b377b27b0795ac1c55e8cbee1b5b96 sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8d75597028ee..581f6a288abf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6444,6 +6444,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x085b, "Dell Optiplex 5060", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x085a, "Dell Optiplex 7060", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.17.1

2 1

[alsa-devel] RME Fireface alsa driver
by thet 11 Dec '18

11 Dec '18

I posted this a couple of weeks ago in the alsa users list but there was no response. I hope someone here can help. I have a FF400 and I have been using ffado with it. It came to my attention that there exists a ALSA driver for the FF400 since the midi was working over alsa and midi is not supported in the ffado driver. Also my system can see the FF400 in the sound control panel without jack or ffado running. I am using Mint 19 cinnamon and the liquorix kernel. (though so far I have not noticed any difference between the liquorix kernel and the generic one except pre-empt performance). I would like to try using just the alsa driver and not ffado. Since I'm using the alsa midi support anyway it would be less total software to use the alsa driver for everything - assuming it supports everything I need for serious music recording. However I can't get any sound out of the FF400 using the ALSA driver - it works ok with FFADO though. Also the midi works - presumably using the alsa driver. I have tried using alsa directly in the sound control panel, using the alsa output from audacious, and the alsa driver from jack. I get an error about setting the frames/period/buffer size. The red light on the FF400 that shows no connection to a host stays on - even though the system can see it is present. This light goes off if I start ffado and jack, or if I start ffado mixer. Is there a guide somewhere on how to set up the ALSA drivers for the FF400? thanks! Thet

2 1

[alsa-devel] [PATCH 0/5 v2] ASoC: merge simple-scu-card into simple-card - step1
by Kuninori Morimoto 11 Dec '18

11 Dec '18

Hi Mark These are based on latest "for-4.21" branch. Kuninori Morimoto (5): ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf sound/soc/generic/simple-scu-card.c | 240 +++++++++++++++++++++++++++--------- 1 file changed, 185 insertions(+), 55 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 6

[alsa-devel] Applied "ASoC: simple-scu-card: care link / dai count" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: care link / dai count has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b8d8f9a58fd971ec215fb8bfec5a4fb6e539a3ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:08 +0000 Subject: [PATCH] ASoC: simple-scu-card: care link / dai count In DPCM case, it uses CPU-dummy / dummy-Codec dai links. If sound card is caring only DPCM, link count = dai count, but, if non DPCM case, link count != dai count. Now, we want to merge simple-card and simple-scu-card, then, we need to care both link / dai count more carefly This patch cares it, and prepare for merging simple card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 90 ++++++++++++++++++++++++++--- 1 file changed, 83 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 666ace6175be..1c6382fa54a0 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -256,6 +256,80 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) return 0; } +static void asoc_simple_card_get_dais_count(struct device *dev, + int *link_num, + int *dais_num, + int *ccnf_num) +{ + struct device_node *top = dev->of_node; + struct device_node *node; + int loop; + int num; + + /* + * link_num : number of links. + * CPU-Codec / CPU-dummy / dummy-Codec + * dais_num : number of DAIs + * ccnf_num : number of codec_conf + * same number for "dummy-Codec" + * + * ex1) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 7 + * CPU2 -/ ccnf : 1 + * CPU3 --- Codec2 + * + * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec + * => 7 DAIs = 4xCPU + 3xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex2) + * CPU0 --- Codec0 link : 5 + * CPU1 --- Codec1 dais : 6 + * CPU2 -/ ccnf : 1 + * CPU3 -/ + * + * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec + * + * ex3) + * CPU0 --- Codec0 link : 6 + * CPU1 -/ dais : 6 + * CPU2 --- Codec1 ccnf : 2 + * CPU3 -/ + * + * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec + * => 6 DAIs = 4xCPU + 2xCodec + * => 2 ccnf = 2xdummy-Codec + */ + if (!top) { + (*link_num) = 1; + (*dais_num) = 2; + (*ccnf_num) = 0; + return; + } + + loop = 1; + node = of_get_child_by_name(top, PREFIX "dai-link"); + if (!node) { + node = top; + loop = 0; + } + + do { + num = of_get_child_count(node); + (*dais_num) += num; + if (num > 2) { + (*link_num) += num; + (*ccnf_num)++; + } else { + (*link_num)++; + } + node = of_get_next_child(top, node); + } while (loop && node); +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -263,18 +337,20 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_dai_props *dai_props; struct snd_soc_card *card; struct device *dev = &pdev->dev; - struct device_node *np = dev->of_node; - int num, ret, i; + int ret, i; + int lnum = 0, dnum = 0, cnum = 0; /* Allocate the private data */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; - num = of_get_child_count(np); + asoc_simple_card_get_dais_count(dev, &lnum, &dnum, &cnum); + if (!lnum || !dnum) + return -EINVAL; - dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); + dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); if (!dai_props || !dai_link) return -ENOMEM; @@ -284,7 +360,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) * see * soc-core.c :: snd_soc_init_multicodec() */ - for (i = 0; i < num; i++) { + for (i = 0; i < lnum; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; dai_link[i].platform = &dai_props[i].platform; @@ -298,7 +374,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->dai_link = priv->dai_link; - card->num_links = num; + card->num_links = lnum; card->codec_conf = &priv->codec_conf; card->num_configs = 1; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 09ee833679761f4995daaa6fd6200c77b8fcdaef Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:18 +0000 Subject: [PATCH] ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing simple-scu-card.c is supporting "convert-rate/channels" which is used for DPCM. But, sound card might have multi codecs, and each codec might need each convert-rate/channels. This patch supports each codec's convert-rate/channles support. top node convert-rate/channels will overwrite settings if exist. It can't support each codec's convert-rate/channels if sound card had multi codecs without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 1c6382fa54a0..3c4e28995872 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -26,6 +26,7 @@ struct simple_card_data { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; + struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -86,7 +87,11 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + /* overwrite by top level adata if exist */ asoc_simple_card_convert_fixup(&priv->adata, params); return 0; @@ -177,6 +182,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, "prefix"); } + asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); + ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); if (ret) return ret; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e6a3ff2ed4c9c4df217c64e0fc0a280a24055790 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:27 +0000 Subject: [PATCH] ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge simple-card and simple-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on simple-scu-card. It is same logic with simple-card, thus easy merging. This is prepare for merging simple card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 77 +++++++++++++++++++---------- 1 file changed, 52 insertions(+), 25 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 3c4e28995872..944a6f435f88 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -23,12 +23,14 @@ struct simple_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; struct simple_dai_props { - struct asoc_simple_dai dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; }; @@ -47,8 +49,17 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); - return asoc_simple_card_clk_enable(&dai_props->dai); + return ret; } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) @@ -58,7 +69,9 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_simple_card_ops = { @@ -68,19 +81,21 @@ static const struct snd_soc_ops asoc_simple_card_ops = { static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct simple_dai_props *dai_props; - int num = rtd->num; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; - dai_link = simple_priv_to_link(priv, num); - dai_props = simple_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, &dai_props->dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; } static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -101,13 +116,14 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int idx, bool is_fe, + int *dai_idx, int link_idx, int is_fe, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, link_idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, link_idx); struct snd_soc_card *card = simple_priv_to_card(priv); + struct asoc_simple_dai *dai; char *prefix = ""; int ret; @@ -129,12 +145,15 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -155,11 +174,14 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); if (ret < 0) return ret; @@ -184,7 +206,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, asoc_simple_card_parse_convert(dev, link, prefix, &dai_props->adata); - ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; @@ -215,7 +237,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); bool is_fe; - int ret, i, loop; + int ret, loop; + int dai_idx, link_idx; if (!top) return -EINVAL; @@ -231,7 +254,8 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) asoc_simple_card_parse_convert(dev, top, PREFIX, &priv->adata); loop = 1; - i = 0; + link_idx = 0; + dai_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -248,10 +272,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) is_fe = (np != codec); ret = asoc_simple_card_dai_link_of(node, np, codec, priv, - i, is_fe, !loop); + &dai_idx, link_idx++, + is_fe, !loop); if (ret < 0) return ret; - i++; } node = of_get_next_child(top, node); } while (loop && node); @@ -342,6 +366,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; + struct asoc_simple_dai *dais; struct snd_soc_card *card; struct device *dev = &pdev->dev; int ret, i; @@ -358,7 +383,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -375,6 +401,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = simple_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: care multi DPCM codec_conf" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: care multi DPCM codec_conf has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0a26d6df4fd8aa8f4fb444446d162e01c3e349c0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 11 Dec 2018 03:25:37 +0000 Subject: [PATCH] ASoC: simple-scu-card: care multi DPCM codec_conf Current simple-scu-card didn't care about codec_conf for multi DPCM case. This patch cares it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 30 +++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 944a6f435f88..9d7299d536a8 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -21,17 +21,18 @@ struct simple_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; struct simple_dai_props { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) @@ -116,7 +117,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, struct device_node *np, struct device_node *codec, struct simple_card_data *priv, - int *dai_idx, int link_idx, int is_fe, + int *dai_idx, int link_idx, + int *conf_idx, int is_fe, bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); @@ -165,6 +167,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -177,6 +181,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, dai = dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); if (ret < 0) return ret; @@ -192,14 +199,12 @@ static int asoc_simple_card_dai_link_of(struct device_node *link, return ret; /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, + snd_soc_of_parse_audio_prefix(card, cconf, dai_link->codecs->of_node, PREFIX "prefix"); /* check "prefix" from each node if top doesn't have */ - if (!priv->codec_conf.of_node) - snd_soc_of_parse_node_prefix(np, - &priv->codec_conf, + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(np, cconf, dai_link->codecs->of_node, "prefix"); } @@ -238,7 +243,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct snd_soc_card *card = simple_priv_to_card(priv); bool is_fe; int ret, loop; - int dai_idx, link_idx; + int dai_idx, link_idx, conf_idx; if (!top) return -EINVAL; @@ -256,6 +261,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) loop = 1; link_idx = 0; dai_idx = 0; + conf_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { node = dev->of_node; @@ -273,6 +279,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) ret = asoc_simple_card_dai_link_of(node, np, codec, priv, &dai_idx, link_idx++, + &conf_idx, is_fe, !loop); if (ret < 0) return ret; @@ -368,6 +375,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; struct snd_soc_card *card; + struct snd_soc_codec_conf *cconf; struct device *dev = &pdev->dev; int ret, i; int lnum = 0, dnum = 0, cnum = 0; @@ -384,6 +392,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -402,6 +411,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = simple_priv_to_card(priv); @@ -409,8 +419,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = lnum; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_simple_card_parse_of(priv); if (ret < 0) { -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 0/7] ALSA: HD-audio display power fixes
by Takashi Iwai 11 Dec '18

11 Dec '18

Hi, this patch is about fixing possible HD-audio display power unbalance. Basically the "fix" is done by refactoring the whole relevant code. It starts from the Intel HD-audio runtime PM refactoring, followed by the display PM API change, and lots of code cleanups. Since it changes the display power API function, it hits both legacy and ASoC drivers. The patches are kept in topic/hda-pm-refactor branch of my sound git tree. This will be an immutable branch once when merged to for-next, so that it can be merged to ASoC tree if any conflicting change needs to be applied. Takashi === Takashi Iwai (7): ALSA: hda/intel: Refactoring PM code ALSA: hda: Refactor display power management ALSA: hda/intel: Drop superfluous AZX_DCAPS_I915_POWERWELL checks ALSA: hda/intel: Properly free the display power at error path ALSA: hda: Make snd_hdac_display_power() void function ASoC: hdac_hdmi: Add missing display power-off at driver removal ALSA: hda/hdmi: Always set display_power_control for Intel HSW+ codecs include/sound/hda_codec.h | 1 + include/sound/hda_component.h | 9 +- include/sound/hdaudio.h | 7 +- sound/hda/hdac_component.c | 39 ++++--- sound/hda/hdac_device.c | 17 --- sound/pci/hda/hda_codec.c | 16 ++- sound/pci/hda/hda_controller.c | 11 -- sound/pci/hda/hda_intel.c | 192 +++++++++++++-------------------- sound/pci/hda/patch_hdmi.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 22 ++-- sound/soc/intel/skylake/skl.c | 40 ++----- 11 files changed, 134 insertions(+), 228 deletions(-) -- 2.19.2

3 15

[alsa-devel] Realtek ALC294 doesn't work in 4.20-rc6
by Anthony Ettinger 11 Dec '18

11 Dec '18

Does anyone know when a fix will be applied? This is affecting Asus Zenbook UX433F -- Anthony Ettinger http://linkedin.com/in/chovy

2 1

[alsa-devel] [PATCH 0/8] ALSA: fireface: add support for Fireface 800 with MIDI functionality only
by Takashi Sakamoto 11 Dec '18

11 Dec '18

Hi, In 2004, RME GmbH shipped Fireface 800 as a flagship model of its Fireface series. This model consists of some chips: - TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus - TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus - Xilinx Spartan-3 FPGA XC3S400 - Xilinx High-Performance CPLD XC9572XL A remarkable point of this model is that its physical/link layer support S800 (800Mbps) mode of IEEE 1394b. Within devices of audio and music unit on IEEE 1394 bus, this is an unique model to support it, as long as I know. This commit adds support for the model with its MIDI functionality only. The functionality is quite similar to the one of Fireface 400. In patch 01-06, some codes specific to Fireface 400 are arranged to be shared by both models. On the other hand, there're some differences. The largest difference is that the model has no register to switch enabling/disabling tx asynchronous transactions for MIDI messages. Therefore the functionality work well without any assist by userspace. For this work, I use v2.77 firmware and v3.1.24 driver shipped by RME GmbH. If finding some differences between older firmware, please report it with enough information for debug. To help users for test, I prepare a remote branch including these patches. This can be backport to Linux v4.17 or later. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Takashi Sakamoto (8): ALSA: fireface: share some registers for status of clock synchronization ALSA: fireface: share status and configuration dump ALSA: fireface: share helper function to get current sampling rate and clock source ALSA: fireface: add support for second optical interface for ADAT stream ALSA: fireface: share register for async transaction of MIDI messages ALSA: fireface: add driver data for register for MIDI high address ALSA: fireface: localize a handler for MIDI messages on tx transaction ALSA: fireface: add support for Fireface 800 with MIDI functionality only sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 3 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 193 +++++++++++++- sound/firewire/fireface/ff-protocol-ff400.c | 282 ++------------------ sound/firewire/fireface/ff-protocol-ff800.c | 27 ++ sound/firewire/fireface/ff-stream.c | 2 +- sound/firewire/fireface/ff-transaction.c | 157 ++++++----- sound/firewire/fireface/ff.c | 54 +++- sound/firewire/fireface/ff.h | 31 ++- 10 files changed, 406 insertions(+), 346 deletions(-) create mode 100644 sound/firewire/fireface/ff-protocol-ff800.c -- 2.19.1

2 9

[alsa-devel] [PATCH] ALSA: hda/ca0132 - make pci_iounmap() call conditional
by Arnd Bergmann 11 Dec '18

11 Dec '18

When building without CONFIG_PCI, we can (depending on the architecture) get a link failure: ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined! Adding a compile-time check for PCI gets it to work correctly on 32-bit ARM. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c40cb6336017..e5bdbc245682 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8451,7 +8451,7 @@ static void ca0132_free(struct hda_codec *codec) ca0132_exit_chip(codec); snd_hda_power_down(codec); - if (spec->mem_base) + if (IS_ENABLED(CONFIG_PCI) && spec->mem_base) pci_iounmap(codec->bus->pci, spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); -- 2.20.0

2 1

[alsa-devel] [PATCH 0/7] ASoC: merge simple-scu-card into simple-card - step1
by Kuninori Morimoto 11 Dec '18

11 Dec '18

Hi Mark I'm posting "merge audio-graph-card" patch-set now. And, I decided to samethings to "simple-card". These are prepare for merging. Kuninori Morimoto (7): ASoC: simple-card: use cpu/codec pointer on simple_dai_props ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf sound/soc/generic/simple-card.c | 62 +++++---- sound/soc/generic/simple-scu-card.c | 261 +++++++++++++++++++++++++++--------- 2 files changed, 235 insertions(+), 88 deletions(-) -- 2.7.4

2 10

[alsa-devel] [PATCH 1/2] ASoC: Intel: bytcr_rt5640: Add quirk for the Prowise PT301 tablet
by Hans de Goede 11 Dec '18

11 Dec '18

Add a quirk for the Prowise PT301 tablet, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. Also it uses IN1 for its MIC and JD2 for jack-detect, rather then the default IN3 and JD1. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8587bd3d1cc1..6df902e3a380 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -674,6 +674,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { + /* Prowise PT301 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Prowise"), + DMI_MATCH(DMI_PRODUCT_NAME, "PT301"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), -- 2.19.2

3 5

[alsa-devel] Applied "ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f63196eaa83e033de523602324ff91288390a67 Mon Sep 17 00:00:00 2001 From: Zhuohao Lee <zhuohao(a)chromium.org> Date: Mon, 10 Dec 2018 12:32:13 +0800 Subject: [PATCH] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping >From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND. Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 58eb0fe69978..723a4935ed76 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: sdm845: add rt5663 codec select
by Arnd Bergmann 11 Dec '18

11 Dec '18

Calling into the codec driver adds a dependency on that being reachable from the module: ERROR: "rt5663_sel_asrc_clk_src" [sound/soc/qcom/snd-soc-sdm845.ko] undefined! Add the corresponding select statement, as it is done in the other user (Intel). Fixes: f7485875a687 ("ASoC: sdm845: Add configuration for headset codec") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 2a4c912d1e48..1b9fc0665792 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -100,6 +100,7 @@ config SND_SOC_SDM845 depends on QCOM_APR select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON + select SND_SOC_RT5663 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.20.0

2 1

[alsa-devel] [PATCH] ASoC: simple-card-utils: fix build warning without CONFIG_OF
by Arnd Bergmann 11 Dec '18

11 Dec '18

When CONFIG_OF is disabled, of_graph_parse_endpoint() does not initialize 'info', and gcc can see that: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_card_parse_graph_dai': sound/soc/generic/simple-card-utils.c:284:13: error: 'info.port' may be used uninitialized in this function [-Werror=maybe-uninitialized] It's probably best to check the return code anyway, and that also takes care of the warning. Fixes: b6f3fc005a2c ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/generic/simple-card-utils.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 6a31d07976b9..17d8aee43835 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -280,7 +280,10 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) * Non HDMI sound case, counting port/endpoint on its DT * is enough. Let's count it. */ - of_graph_parse_endpoint(ep, &info); + ret = of_graph_parse_endpoint(ep, &info); + if (ret) + return -ENXIO; + return info.port; } -- 2.20.0

3 2

[alsa-devel] Applied "ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dbd08fe59cf8c2f4bad0a72e912913c20db4774e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 4 Dec 2018 08:20:13 +0000 Subject: [PATCH] ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing Current simple-scu-card driver is parsing codec position for DPCM and consider DAI format. But, current operation is doing totally pointless, because it should be called for each CPU/Codec pair. Let's tidyup asoc_simple_card_parse_daifmt() timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 37 +++++++++++++++++------------ 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 656abe2015e1..24099e61d1b1 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -92,17 +92,24 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int asoc_simple_card_dai_link_of(struct device_node *np, +static int asoc_simple_card_dai_link_of(struct device_node *link, + struct device_node *np, + struct device_node *codec, struct simple_card_data *priv, - unsigned int daifmt, - int idx, bool is_fe) + int idx, bool is_fe, + bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct snd_soc_card *card = simple_priv_to_card(priv); + char *prefix = ""; int ret; + /* For single DAI link & old style of DT node */ + if (is_top_level_node) + prefix = PREFIX; + if (is_fe) { int is_single_links = 0; struct snd_soc_dai_link_component *codecs; @@ -178,7 +185,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - dai_link->dai_fmt = daifmt; + ret = asoc_simple_card_parse_daifmt(dev, link, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + return ret; + dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &asoc_simple_card_ops; @@ -191,10 +202,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + struct device_node *node = dev->of_node; struct device_node *np; + struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); - struct device_node *node = dev->of_node; - unsigned int daifmt = 0; bool is_fe; int ret, i; @@ -211,22 +222,18 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) asoc_simple_card_parse_convert(dev, node, PREFIX, &priv->adata); - /* find 1st codec */ - np = of_get_child_by_name(node, PREFIX "codec"); - if (!np) + i = 0; + codec = of_get_child_by_name(node, PREFIX "codec"); + if (!codec) return -ENODEV; - ret = asoc_simple_card_parse_daifmt(dev, node, np, PREFIX, &daifmt); - if (ret < 0) - return ret; - - i = 0; for_each_child_of_node(node, np) { is_fe = false; if (of_node_name_eq(np, PREFIX "cpu")) is_fe = true; - ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); + ret = asoc_simple_card_dai_link_of(node, np, codec, priv, + i, is_fe, true); if (ret < 0) return ret; i++; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH - AAF PCM plugin 0/7] Follow-up improvements
by Andre Guedes 10 Dec '18

10 Dec '18

Hi all, This patch series provides some follow-up improvements to the AVTP Audio Format (AAF) plugin. The highlight of this series is the implementation of a transmission offload mechanism which improves the plugin performance considerably. Details are provided in the following paragraphs. Currently, the AVTPDU transmission interval is controlled by the AAF plugin, in software. At every timer expiration, the plugin transmits one AVTPDU. This has some implications in terms of task scheduling. For instance, in class A streams, AVTPDUs are transmitted at every 125 us which means the application task should be scheduled-in at every 125 us. In a general purpose Linux system, such scheduling interval can be hard to cope with in the long-run. To mitigate that issue, this series introduces a transmission offload mechanism that leverages the SO_TXTIME sockopt and ETF qdisc features recently introduced to kernel 4.19. Instead of sending one AVTPDU at every timer expiration, the plugin sends several AVTPDUs at once to the kernel, configuring their Tx time so the transmission interval is maintained. The kernel can then offload packet transmission to the hardware (if the network controller supports it), providing more transmission time accuracy. This offloading mechanism enables the application to sleep for longer times, easing on task scheduling. To illustrate the improvements provided by this series, I ran a before-after experiment. The experiment setup consisted in 2 PCs with Intel i210 card connected back-to-back running an up-to-date Archlinux with kernel 4.19.4. I ran 'aplay' for 5 minutes on one PC and captured the AVTPDUs on the other PC. The metric under evaluation is the transmission interval and it is measured by checking the 'time_delta' information from ethernet frames captured at the receiving side. If you're interested in reproducing the experiments, let me know and I can share my helper scripts. The table below shows the experiment outcome for a Class A, stereo, 16-bit sample, 48 kHz stream. The unit is nanoseconds. | Mean | Stdev | Min | Max | Range | -------+--------+---------+---------+---------+---------+ Before | 125000 | 1154 | 75311 | 172144 | 96833 | After | 125000 | 18 | 124960 | 125048 | 88 | Before this patchset, the transmission interval mean is equal to the optimal value (Class A stream -> 125 us interval), and it is kept the same after the patchset. However, the dispersion measurements had improved considerably, meaning the system is consistently transmitting AVTPDUs at the correct interval. Finally, to help the review process, here follows a quick summary of the patches within this series: * PATCH 1: fixes a type in the plugin documentation. * PATCH 2: enables the user to configure the presentation time tolerance. * PATCH 3-5: refactor the code in order to land the offload mechanism code smoothly. * PATCH 6-7: implement the transmission offload mechanism. This series can also be found in my alsa-plugins tree in github [1]. Regards, Andre [1] https://github.com/aguedes/alsa-plugins Andre Guedes (7): doc: Fix typo in AAF doc aaf: Add presentation time tolerance aaf: Refactor AVTPDU transmission routines aaf: Refactor AVTPDU reception routines aaf: Refactor timeout routines aaf: Tx multiple AVTPDUs per media clock tick aaf: AVTPDU transmission periodicity aaf/pcm_aaf.c | 600 +++++++++++++++++++++++++++++++++----------------- configure.ac | 2 +- doc/aaf.txt | 40 +++- 3 files changed, 429 insertions(+), 213 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH v1] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND Change-Id: I67f929f226883bd289cb7beb0a7b23f40e7121e2 Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

3 4

[alsa-devel] [PATCH 0/2] ALSA: HDAudio: enable dynamic selection between legacy and Skylake drivers
by Pierre-Louis Bossart 10 Dec '18

10 Dec '18

This patchset depends on the "ASoC: Intel: Skylake: probe and Kconfig simplifications" series. Add the same logic based on PCI class/subclass/prog-if in the HDaudio legacy probe, and add a module parameter to bypass or force the automatic detection should it ever fail. This patchset was tested on a SKL device with the DSP disabled and a WHL device with the DSP enabled. All 6 combinations of module parameters provided the expected behavior. The SOF driver will use exactly the same mechanism, and add a mutual exclusion with the Skylake/SST driver to avoid having 3 drivers for the same PCI ID. Two is complicated enough. Pierre-Louis Bossart (2): ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection include/sound/hdaudio.h | 6 ++++ sound/pci/hda/Kconfig | 46 +++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 53 +++++++++++++++++++++++++++++----- sound/soc/intel/Kconfig | 6 ++++ sound/soc/intel/skylake/skl.c | 44 +++++++++++++++++++--------- 6 files changed, 135 insertions(+), 22 deletions(-) -- 2.17.1

2 12

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND. Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

2 1

[alsa-devel] [PATCH - snd_pcm_extplug_set_param_link() 1/1] pcm: extplug: Keep format and channels the same if requested
by twischer＠de.adit-jv.com 10 Dec '18

10 Dec '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this patch it is not possible to link the channel and format parameter if snd_pcm_extplug_set_param_*() or snd_pcm_extplug_set_slave_param_*() is called. Therefore the client and slave parameter can differ. So the extplug has to implement conversion. To avoid this the new snd_pcm_extplug_set_param_link() function can be called. As a reproduction sceanrio the following extplug source code can be used: static snd_pcm_sframes_t my_transfer(snd_pcm_extplug_t *e, const snd_pcm_channel_area_t *da, snd_pcm_uframes_t dof, const snd_pcm_channel_area_t *sa, snd_pcm_uframes_t sof, snd_pcm_uframes_t s) { return s; } static const snd_pcm_extplug_callback_t my_own_callback = { .transfer = my_transfer }; SND_PCM_PLUGIN_DEFINE_FUNC(my_plug) { snd_config_iterator_t i, next; snd_config_t *slave = NULL; snd_pcm_extplug_t *myplug; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; if (snd_config_get_id(n, &id) < 0) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0) continue; if (strcmp(id, "slave") == 0) { slave = n; continue; } return -EINVAL; } myplug = calloc(1, sizeof(*myplug)); myplug->version = SND_PCM_EXTPLUG_VERSION; myplug->callback = &my_own_callback; snd_pcm_extplug_create(myplug, name, root, slave, stream, mode); snd_pcm_extplug_set_param_minmax(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1, 16); // snd_pcm_extplug_set_param_link(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1); *pcmp = myplug->pcm; return 0; } SND_PCM_PLUGIN_SYMBOL(my_plug); To use this plugin the following ALSA configuration is required: pcm.myplug { type my_plug slave.pcm hw:Dummy } With this configuration without this patch snd_pcm_hw_params_get_channels_max() will always return 16 channels independent of the supported channels of the dummy device. Due to that for example the start up of JACK would fail: $ modprobe snd_dummy $ jackd -d alsa -P myplug ALSA: cannot set channel count to 16 for playback ALSA: cannot configure playback channel Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/include/pcm_extplug.h b/include/pcm_extplug.h index ced934f..e5c02d4 100644 --- a/include/pcm_extplug.h +++ b/include/pcm_extplug.h @@ -184,6 +184,8 @@ int snd_pcm_extplug_set_param_list(snd_pcm_extplug_t *extplug, int type, unsigne int snd_pcm_extplug_set_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max); int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, unsigned int num_list, const unsigned int *list); int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max); +int snd_pcm_extplug_set_param_link(snd_pcm_extplug_t *extplug, int type, + int keep_link); /** * set the parameter constraint with a single value diff --git a/src/pcm/pcm_ext_parm.h b/src/pcm/pcm_ext_parm.h index d25f2ab..7b99bef 100644 --- a/src/pcm/pcm_ext_parm.h +++ b/src/pcm/pcm_ext_parm.h @@ -5,6 +5,7 @@ struct snd_ext_parm { unsigned int *list; unsigned int active: 1; unsigned int integer: 1; + unsigned int keep_link: 1; }; static inline snd_mask_t *hw_param_mask(snd_pcm_hw_params_t *params, diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index 1f887c5..94002dc 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -249,7 +249,7 @@ static unsigned int get_links(struct snd_ext_parm *params) SND_PCM_HW_PARBIT_TICK_TIME); for (i = 0; i < SND_PCM_EXTPLUG_HW_PARAMS; i++) { - if (params[i].active) + if (params[i].active && !params[i].keep_link) links &= ~excl_parbits[i]; } return links; @@ -642,6 +642,17 @@ as former functions. To clear the parameter constraints, call #snd_pcm_extplug_params_reset() function. +When using snd_pcm_extplug_set_param_*() or snd_pcm_extplug_set_slave_param_*() +for any parameter. This parameter is no longer linked between the client and +slave PCM. Therefore it could differ and the extplug has to support conversion +between all valid parameter configurations. To keep the client and slave +parameter linked #snd_pcm_extplug_set_param_link() can be used for the +corresponding parameter. For example if the extplug does not support channel nor +format conversion the supported client parameters can be limited with +snd_pcm_extplug_set_param_*() and afterwards +#snd_pcm_extplug_set_param_link(ext, SND_PCM_EXTPLUG_HW_FORMAT, 1) and +#snd_pcm_extplug_set_param_link(ext, SND_PCM_EXTPLUG_HW_CHANNELS, 1) should be +called to keep the client and slave parameters the same. */ /** @@ -849,3 +860,26 @@ int snd_pcm_extplug_set_param_minmax(snd_pcm_extplug_t *extplug, int type, unsig return snd_ext_parm_set_minmax(&ext->params[type], min, max); } +/** + * @brief Keep the client and slave format/channels the same if requested. This + * is for example useful if this extplug does not support any channel + * conversion. + * @param extplug the extplug handle + * @param type parameter type + * @param keep_link if 1 the parameter identified by type will be kept the same + * for the client and slave PCM of this extplug + * @return 0 if successful, or a negative error code + */ +int snd_pcm_extplug_set_param_link(snd_pcm_extplug_t *extplug, int type, + int keep_link) +{ + extplug_priv_t *ext = extplug->pcm->private_data; + + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { + SNDERR("EXTPLUG: invalid parameter type %d", type); + return -EINVAL; + } + ext->params[type].keep_link = keep_link ? 1 : 0; + ext->sparams[type].keep_link = keep_link ? 1 : 0; + return 0; +} -- 2.7.4

2 1

Re: [alsa-devel] [PATCH 1/3] ALSA: hda/realtek: ALC294 mic and headset-mode fixups for ASUS X542UN
by Jian-Hong Pan 10 Dec '18

10 Dec '18

> Kailang <kailang(a)realtek.com> 於 2018年12月5日週三下午4:36寫道： > > > > Hi Jian-Hong, > > > > Could you test to change the model to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC? > > > > .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE ==> change to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC We do not have ASUS X542UN in hand right now, but we have ASUS X542UQ which goes with the same quirk as X542UN. I tested with ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, and it also works. So, I am going to send new patch set using ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC for ALC294. Thanks, Jian-Hong Pan

3 13

[alsa-devel] Patch for ALC294 Realtek soundcard in ASUS Zenbook UX433F?
by Anthony Ettinger 10 Dec '18

10 Dec '18

I saw something come across the mailing list earlier today, that supposedly fixes the no sound issues with this asus zenbook. Does anyone know when it will make it to the kernel master? I tried 4.20-rc6 but sound still doesn't work. -- Anthony Ettingers http://linkedin.com/in/chovy

1 0

[alsa-devel] [RFC PATCH v8 0/4] Media Device Allocator API
by shuah＠kernel.org 09 Dec '18

09 Dec '18

From: Shuah Khan <shuah(a)kernel.org> This patch series has been on the ice for the last couple of years. Mauro asked me to restart the discussion on this and see if we can make progress. I rebased the series to Linux 4.19. It has been surprisingly well preserved with one small merge conflict resolved in 0004. Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Please review. I am sending this as RFC even though it has been tested several times prior to this non-event rebased v8. I ran sanity tests. Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 +++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 320 +++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 692 insertions(+), 28 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.0

5 15

[alsa-devel] [PATCH] ALSA: fireface: fix reference to wrong register for clock configuration
by Takashi Sakamoto 09 Dec '18

09 Dec '18

In an initial commit, 'SYNC_STATUS' register is referred to get clock configuration, however this is wrong, according to my local note at hand for reverse-engineering about packet dump. It should be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config() is correctly programmed. This commit fixes the bug. Cc: <stable(a)vger.kernel.org> # v4.12+ Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-ff400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index 64c3cb0fb926..654a50319198 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -30,7 +30,7 @@ static int ff400_get_clock(struct snd_ff *ff, unsigned int *rate, int err; err = snd_fw_transaction(ff->unit, TCODE_READ_QUADLET_REQUEST, - FF400_SYNC_STATUS, &reg, sizeof(reg), 0); + FF400_CLOCK_CONFIG, &reg, sizeof(reg), 0); if (err < 0) return err; data = le32_to_cpu(reg); -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the mute LED regresion on Lenovo X1 Carbon
by Hui Wang 09 Dec '18

09 Dec '18

Users reported a mute LED regression on Lenovo X1 Carbon, the root cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE to this machine, then the machine can't apply the fixup of ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup together. Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56882d9b1e43..5b13d9e8d6c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6450,6 +6450,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_invalidate_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, [ALC295_FIXUP_HP_AUTO_MUTE] = { .type = HDA_FIXUP_FUNC, -- 2.17.1

2 1

[alsa-devel] [PATCH 0/6] Introduce audio-mute LED trigger (and conversions to it)
by Takashi Iwai 08 Dec '18

08 Dec '18

Hi, this is patch series I've hacked after useful conversation with Pavel. Basically this adds a new LED trigger audio-mute and audio-micmute, and convert the HD-audio driver and the platform drivers to use the LED trigger instead of the ugly direct dynamic symbol binding. The latest version of patches are found in topic/leds-trigger branch in my sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git As these are cross-tree patches, the branch above is based cleanly on v4.20-rc3, so that it can be merged well to multiple trees. Once after getting the ACK's, I'll add tags and fixate for merges. This patch series don't include huawei-wmi stuff; so Huawei patches need rework. I already have some piece of changes for huawei-wmi, so please ping me if needed. I checked briefly on my Dell laptop, and a Thinkpad model. Wider tests are appreciated, of course. thanks, Takashi === Takashi Iwai (6): leds: trigger: Introduce audio mute LED trigger platform/x86: dell-laptop: Add micmute LED trigger support platform/x86: thinkpad_acpi: Add audio mute LED classdev support ALSA: hda - Support led audio trigger platform/x86: dell-laptop: Drop superfluous exported function platform/x86: thinkpad_acpi: Drop superfluous exported function drivers/leds/trigger/Kconfig | 7 +++ drivers/leds/trigger/Makefile | 1 + drivers/leds/trigger/ledtrig-audio.c | 45 +++++++++++++++++++ drivers/platform/x86/Kconfig | 4 ++ drivers/platform/x86/dell-laptop.c | 27 +++++++++--- drivers/platform/x86/thinkpad_acpi.c | 66 +++++++++++++++++++++------- include/linux/dell-led.h | 7 --- include/linux/leds.h | 20 +++++++++ include/linux/thinkpad_acpi.h | 16 ------- sound/pci/hda/dell_wmi_helper.c | 48 -------------------- sound/pci/hda/hda_generic.c | 31 +++++++++++++ sound/pci/hda/hda_generic.h | 2 + sound/pci/hda/patch_realtek.c | 17 +++---- sound/pci/hda/thinkpad_helper.c | 43 +++--------------- 14 files changed, 194 insertions(+), 140 deletions(-) create mode 100644 drivers/leds/trigger/ledtrig-audio.c delete mode 100644 include/linux/dell-led.h delete mode 100644 include/linux/thinkpad_acpi.h delete mode 100644 sound/pci/hda/dell_wmi_helper.c -- 2.19.1

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Re: [alsa-devel] [PATCH v9 2/3] x86: add support for Huawei WMI hotkeys.
by ayman.bagabas＠gmail.com 08 Dec '18

08 Dec '18

On Fri, 2018-12-07 at 00:52 -0500, ayman.bagabas(a)gmail.com wrote: On Mon, 2018-12-03 at 21:17 +0200, Andy Shevchenko wrote: > On Mon, Dec 3, 2018 at 9:04 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Mon, 03 Dec 2018 19:53:39 +0100, > > Ayman Bagabas wrote: > > > + if (code == 0x80) { > > > + acpi_status status; > > > + acpi_handle handle; > > > + unsigned long long result; > > > + union acpi_object args[1]; > > > + struct acpi_object_list arg_list = { > > > + .pointer = args, > > > + .count = ARRAY_SIZE(args), > > > + }; > > > + > > > + args[0].type = ACPI_TYPE_INTEGER; > > > + args[0].integer.value = 0; > > > + > > > + status = acpi_get_handle(NULL, "\\WMI0", &handle); > > > + if (ACPI_FAILURE(status)) { > > > + dev_err(&wdev->dev, "Unable to get ACPI > > > handle\n"); > > > + return; > > > + } > > > + > > > + status = acpi_evaluate_integer(NULL, "WQ00", > > > &arg_list, &result); > > > > I guess you need to pass handle here? In the earlier version, you > > passed \\WMI0.WQ00, so it worked with NULL handle. But now it's no > > longer so... > > I think in this case we don't need to have a separate call to get > handle and try to get integer directly. In either we will have an > error case if method / namespace / etc is not found. > > I was digging further into the DSDT table of the laptop that has WQ00 method and it turned out that this method is a ACPI_WMI_EXPENSIVE. It has its own GUID of "39142400-C6A3-40fa-BADB-8A2652834100". I believe wmi_query_block, which is deprecated BTW, can be used to access this method instead of acpi_evaluate. However, I don't have a machine to test it on. Any ideas?

1 0

[alsa-devel] [PATCH - extplug without conversion 0/1] pcm: extplug: Use same format for slave
by twischer＠de.adit-jv.com 07 Dec '18

07 Dec '18

Hi Takashi, I have the following new idea for [PATCH - extplug refinement from hw_params 1/1] pcm: extplug: Update slave PCM if extplug slave params were changed May be you could give a short feedback. I will try to provide a reproduction scenario. Do you have an idea how to handle this additional function parameter without breaking the API. Best regards Timo

2 2

[alsa-devel] [PATCH - extplug refinement from hw_params 1/1] pcm: extplug: Update slave PCM if extplug slave params were changed
by twischer＠de.adit-jv.com 07 Dec '18

07 Dec '18

From: Timo Wischer <twischer(a)de.adit-jv.com> from extplug hw_params() callback function. This feature is for example required in the following use case: - A filter plugin supports only 1-8 channels. Therefore it calls snd_pcm_extplug_set_slave_param_minmax(SND_PCM_EXTPLUG_HW_CHANNELS, 1, 8) to provide the salve PCM parameters limited to 1-8 channels to the user application - The user application requests 2 channels. But in this case the slave PCM will be configured with 1 channel because it is the first valid configuration - To update the salve PCM snd_pcm_extplug_set_slave_param_minmax(SND_PCM_EXTPLUG_HW_CHANNELS, 2, 2) could be called from the extplug hw_params() callback function Without this patch the call to snd_pcm_extplug_set_slave_param_minmax() would not have any effect. With this patch the slave device will also be configured with 2 channels and no down mixing has to be done by the extplug. This new feature can also be used for some specific dependencies. For example a down mixing extplug supports only 2to1 and 4to2 channel down mixing. Therefore the salve PCM has to be opened with 2 channels if the user application requests 4 channels. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index 1f887c5..44afadb 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -26,6 +26,7 @@ * */ +#include <stdbool.h> #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_extplug.h" @@ -43,6 +44,7 @@ typedef struct snd_pcm_extplug_priv { snd_pcm_extplug_t *data; struct snd_ext_parm params[SND_PCM_EXTPLUG_HW_PARAMS]; struct snd_ext_parm sparams[SND_PCM_EXTPLUG_HW_PARAMS]; + bool sparams_changed; } extplug_priv_t; static const int hw_params_type[SND_PCM_EXTPLUG_HW_PARAMS] = { @@ -60,18 +62,49 @@ static const unsigned int excl_parbits[SND_PCM_EXTPLUG_HW_PARAMS] = { SND_PCM_HW_PARBIT_FRAME_BITS), }; +static bool snd_ext_parm_equal(const struct snd_ext_parm * const a, + const struct snd_ext_parm * const b) +{ + if (a != b || a->active != b->active || a->num_list != b->num_list) + return false; + + if (a->num_list > 0) { + for (unsigned int i = 0; i++; i < a->num_list) { + if (a->list[i] != b->list[i]) + return false; + } + } else { + if (a->min != b->min || a->max != b->max) + return false; + } + + return true; +} + +static int snd_ext_parm_set(struct snd_ext_parm *parm, unsigned int min, + unsigned int max, unsigned int num_list, + unsigned int *list) +{ + const struct snd_ext_parm new_parm = { + .min = min, + .max = max, + .num_list = num_list, + .list = list, + .active = 1, + }; + const int changed = snd_ext_parm_equal(parm, &new_parm) ? 0 : 1; + + free(parm->list); + *parm = new_parm; + return changed; +} + /* * set min/max values for the given parameter */ int snd_ext_parm_set_minmax(struct snd_ext_parm *parm, unsigned int min, unsigned int max) { - parm->num_list = 0; - free(parm->list); - parm->list = NULL; - parm->min = min; - parm->max = max; - parm->active = 1; - return 0; + return snd_ext_parm_set(parm, min, max, 0, NULL); } /* @@ -92,11 +125,7 @@ int snd_ext_parm_set_list(struct snd_ext_parm *parm, unsigned int num_list, cons memcpy(new_list, list, sizeof(*new_list) * num_list); qsort(new_list, num_list, sizeof(*new_list), val_compar); - free(parm->list); - parm->num_list = num_list; - parm->list = new_list; - parm->active = 1; - return 0; + return snd_ext_parm_set(parm, 0, 0, num_list, new_list); } void snd_ext_parm_clear(struct snd_ext_parm *parm) @@ -291,29 +320,37 @@ static int snd_pcm_extplug_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params */ static int snd_pcm_extplug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { - extplug_priv_t *ext = pcm->private_data; snd_pcm_t *slave = ext->plug.gen.slave; - int err = snd_pcm_hw_params_slave(pcm, params, - snd_pcm_extplug_hw_refine_cchange, - snd_pcm_extplug_hw_refine_sprepare, - snd_pcm_extplug_hw_refine_schange, - snd_pcm_generic_hw_params); - if (err < 0) - return err; - ext->data->slave_format = slave->format; - ext->data->slave_subformat = slave->subformat; - ext->data->slave_channels = slave->channels; - ext->data->rate = slave->rate; - INTERNAL(snd_pcm_hw_params_get_format)(params, &ext->data->format); - INTERNAL(snd_pcm_hw_params_get_subformat)(params, &ext->data->subformat); - INTERNAL(snd_pcm_hw_params_get_channels)(params, &ext->data->channels); - - if (ext->data->callback->hw_params) { - err = ext->data->callback->hw_params(ext->data, params); + + do { + ext->sparams_changed = false; + + int err = snd_pcm_hw_params_slave(pcm, params, + snd_pcm_extplug_hw_refine_cchange, + snd_pcm_extplug_hw_refine_sprepare, + snd_pcm_extplug_hw_refine_schange, + snd_pcm_generic_hw_params); if (err < 0) return err; - } + ext->data->slave_format = slave->format; + ext->data->slave_subformat = slave->subformat; + ext->data->slave_channels = slave->channels; + ext->data->rate = slave->rate; + INTERNAL(snd_pcm_hw_params_get_format)( + params, &ext->data->format); + INTERNAL(snd_pcm_hw_params_get_subformat)( + params, &ext->data->subformat); + INTERNAL(snd_pcm_hw_params_get_channels)( + params, &ext->data->channels); + + if (ext->data->callback->hw_params) { + err = ext->data->callback->hw_params(ext->data, params); + if (err < 0) + return err; + } + } while (ext->sparams_changed); + return 0; } @@ -411,6 +448,7 @@ static void clear_ext_params(extplug_priv_t *ext) int i; for (i = 0; i < SND_PCM_EXTPLUG_HW_PARAMS; i++) { snd_ext_parm_clear(&ext->params[i]); + ext->sparams_changed |= ext->sparams[i].active; snd_ext_parm_clear(&ext->sparams[i]); } } @@ -637,7 +675,9 @@ parameters are sample format and channels. To define the constraints of the slave PCM configuration, use either #snd_pcm_extplug_set_slave_param_minmax() and #snd_pcm_extplug_set_slave_param_list(). The arguments are as same -as former functions. +as former functions. Both functions can also be called from the hw_params +callback. This can be used to further limit the configuration of the slave +device depending on the configuration of the user device. To clear the parameter constraints, call #snd_pcm_extplug_params_reset() function. @@ -768,11 +808,17 @@ void snd_pcm_extplug_params_reset(snd_pcm_extplug_t *extplug) int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, unsigned int num_list, const unsigned int *list) { extplug_priv_t *ext = extplug->pcm->private_data; + int changed = 0; + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; } - return snd_ext_parm_set_list(&ext->sparams[type], num_list, list); + changed = snd_ext_parm_set_list(&ext->sparams[type], num_list, list); + if (changed > 0) + ext->sparams_changed = true; + + return changed; } /** @@ -790,6 +836,8 @@ int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, u int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max) { extplug_priv_t *ext = extplug->pcm->private_data; + int changed = 0; + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; @@ -798,7 +846,11 @@ int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; } - return snd_ext_parm_set_minmax(&ext->sparams[type], min, max); + changed = snd_ext_parm_set_minmax(&ext->sparams[type], min, max); + if (changed > 0) + ext->sparams_changed = true; + + return changed; } /** -- 2.7.4

3 4

[alsa-devel] Applied "ASoC: max9867: Remove useless assignment" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Remove useless assignment has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8efc1afd7e0570d9bb37f5127f0742304a8ad2fc Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:18:17 +0100 Subject: [PATCH] ASoC: max9867: Remove useless assignment ret is assigned later, no need to initialize it. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 1cda54b59854..1b155f40b106 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -476,7 +476,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) @@ -497,10 +497,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2 0/5] max9867 driver update
by Ladislav Michl 07 Dec '18

07 Dec '18

Hi Mark, this is yet another round of driver update. Tested on AT91SAM9G20 based board with codec in master mode. All implemeted controls do work now. Comments from previous serie taken into account. Ladislav Michl (5): ASoC: max9867: Remove useless assignment ASoC: max9867: Fix power management ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix signal paths ASoC: max9867: Add copyright and module author

2 9

[alsa-devel] Applied "ASoC: max9867: Calculate LRCLK divider" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Calculate LRCLK divider has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 715ee1916b95bfd0773487397a0354ad9a4660e6 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:04 +0100 Subject: [PATCH] ASoC: max9867: Calculate LRCLK divider Drop "Common NI Values Table" and calculate LRCLK divider, then add allowed rate constraints based on master clock frequency. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 207 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 31 ++---- 2 files changed, 110 insertions(+), 128 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index c454de434d76..faec094161c1 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -126,124 +126,106 @@ static const struct snd_soc_dapm_route max9867_audio_map[] = { {"LINE_IN", NULL, "Right Line"}, }; -enum rates { - pcm_rate_8, pcm_rate_16, pcm_rate_24, - pcm_rate_32, pcm_rate_44, - pcm_rate_48, max_pcm_rate, +static const unsigned int max9867_rates_44k1[] = { + 11025, 22050, 44100, }; -static const struct ni_div_rates { - u32 mclk; - u16 ni[max_pcm_rate]; -} ni_div[] = { - {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, - {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, - {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, - {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, - {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, - {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, - {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, - {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +static const struct snd_pcm_hw_constraint_list max9867_constraints_44k1 = { + .list = max9867_rates_44k1, + .count = ARRAY_SIZE(max9867_rates_44k1), }; -static inline int get_ni_value(int mclk, int rate) +static const unsigned int max9867_rates_48k[] = { + 8000, 16000, 32000, 48000, +}; + +static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { + .list = max9867_rates_48k, + .count = ARRAY_SIZE(max9867_rates_48k), +}; + +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + +static int max9867_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - int i, ret = 0; + struct max9867_priv *max9867 = + snd_soc_component_get_drvdata(dai->component); - /* find the closest rate index*/ - for (i = 0; i < ARRAY_SIZE(ni_div); i++) { - if (ni_div[i].mclk >= mclk) - break; - } - if (i == ARRAY_SIZE(ni_div)) - return -EINVAL; + if (max9867->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, max9867->constraints); - switch (rate) { - case 8000: - return ni_div[i].ni[pcm_rate_8]; - case 16000: - return ni_div[i].ni[pcm_rate_16]; - case 32000: - return ni_div[i].ni[pcm_rate_32]; - case 44100: - return ni_div[i].ni[pcm_rate_44]; - case 48000: - return ni_div[i].ni[pcm_rate_48]; - default: - pr_err("%s wrong rate %d\n", __func__, rate); - ret = -EINVAL; - } - return ret; + return 0; } static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + int value; + unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - unsigned int ni_h, ni_l; - int value; + unsigned int ni = DIV_ROUND_CLOSEST_ULL(96ULL * 0x10000 * params_rate(params), + max9867->pclk); - value = get_ni_value(max9867->sysclk, params_rate(params)); - if (value < 0) - return value; - - ni_h = (0xFF00 & value) >> 8; - ni_l = 0x00FF & value; /* set up the ni value */ regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_NI_HIGH_MASK, ni_h); + MAX9867_NI_HIGH_MASK, (0xFF00 & ni) >> 8); regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_NI_LOW_MASK, ni_l); - if (!max9867->master) { - /* - * digital pll locks on to any externally supplied LRCLK signal - * and also enable rapid lock mode. - */ - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_PLL, MAX9867_PLL); - } else { - unsigned long int bclk_rate, pclk_bclk_ratio; - int bclk_value; - - bclk_rate = params_rate(params) * 2 * params_width(params); - pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (params_width(params)) { - case 8: - case 16: - switch (pclk_bclk_ratio) { - case 2: - bclk_value = MAX9867_IFC1B_PCLK_2; - break; - case 4: - bclk_value = MAX9867_IFC1B_PCLK_4; - break; + MAX9867_NI_LOW_MASK, 0x00FF & ni); + if (max9867->master) { + if (max9867->dsp_a) { + value = MAX9867_IFC1B_48X; + } else { + rate = params_rate(params) * 2 * params_width(params); + ratio = max9867->pclk / rate; + switch (params_width(params)) { case 8: - bclk_value = MAX9867_IFC1B_PCLK_8; - break; case 16: - bclk_value = MAX9867_IFC1B_PCLK_16; + switch (ratio) { + case 2: + value = MAX9867_IFC1B_PCLK_2; + break; + case 4: + value = MAX9867_IFC1B_PCLK_4; + break; + case 8: + value = MAX9867_IFC1B_PCLK_8; + break; + case 16: + value = MAX9867_IFC1B_PCLK_16; + break; + default: + return -EINVAL; + } + break; + case 24: + value = MAX9867_IFC1B_48X; + break; + case 32: + value = MAX9867_IFC1B_64X; break; default: - dev_err(component->dev, - "unsupported sampling rate\n"); return -EINVAL; } - break; - case 24: - bclk_value = MAX9867_IFC1B_24BIT; - break; - case 32: - bclk_value = MAX9867_IFC1B_32BIT; - break; - default: - dev_err(component->dev, "unsupported sampling rate\n"); - return -EINVAL; } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, - MAX9867_IFC1B_BCLK_MASK, bclk_value); + MAX9867_IFC1B_BCLK_MASK, value); + } else { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); } return 0; } @@ -285,8 +267,16 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, freq); return -EINVAL; } - value = value << MAX9867_PSCLK_SHIFT; + if (freq % 48000 == 0) + max9867->constraints = &max9867_constraints_48k; + else if (freq % 44100 == 0) + max9867->constraints = &max9867_constraints_44k1; + else + dev_warn(component->dev, + "Unable to set exact rate with %uHz clock frequency\n", + freq); max9867->sysclk = freq; + value = value << MAX9867_PSCLK_SHIFT; /* exact integer mode is not supported */ value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, @@ -299,16 +289,17 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - u8 iface1A = 0, iface1B = 0; + u8 iface1A, iface1B; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - max9867->master = 1; - iface1A |= MAX9867_MASTER; + max9867->master = true; + iface1A = MAX9867_MASTER; + iface1B = MAX9867_IFC1B_48X; break; case SND_SOC_DAIFMT_CBS_CFS: - max9867->master = 0; - iface1A &= ~MAX9867_MASTER; + max9867->master = false; + iface1A = iface1B = 0; break; default: return -EINVAL; @@ -316,9 +307,11 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + max9867->dsp_a = false; iface1A |= MAX9867_I2S_DLY; break; case SND_SOC_DAIFMT_DSP_A: + max9867->dsp_a = true; iface1A |= MAX9867_TDM_MODE | MAX9867_SDOUT_HIZ; break; default: @@ -344,20 +337,18 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; } static const struct snd_soc_dai_ops max9867_dai_ops = { - .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, + .set_fmt = max9867_dai_set_fmt, .digital_mute = max9867_mute, - .hw_params = max9867_dai_hw_params, + .startup = max9867_startup, + .hw_params = max9867_dai_hw_params, }; -#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) - static struct snd_soc_dai_driver max9867_dai[] = { { .name = "max9867-aif1", @@ -365,15 +356,15 @@ static struct snd_soc_dai_driver max9867_dai[] = { .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &max9867_dai_ops, .symmetric_rates = 1, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index d9170850c96e..7f037ab701a5 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -26,13 +26,11 @@ #define MAX9867_PSCLK_10_20 0x1 #define MAX9867_PSCLK_20_40 0x2 #define MAX9867_PSCLK_40_60 0x3 -#define MAX9867_AUDIOCLKHIGH 0x06 -#define MAX9867_NI_HIGH_WIDTH 0x7 -#define MAX9867_NI_HIGH_MASK 0x7F -#define MAX9867_NI_LOW_MASK 0x7F -#define MAX9867_NI_LOW_SHIFT 0x1 -#define MAX9867_PLL (1<<7) -#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0xFE +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 #define MAX9867_RAPID_LOCK 0x01 #define MAX9867_IFC1A 0x08 #define MAX9867_MASTER (1<<7) @@ -43,12 +41,12 @@ #define MAX9867_BCI_MODE (1<<5) #define MAX9867_IFC1B 0x09 #define MAX9867_IFC1B_BCLK_MASK 7 -#define MAX9867_IFC1B_32BIT 0x01 -#define MAX9867_IFC1B_24BIT 0x02 -#define MAX9867_IFC1B_PCLK_2 4 -#define MAX9867_IFC1B_PCLK_4 5 -#define MAX9867_IFC1B_PCLK_8 6 -#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_IFC1B_64X 0x01 +#define MAX9867_IFC1B_48X 0x02 +#define MAX9867_IFC1B_PCLK_2 0x04 +#define MAX9867_IFC1B_PCLK_4 0x05 +#define MAX9867_IFC1B_PCLK_8 0x06 +#define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a #define MAX9867_DACGAIN 0x0b #define MAX9867_DACLEVEL 0x0c @@ -72,11 +70,4 @@ #define MAX9867_CACHEREGNUM 10 -/* codec private data */ -struct max9867_priv { - struct regmap *regmap; - unsigned int sysclk; - unsigned int pclk; - unsigned int master; -}; #endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max9867: Fix signal paths" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Fix signal paths has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bc2610a6c4c92f0bf6bbf4c6641407536f30daaf Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:33 +0100 Subject: [PATCH] ASoC: max9867: Fix signal paths Sound capture and line bypass currently do not work as well as some mixer controls. Fix that by building proper audio paths and adjusting volume controls to match datasheet. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 188 ++++++++++++++++++++----------------- sound/soc/codecs/max9867.h | 8 +- 2 files changed, 102 insertions(+), 94 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index faec094161c1..7a08e2c70e17 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -23,107 +23,124 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), +}; + +/* Output mixer */ +static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Line Bypass Switch", + MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* Line out switch */ +static const struct snd_kcontrol_new max9867_line_out_control = + SOC_DAPM_DOUBLE_R("Switch", + MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); -static const struct snd_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + max9867_input_mixer_controls, + ARRAY_SIZE(max9867_input_mixer_controls)), + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + + SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, + max9867_sidetone_mixer_controls, + ARRAY_SIZE(max9867_sidetone_mixer_controls)), + SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", SND_SOC_NOPM, 0, 0, + max9867_output_mixer_controls, + ARRAY_SIZE(max9867_output_mixer_controls)), + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, + &max9867_line_out_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { - {"Left DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, }; static const unsigned int max9867_rates_44k1[] = { @@ -235,13 +252,8 @@ static int max9867_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -464,8 +476,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 7f037ab701a5..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -48,20 +48,16 @@ #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2] ASoC: max9867: Add copyright and module author
by Ladislav Michl 07 Dec '18

07 Dec '18

Driver rewritten, assign copyright notice and change module author as original one remains silent and I want to be notified about bugs. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: -v2: convert entire comment block to C++ style sound/soc/codecs/max9867.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 7a08e2c70e17..8600c5439e1e 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -1,12 +1,10 @@ -/* - * max9867.c -- max9867 ALSA SoC Audio driver - * - * Copyright 2013-15 Maxim Integrated Products - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// MAX9867 ALSA SoC codec driver +// +// Copyright 2013-2015 Maxim Integrated Products +// Copyright 2018 Ladislav Michl <ladis(a)linux-mips.org> +// #include <linux/delay.h> #include <linux/i2c.h> @@ -551,6 +549,6 @@ static struct i2c_driver max9867_i2c_driver = { module_i2c_driver(max9867_i2c_driver); -MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_AUTHOR("Ladislav Michl <ladis(a)linux-mips.org>"); +MODULE_DESCRIPTION("ASoC MAX9867 driver"); MODULE_LICENSE("GPL"); -- 2.20.0.rc2

2 1

Re: [alsa-devel] HDA headset button support
by Takashi Iwai 07 Dec '18

07 Dec '18

On Wed, 28 Nov 2018 13:03:58 +0100, Kailang wrote: > > Hi Takashi, > > >I'd rather like to extend snd_hda_jack_add_kctl() with the key > >support (maybe adding a new function). You can pass there the > >combination of SND_JACK_BTN_X with KEY_XXX in an array. > > Do you mean create new function like below? > snd_hda_jack_add_key_kctl() > > It couldn't use snd_hda_jack_add_kctl() directly. > Because it had report pin state and get type from NID. > > type = get_input_jack_type(codec, nid); > ..... > ..... > state = snd_hda_jack_detect(codec, nid); > snd_jack_report(jack->jack, state ? jack->type : 0); > > So, to create a new function in hda_jack.c was better. Well, this seems more deeper than I thought. Below is my quick attempt to add the infrastructure. This adds more flexibility but still easiness for other possible button implementations, hopefully. Could you check the series below? thanks, Takashi

2 10

[alsa-devel] [PATCH 0/4] alsa-utils: axfer: add help texts
by GitHub pull_request - opened 07 Dec '18

07 Dec '18

alsa-project/alsa-utils pull request #7 was opened from takaswie: This patchset adds help texts for command entry, list and transfer subcommands. In this time, contents for each backend of transfer subcommand is not added yet. Unfortunately, execution of aliases (arecord/aplay) can print help text for command text only. This bug will be fixed in future commits. Additionally, due to program design, execution of the aliases cannot print help text of list subcommand. This is unavoidable loss of backward compatibility. ``` Takashi Sakamoto (4): axfer: print help text of command entry axfer: print help text of list subcommand axfer: print help text of transfer subcommand just for common options axfer: enable each backend to print own help axfer/main.c | 12 +++++++++++- axfer/subcmd-list.c | 9 ++++++++- axfer/xfer-libasound.c | 6 ++++++ axfer/xfer-libffado.c | 6 ++++++ axfer/xfer-options.c | 35 +++++++++++++++++++++++++++++++++++ axfer/xfer.c | 5 +++++ axfer/xfer.h | 2 ++ 7 files changed, 73 insertions(+), 2 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/7 Patch URL : https://github.com/alsa-project/alsa-utils/pull/7.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] ALC700 headphone no sound issue
by Kailang 07 Dec '18

07 Dec '18

Hi Takashi, In Intel ENG board with ALC700. It had headphone no sound issue. If it plugged headphone or headset into the jack, then do the reboot, it will have a chance to cause headphone no sound. It just need to run the headphone mode procedure after boot time. The issue will be fixed. It also suitable for ALC234 ALC274 and ALC294. BR, Kailang

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[alsa-devel] [GIT PULL] soundwire updates
by Vinod Koul 07 Dec '18

07 Dec '18

Hey Greg, Please pull to receive a single commit for soundwire tree for this cycle. Am off next week so sending this bit early The following changes since commit 651022382c7f8da46cb4872a545ee1da6d097d2a: Linux 4.20-rc1 (2018-11-04 15:37:52 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-4.21-rc1 for you to fetch changes up to b1635596860dd3229645e41276f6c83ca788e4c1: soundwire: intel: constify snd_soc_dai_ops structures (2018-11-12 15:36:06 +0530) ---------------------------------------------------------------- soundwire updates for 4.21-rc1 A single change: - constify snd_soc_dai_ops struct by Julia ---------------------------------------------------------------- Julia Lawall (1): soundwire: intel: constify snd_soc_dai_ops structures drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- ~Vinod

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[alsa-devel] [PATCH] tinycompress: Fix missing '}' around extern C
by Charles Keepax 07 Dec '18

07 Dec '18

From: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/tinycompress/tinycompress.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 780dce6..f0fae30 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -276,4 +276,8 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} // extern "C" +#endif + #endif -- 2.11.0

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[alsa-devel] [PATCH 1/3] ASoC: rt5660: Add a new ACPI match ID
by Hui Wang 07 Dec '18

07 Dec '18

The Realtek codec ALC3277 is 100% compatible with the codec RT5660 in I2S mode. And on the Dell IoT platform, the codec is ALC3277, and the HID of the codec in the BIOS is 10EC3277, so adding this ID to the ACPI match table. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/rt5660.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 27f7445b2432..e74b2e8cd423 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1246,6 +1246,7 @@ MODULE_DEVICE_TABLE(of, rt5660_of_match); static const struct acpi_device_id rt5660_acpi_match[] = { { "10EC5660", 0 }, + { "10EC3277", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); -- 2.17.1

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[alsa-devel] [PATCH v8 0/7] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 07 Dec '18

07 Dec '18

v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine v8 - drop merged patches - add regulator property for simple amplifier, make enable-gpio optional - add regulator to speaker amp, introduce headphone amp Vasily Khoruzhick (7): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget ASoC: dt-bindings: make enable-gpio optional for simple amplifier ASoC: simple-amplifier: make enable-gpio optional arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../bindings/sound/simple-amplifier.txt | 5 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 35 ++++++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 53 ++++++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 35 ++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ sound/soc/codecs/simple-amplifier.c | 7 ++- 6 files changed, 188 insertions(+), 1 deletion(-) -- 2.19.1

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[alsa-devel] [PATCH v2] ASoC: max9867: Remove useless assignment
by Ladislav Michl 06 Dec '18

06 Dec '18

ret is assigned later, no need to initialize it. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: -v2: add missing Signed-off-by (I'm sorry for that...) sound/soc/codecs/max9867.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 1cda54b59854..1b155f40b106 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -476,7 +476,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) @@ -497,10 +497,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } -- 2.20.0.rc2

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Gnawty
by Hans de Goede 06 Dec '18

06 Dec '18

The Gnawty model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3 for the mclk, just like the Clapper and Swanky models. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Gnawty that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201787 Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Jaime Pérez <19.jaime.91(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index ad0c98383853..08a5152e635a 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -396,6 +396,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Gnawty"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { -- 2.19.2

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[alsa-devel] [PATCH 1/2] ASoC: use DMA addr rather than CPU pa for acp_audio_dma
by Yu Zhao 06 Dec '18

06 Dec '18

We shouldn't assume CPU physical address we get from page_to_phys() is same as DMA address we get from dma_alloc_coherent(). On x86_64, we won't run into any problem with the assumption when dma_ops is nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled. And it's most likely different from CPU physical address when AMD IOMMU is not in passthrough mode. Signed-off-by: Yu Zhao <yuzhao(a)google.com> --- sound/soc/amd/acp-pcm-dma.c | 15 +++++---------- sound/soc/amd/acp.h | 2 +- 2 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cdebab2f8ce5..fd3db4c37882 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -303,11 +303,10 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, } /* Create page table entries in ACP SRAM for the allocated memory */ -static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, +static void acp_pte_config(void __iomem *acp_mmio, dma_addr_t addr, u16 num_of_pages, u32 pte_offset) { u16 page_idx; - u64 addr; u32 low; u32 high; u32 offset; @@ -317,7 +316,6 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); @@ -333,7 +331,7 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, acp_reg_write(high, acp_mmio, mmACP_SRBM_Targ_Idx_Data); /* Move to next physically contiguos page */ - pg++; + addr += PAGE_SIZE; } } @@ -343,7 +341,7 @@ static void config_acp_dma(void __iomem *acp_mmio, { u16 ch_acp_sysmem, ch_acp_i2s; - acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + acp_pte_config(acp_mmio, rtd->dma_addr, rtd->num_of_pages, rtd->pte_offset); if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { @@ -850,7 +848,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, int status; uint64_t size; u32 val = 0; - struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -986,16 +983,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return status; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - pg = virt_to_page(substream->dma_buffer.area); - if (pg) { + if (substream->dma_buffer.area) { acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->pg = pg; + rtd->dma_addr = substream->dma_buffer.addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ - rtd->pg = pg; rtd->size = size; rtd->num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; rtd->direction = substream->stream; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index dbbb1a85638d..e5ab6c6040a6 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -123,7 +123,7 @@ enum acp_dma_priority_level { }; struct audio_substream_data { - struct page *pg; + dma_addr_t dma_addr; unsigned int order; u16 num_of_pages; u16 i2s_instance; -- 2.20.0.rc1.387.gf8505762e3-goog

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[alsa-devel] [PATCH] ASoC: Use of_node_name_eq for node name comparisons
by Rob Herring 06 Dec '18

06 Dec '18

Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. For the FSL ASoC card, the full node names appear to be "ssi", "esai", and "sai", so there's not any reason to use strstr and of_node_name_eq can be used instead. Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 6 +++--- sound/soc/generic/simple-scu-card.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 44433b20435c..81f2fe2c6d23 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -571,17 +571,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } /* Common settings for corresponding Freescale CPU DAI driver */ - if (strstr(cpu_np->name, "ssi")) { + if (of_node_name_eq(cpu_np, "ssi")) { /* Only SSI needs to configure AUDMUX */ ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(&pdev->dev, "failed to init audmux\n"); goto asrc_fail; } - } else if (strstr(cpu_np->name, "esai")) { + } else if (of_node_name_eq(cpu_np, "esai")) { priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; - } else if (strstr(cpu_np->name, "sai")) { + } else if (of_node_name_eq(cpu_np, "sai")) { priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 85b46f0eae0f..e9a1acffcf5b 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -216,7 +216,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) i = 0; for_each_child_of_node(node, np) { is_fe = false; - if (strcmp(np->name, PREFIX "cpu") == 0) + if (of_node_name_eq(np, PREFIX "cpu")) is_fe = true; ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); -- 2.19.1

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[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: cs4270: use 'reset-gpios' rather than 'reset-gpio'
by Daniel Mack 06 Dec '18

06 Dec '18

Bindings should use 'reset-gpios', not 'reset-gpio'. The driver needs to switch to the gpiod consume API to handle this correctly. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- Documentation/devicetree/bindings/sound/cs4270.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt index 6b222f9b8ef5..c33770ec4c3c 100644 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -10,8 +10,8 @@ Required properties: Optional properties: - - reset-gpio : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the codec starts. + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. Example: -- 2.19.2

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[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: ak4104: use 'reset-gpios' rather than 'reset-gpio'
by Daniel Mack 06 Dec '18

06 Dec '18

Bindings should use 'reset-gpios', not 'reset-gpio'. The driver needs to switch to the gpiod consume API to handle this correctly. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- Documentation/devicetree/bindings/sound/ak4104.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ak4104.txt b/Documentation/devicetree/bindings/sound/ak4104.txt index deca5e18f304..ae5f7f057dc3 100644 --- a/Documentation/devicetree/bindings/sound/ak4104.txt +++ b/Documentation/devicetree/bindings/sound/ak4104.txt @@ -12,8 +12,8 @@ Required properties: Optional properties: - - reset-gpio : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the device starts. + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the device starts. Example: -- 2.19.2

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[alsa-devel] Applied "ASoC: max9867: Calculate LRCLK divider" to the asoc tree
by Mark Brown 06 Dec '18

06 Dec '18

The patch ASoC: max9867: Calculate LRCLK divider has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 127f8a2145957fb53f0d65444720a583b54a58f3 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:04 +0100 Subject: [PATCH] ASoC: max9867: Calculate LRCLK divider Drop "Common NI Values Table" and calculate LRCLK divider, then add allowed rate constraints based on master clock frequency. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 207 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 31 ++---- 2 files changed, 110 insertions(+), 128 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index c454de434d76..faec094161c1 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -126,124 +126,106 @@ static const struct snd_soc_dapm_route max9867_audio_map[] = { {"LINE_IN", NULL, "Right Line"}, }; -enum rates { - pcm_rate_8, pcm_rate_16, pcm_rate_24, - pcm_rate_32, pcm_rate_44, - pcm_rate_48, max_pcm_rate, +static const unsigned int max9867_rates_44k1[] = { + 11025, 22050, 44100, }; -static const struct ni_div_rates { - u32 mclk; - u16 ni[max_pcm_rate]; -} ni_div[] = { - {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, - {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, - {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, - {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, - {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, - {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, - {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, - {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +static const struct snd_pcm_hw_constraint_list max9867_constraints_44k1 = { + .list = max9867_rates_44k1, + .count = ARRAY_SIZE(max9867_rates_44k1), }; -static inline int get_ni_value(int mclk, int rate) +static const unsigned int max9867_rates_48k[] = { + 8000, 16000, 32000, 48000, +}; + +static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { + .list = max9867_rates_48k, + .count = ARRAY_SIZE(max9867_rates_48k), +}; + +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + +static int max9867_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - int i, ret = 0; + struct max9867_priv *max9867 = + snd_soc_component_get_drvdata(dai->component); - /* find the closest rate index*/ - for (i = 0; i < ARRAY_SIZE(ni_div); i++) { - if (ni_div[i].mclk >= mclk) - break; - } - if (i == ARRAY_SIZE(ni_div)) - return -EINVAL; + if (max9867->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, max9867->constraints); - switch (rate) { - case 8000: - return ni_div[i].ni[pcm_rate_8]; - case 16000: - return ni_div[i].ni[pcm_rate_16]; - case 32000: - return ni_div[i].ni[pcm_rate_32]; - case 44100: - return ni_div[i].ni[pcm_rate_44]; - case 48000: - return ni_div[i].ni[pcm_rate_48]; - default: - pr_err("%s wrong rate %d\n", __func__, rate); - ret = -EINVAL; - } - return ret; + return 0; } static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + int value; + unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - unsigned int ni_h, ni_l; - int value; + unsigned int ni = DIV_ROUND_CLOSEST_ULL(96ULL * 0x10000 * params_rate(params), + max9867->pclk); - value = get_ni_value(max9867->sysclk, params_rate(params)); - if (value < 0) - return value; - - ni_h = (0xFF00 & value) >> 8; - ni_l = 0x00FF & value; /* set up the ni value */ regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_NI_HIGH_MASK, ni_h); + MAX9867_NI_HIGH_MASK, (0xFF00 & ni) >> 8); regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_NI_LOW_MASK, ni_l); - if (!max9867->master) { - /* - * digital pll locks on to any externally supplied LRCLK signal - * and also enable rapid lock mode. - */ - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_PLL, MAX9867_PLL); - } else { - unsigned long int bclk_rate, pclk_bclk_ratio; - int bclk_value; - - bclk_rate = params_rate(params) * 2 * params_width(params); - pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (params_width(params)) { - case 8: - case 16: - switch (pclk_bclk_ratio) { - case 2: - bclk_value = MAX9867_IFC1B_PCLK_2; - break; - case 4: - bclk_value = MAX9867_IFC1B_PCLK_4; - break; + MAX9867_NI_LOW_MASK, 0x00FF & ni); + if (max9867->master) { + if (max9867->dsp_a) { + value = MAX9867_IFC1B_48X; + } else { + rate = params_rate(params) * 2 * params_width(params); + ratio = max9867->pclk / rate; + switch (params_width(params)) { case 8: - bclk_value = MAX9867_IFC1B_PCLK_8; - break; case 16: - bclk_value = MAX9867_IFC1B_PCLK_16; + switch (ratio) { + case 2: + value = MAX9867_IFC1B_PCLK_2; + break; + case 4: + value = MAX9867_IFC1B_PCLK_4; + break; + case 8: + value = MAX9867_IFC1B_PCLK_8; + break; + case 16: + value = MAX9867_IFC1B_PCLK_16; + break; + default: + return -EINVAL; + } + break; + case 24: + value = MAX9867_IFC1B_48X; + break; + case 32: + value = MAX9867_IFC1B_64X; break; default: - dev_err(component->dev, - "unsupported sampling rate\n"); return -EINVAL; } - break; - case 24: - bclk_value = MAX9867_IFC1B_24BIT; - break; - case 32: - bclk_value = MAX9867_IFC1B_32BIT; - break; - default: - dev_err(component->dev, "unsupported sampling rate\n"); - return -EINVAL; } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, - MAX9867_IFC1B_BCLK_MASK, bclk_value); + MAX9867_IFC1B_BCLK_MASK, value); + } else { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); } return 0; } @@ -285,8 +267,16 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, freq); return -EINVAL; } - value = value << MAX9867_PSCLK_SHIFT; + if (freq % 48000 == 0) + max9867->constraints = &max9867_constraints_48k; + else if (freq % 44100 == 0) + max9867->constraints = &max9867_constraints_44k1; + else + dev_warn(component->dev, + "Unable to set exact rate with %uHz clock frequency\n", + freq); max9867->sysclk = freq; + value = value << MAX9867_PSCLK_SHIFT; /* exact integer mode is not supported */ value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, @@ -299,16 +289,17 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - u8 iface1A = 0, iface1B = 0; + u8 iface1A, iface1B; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - max9867->master = 1; - iface1A |= MAX9867_MASTER; + max9867->master = true; + iface1A = MAX9867_MASTER; + iface1B = MAX9867_IFC1B_48X; break; case SND_SOC_DAIFMT_CBS_CFS: - max9867->master = 0; - iface1A &= ~MAX9867_MASTER; + max9867->master = false; + iface1A = iface1B = 0; break; default: return -EINVAL; @@ -316,9 +307,11 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + max9867->dsp_a = false; iface1A |= MAX9867_I2S_DLY; break; case SND_SOC_DAIFMT_DSP_A: + max9867->dsp_a = true; iface1A |= MAX9867_TDM_MODE | MAX9867_SDOUT_HIZ; break; default: @@ -344,20 +337,18 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; } static const struct snd_soc_dai_ops max9867_dai_ops = { - .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, + .set_fmt = max9867_dai_set_fmt, .digital_mute = max9867_mute, - .hw_params = max9867_dai_hw_params, + .startup = max9867_startup, + .hw_params = max9867_dai_hw_params, }; -#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) - static struct snd_soc_dai_driver max9867_dai[] = { { .name = "max9867-aif1", @@ -365,15 +356,15 @@ static struct snd_soc_dai_driver max9867_dai[] = { .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &max9867_dai_ops, .symmetric_rates = 1, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index d9170850c96e..7f037ab701a5 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -26,13 +26,11 @@ #define MAX9867_PSCLK_10_20 0x1 #define MAX9867_PSCLK_20_40 0x2 #define MAX9867_PSCLK_40_60 0x3 -#define MAX9867_AUDIOCLKHIGH 0x06 -#define MAX9867_NI_HIGH_WIDTH 0x7 -#define MAX9867_NI_HIGH_MASK 0x7F -#define MAX9867_NI_LOW_MASK 0x7F -#define MAX9867_NI_LOW_SHIFT 0x1 -#define MAX9867_PLL (1<<7) -#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0xFE +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 #define MAX9867_RAPID_LOCK 0x01 #define MAX9867_IFC1A 0x08 #define MAX9867_MASTER (1<<7) @@ -43,12 +41,12 @@ #define MAX9867_BCI_MODE (1<<5) #define MAX9867_IFC1B 0x09 #define MAX9867_IFC1B_BCLK_MASK 7 -#define MAX9867_IFC1B_32BIT 0x01 -#define MAX9867_IFC1B_24BIT 0x02 -#define MAX9867_IFC1B_PCLK_2 4 -#define MAX9867_IFC1B_PCLK_4 5 -#define MAX9867_IFC1B_PCLK_8 6 -#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_IFC1B_64X 0x01 +#define MAX9867_IFC1B_48X 0x02 +#define MAX9867_IFC1B_PCLK_2 0x04 +#define MAX9867_IFC1B_PCLK_4 0x05 +#define MAX9867_IFC1B_PCLK_8 0x06 +#define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a #define MAX9867_DACGAIN 0x0b #define MAX9867_DACLEVEL 0x0c @@ -72,11 +70,4 @@ #define MAX9867_CACHEREGNUM 10 -/* codec private data */ -struct max9867_priv { - struct regmap *regmap; - unsigned int sysclk; - unsigned int pclk; - unsigned int master; -}; #endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max9867: Fix signal paths" to the asoc tree
by Mark Brown 06 Dec '18

06 Dec '18

The patch ASoC: max9867: Fix signal paths has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2f388effac683ee34016f915f7ff1d50b3f9df81 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:33 +0100 Subject: [PATCH] ASoC: max9867: Fix signal paths Sound capture and line bypass currently do not work as well as some mixer controls. Fix that by building proper audio paths and adjusting volume controls to match datasheet. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 188 ++++++++++++++++++++----------------- sound/soc/codecs/max9867.h | 8 +- 2 files changed, 102 insertions(+), 94 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index faec094161c1..7a08e2c70e17 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -23,107 +23,124 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), +}; + +/* Output mixer */ +static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Line Bypass Switch", + MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* Line out switch */ +static const struct snd_kcontrol_new max9867_line_out_control = + SOC_DAPM_DOUBLE_R("Switch", + MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); -static const struct snd_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + max9867_input_mixer_controls, + ARRAY_SIZE(max9867_input_mixer_controls)), + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + + SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, + max9867_sidetone_mixer_controls, + ARRAY_SIZE(max9867_sidetone_mixer_controls)), + SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", SND_SOC_NOPM, 0, 0, + max9867_output_mixer_controls, + ARRAY_SIZE(max9867_output_mixer_controls)), + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, + &max9867_line_out_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { - {"Left DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, }; static const unsigned int max9867_rates_44k1[] = { @@ -235,13 +252,8 @@ static int max9867_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -464,8 +476,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 7f037ab701a5..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -48,20 +48,16 @@ #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 1/2] ASoC: Fix function return type
by Agrawal, Akshu 06 Dec '18

06 Dec '18

Function returns -1 on error and the return type should be int. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- include/sound/soc.h | 2 +- sound/soc/soc-io.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de8..591d743 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1341,7 +1341,7 @@ static inline int snd_soc_component_cache_sync( /* component IO */ int snd_soc_component_read(struct snd_soc_component *component, unsigned int reg, unsigned int *val); -unsigned int snd_soc_component_read32(struct snd_soc_component *component, +int snd_soc_component_read32(struct snd_soc_component *component, unsigned int reg); int snd_soc_component_write(struct snd_soc_component *component, unsigned int reg, unsigned int val); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 1ff9175..d08f5b1 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -38,7 +38,7 @@ int snd_soc_component_read(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_read); -unsigned int snd_soc_component_read32(struct snd_soc_component *component, +int snd_soc_component_read32(struct snd_soc_component *component, unsigned int reg) { unsigned int val; -- 1.9.1

4 7

[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by Niklas Carlsson 06 Dec '18

06 Dec '18

DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

2 2

[alsa-devel] [PATCH] MAINTAINERS: add ASoC maintainers for sound dt-bindings
by Clément Péron 06 Dec '18

06 Dec '18

Sound dt-bindings are applied by ASoC maintainers and should be submit to ASoC list in addition to the devicetree list. Hence, add this information into the MAINTAINERS file. Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 0abecc528dac..62701e0135af 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13930,6 +13930,7 @@ S: Supported F: Documentation/devicetree/bindings/sound/ F: Documentation/sound/soc/ F: sound/soc/ +F: include/dt-bindings/sound/ F: include/sound/soc* SOUNDWIRE SUBSYSTEM -- 2.19.1

2 2

[alsa-devel] [PATCH v2] ASoC: imx-audmux: complete dt-bindings for i.MX6
by Clément Péron 06 Dec '18

06 Dec '18

From: Colin Didier <colin.didier(a)devialet.com> The i.MX6 audmux differs from i.MX51. This patch adds the audmux for i.MX6 family. Signed-off-by: Colin Didier <colin.didier(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v2: Repost with ASoC maintainer and list include/dt-bindings/sound/fsl-imx-audmux.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 15f138bebe16..a1d0741d9ed1 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -25,6 +25,14 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +#define MX6_AUDMUX_PORT1_SSI1 0 +#define MX6_AUDMUX_PORT2_SSI2 1 +#define MX6_AUDMUX_PORT3 2 +#define MX6_AUDMUX_PORT4 3 +#define MX6_AUDMUX_PORT5 4 +#define MX6_AUDMUX_PORT6 5 +#define MX6_AUDMUX_PORT7_SSI3 6 + /* * TFCSEL/RFCSEL (i.MX27) or TFSEL/TCSEL/RFSEL/RCSEL (i.MX31/51/53/6Q) * can be sourced from Rx/Tx. -- 2.19.1

1 1

[alsa-devel] [PATCH] ALSA: Use of_node_name_eq for node name comparisons
by Rob Herring 06 Dec '18

06 Dec '18

Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. A couple of open coded iterating thru the child node names are converted to use for_each_child_of_node() instead. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: Paul Mackerras <paulus(a)samba.org> Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/fabrics/layout.c | 2 +- sound/aoa/soundbus/i2sbus/core.c | 6 +++--- sound/ppc/pmac.c | 4 ++-- sound/ppc/tumbler.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3a23fd032611..8797d42e2b76 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -776,7 +776,7 @@ static int check_codec(struct aoa_codec *codec, struct codec_connection *cc; /* if the codec has a 'codec' node, we require a reference */ - if (codec->node && (strcmp(codec->node->name, "codec") == 0)) { + if (of_node_name_eq(codec->node, "codec")) { snprintf(propname, sizeof(propname), "platform-%s-codec-ref", codec->name); ref = of_get_property(ldev->sound, propname, NULL); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index bd7c5029fc59..c3f57a3fb1a5 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -154,7 +154,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *np) { struct i2sbus_dev *dev; - struct device_node *child = NULL, *sound = NULL; + struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; char node_name[6]; @@ -177,8 +177,8 @@ static int i2sbus_add_dev(struct macio_dev *macio, return 0; i = 0; - while ((child = of_get_next_child(np, child))) { - if (strcmp(child->name, "sound") == 0) { + for_each_child_of_node(np, child) { + if (of_node_name_eq(child, "sound")) { i++; sound = child; } diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 48dd44f8e914..d692e4070167 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -908,7 +908,7 @@ static void detect_byte_swap(struct snd_pmac *chip) /* if seems that Keylargo can't byte-swap */ for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { if (of_device_is_compatible(mio, "Keylargo")) chip->can_byte_swap = 0; break; @@ -1313,7 +1313,7 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) } else if (chip->is_pbook_G3) { struct device_node* mio; for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { struct resource r; if (of_address_to_resource(mio, 0, &r) == 0) chip->macio_base = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 0779a2912237..6d7ffffcce95 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1365,8 +1365,8 @@ int snd_pmac_tumbler_init(struct snd_pmac *chip) mix->anded_reset = 0; mix->reset_on_sleep = 1; - for (np = chip->node->child; np; np = np->sibling) { - if (!strcmp(np->name, "sound")) { + for_each_child_of_node(chip->node, np) { + if (of_node_name_eq(np, "sound")) { if (of_get_property(np, "has-anded-reset", NULL)) mix->anded_reset = 1; if (of_get_property(np, "layout-id", NULL)) -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: soundbus: Remove direct OF name and type accesses
by Rob Herring 06 Dec '18

06 Dec '18

Convert soundbus uevent and sysfs OF node name and device type usage to use printf specifier and helper functions instead of directly accessing the name and type pointers. This will allow the eventual removal of the pointers. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/soundbus/core.c | 2 +- sound/aoa/soundbus/sysfs.c | 34 +++++++++++++++++++++------------- 2 files changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index f3076a3f7f2f..065d3a55725e 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -74,7 +74,7 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) of = &soundbus_dev->ofdev; /* stuff we want to pass to /sbin/hotplug */ - retval = add_uevent_var(env, "OF_NAME=%s", of->dev.of_node->name); + retval = add_uevent_var(env, "OF_NAME=%pOFn", of->dev.of_node); if (retval) return retval; diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c index 81da020bddef..a2d55e15afbb 100644 --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -1,18 +1,10 @@ // SPDX-License-Identifier: GPL-2.0 #include <linux/kernel.h> +#include <linux/of.h> #include <linux/stat.h> /* FIX UP */ #include "soundbus.h" -#define soundbus_config_of_attr(field, format_string) \ -static ssize_t \ -field##_show (struct device *dev, struct device_attribute *attr, \ - char *buf) \ -{ \ - struct soundbus_dev *mdev = to_soundbus_device (dev); \ - return sprintf (buf, format_string, mdev->ofdev.dev.of_node->field); \ -} - static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, char *buf) { @@ -25,17 +17,33 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, strcat(buf, "\n"); length = strlen(buf); } else { - length = sprintf(buf, "of:N%sT%s\n", - of->dev.of_node->name, of->dev.of_node->type); + length = sprintf(buf, "of:N%pOFn%c%s\n", + of->dev.of_node, 'T', + of_node_get_device_type(of->dev.of_node)); } return length; } static DEVICE_ATTR_RO(modalias); -soundbus_config_of_attr (name, "%s\n"); +static ssize_t name_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%pOFn\n", of->dev.of_node); +} static DEVICE_ATTR_RO(name); -soundbus_config_of_attr (type, "%s\n"); + +static ssize_t type_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%s\n", of_node_get_device_type(of->dev.of_node)); +} static DEVICE_ATTR_RO(type); struct attribute *soundbus_dev_attrs[] = { -- 2.19.1

2 1

[alsa-devel] Invalid TLS certificate for mailman.alsa-project.org
by Antonio Ospite 06 Dec '18

06 Dec '18

Hi, when logging in to mailman.alsa-project.org[1] to change my subscription options I noticed that the TLS certificate used for HTTPS connections does not cover said domain, firefox says: ----------------------------------------------------------------------- mailman.alsa-project.org uses an invalid security certificate. The certificate is only valid for the following names: alsa-project.org, alsa0.perex.cz, bugtrack.alsa-project.org, git.alsa-project.org, mail.alsa-project.org, www.alsa-project.org Error code: SSL_ERROR_BAD_CERT_DOMAIN ----------------------------------------------------------------------- Basically, mailman.alsa-project.org is not in Subject Alternative Names for the certificate. This is not a huge deal as: 1. mail.alsa-project.org works fine with HTTPS; 2. the password is sent by email in the monthly reminder anyways. However it would still be handy to have this fixed as mailman.alsa-project.org URLs are used in the Wiki[2] pages and in the footer of messages coming from the mailing list, and it can be surprising for users that HTTPS does not work. Thank you, Antonio [1] https://mailman.alsa-project.org/mailman/options/alsa-devel [2] https://www.alsa-project.org/main/index.php/Mailing-lists -- Antonio Ospite https://ao2.it https://twitter.com/ao2it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

1 0

[alsa-devel] [PATCH linux-next v3 6/7] ASoC: rsnd: add avb clocks
by Jiada Wang 06 Dec '18

06 Dec '18

There are AVB Counter Clocks in ADG, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch registers 8 AVB Counter Clocks when clock-cells of rcar_sound node is 2, Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/adg.c | 316 +++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 9 ++ sound/soc/sh/rcar/rsnd.h | 9 ++ 3 files changed, 325 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66588eb..0708be1f38d9 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -5,6 +5,7 @@ // Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/clk-provider.h> +#include <linux/of_address.h> #include "rsnd.h" #define CLKA 0 @@ -21,13 +22,30 @@ #define BRRx_MASK(x) (0x3FF & x) +#define AVB_CLK_NUM 8 +#define AVB_MAX_RATE 25000000 +#define AVB_DIV_EN_COM BIT(31) +#define AVB_MAX_DIV 0x3ffc0 + static struct rsnd_mod_ops adg_ops = { .name = "adg", }; +struct clk_avb { + struct clk_hw hw; + unsigned int idx; + struct rsnd_adg *adg; + /* lock reg access */ + spinlock_t *lock; +}; + +#define hw_to_avb(_hw) container_of(_hw, struct clk_avb, hw) + struct rsnd_adg { struct clk *clk[CLKMAX]; struct clk *clkout[CLKOUTMAX]; + struct clk *clkavb[AVB_CLK_NUM]; + struct clk *clkadg; struct clk_onecell_data onecell; struct rsnd_mod mod; u32 flags; @@ -37,6 +55,7 @@ struct rsnd_adg { int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ + spinlock_t avb_lock; }; #define LRCLK_ASYNC (1 << 0) @@ -408,6 +427,248 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) } } +static struct clk *rsnd_adg_clk_src_twocell_get(struct of_phandle_args *clkspec, + void *data) +{ + unsigned int clktype = clkspec->args[0]; + unsigned int clkidx = clkspec->args[1]; + struct rsnd_adg *adg = data; + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + struct rsnd_priv *priv = rsnd_mod_to_priv(adg_mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + + switch (clktype) { + case 0: + if (clkidx >= CLKOUTMAX) { + dev_err(dev, "Invalid fixed clock index %u\n", clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkout[clkidx]; + break; + case 1: + if (clkidx >= AVB_CLK_NUM) { + dev_err(dev, "Invalid avb clock index %u\n", clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkavb[clkidx]; + break; + default: + dev_err(dev, "Invalid ADG clock type %u\n", clktype); + return ERR_PTR(-EINVAL); + } + + return clk; +} + +static void clk_avb_div_write(struct rsnd_adg *adg, u32 data, unsigned int idx) +{ + struct rsnd_mod *mod = rsnd_mod_get(adg); + + switch (idx) { + case 0: + rsnd_mod_write(mod, AVB_CLK_DIV0, data); + break; + case 1: + rsnd_mod_write(mod, AVB_CLK_DIV1, data); + break; + case 2: + rsnd_mod_write(mod, AVB_CLK_DIV2, data); + break; + case 3: + rsnd_mod_write(mod, AVB_CLK_DIV3, data); + break; + case 4: + rsnd_mod_write(mod, AVB_CLK_DIV4, data); + break; + case 5: + rsnd_mod_write(mod, AVB_CLK_DIV5, data); + break; + case 6: + rsnd_mod_write(mod, AVB_CLK_DIV6, data); + break; + case 7: + rsnd_mod_write(mod, AVB_CLK_DIV7, data); + break; + } +} + +static u32 clk_avb_div_read(struct rsnd_adg *adg, unsigned int idx) +{ + struct rsnd_mod *mod = rsnd_mod_get(adg); + u32 val = 0; + + switch (idx) { + case 0: + val = rsnd_mod_read(mod, AVB_CLK_DIV0); + break; + case 1: + val = rsnd_mod_read(mod, AVB_CLK_DIV1); + break; + case 2: + val = rsnd_mod_read(mod, AVB_CLK_DIV2); + break; + case 3: + val = rsnd_mod_read(mod, AVB_CLK_DIV3); + break; + case 4: + val = rsnd_mod_read(mod, AVB_CLK_DIV4); + break; + case 5: + val = rsnd_mod_read(mod, AVB_CLK_DIV5); + break; + case 6: + val = rsnd_mod_read(mod, AVB_CLK_DIV6); + break; + case 7: + val = rsnd_mod_read(mod, AVB_CLK_DIV7); + break; + } + + return val; +} + +static int clk_avb_is_enabled(struct clk_hw *hw) +{ + struct clk_avb *avb = hw_to_avb(hw); + struct rsnd_mod *mod = rsnd_mod_get(avb->adg); + + return rsnd_mod_read(mod, AVB_CLK_CONFIG) & BIT(avb->idx); +} + +static int clk_avb_enabledisable(struct clk_hw *hw, int enable) +{ + struct clk_avb *avb = hw_to_avb(hw); + u32 val; + struct rsnd_mod *mod = rsnd_mod_get(avb->adg); + + spin_lock(avb->lock); + + val = rsnd_mod_read(mod, AVB_CLK_CONFIG); + if (enable) + val |= BIT(avb->idx); + else + val &= ~BIT(avb->idx); + rsnd_mod_write(mod, AVB_CLK_CONFIG, val); + + spin_unlock(avb->lock); + + return 0; +} + +static int clk_avb_enable(struct clk_hw *hw) +{ + return clk_avb_enabledisable(hw, 1); +} + +static void clk_avb_disable(struct clk_hw *hw) +{ + clk_avb_enabledisable(hw, 0); +} + +static unsigned long clk_avb_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_avb *avb = hw_to_avb(hw); + u32 div; + + div = clk_avb_div_read(avb->adg, avb->idx); + if (!div) + return parent_rate; + + return parent_rate * 32 / div; +} + +static unsigned int clk_avb_calc_div(unsigned long rate, + unsigned long parent_rate) +{ + unsigned int div; + + if (!rate) + rate = 1; + + if (rate > AVB_MAX_RATE) + rate = AVB_MAX_RATE; + + div = DIV_ROUND_CLOSEST(parent_rate * 32, rate); + + if (div > AVB_MAX_DIV) + div = AVB_MAX_DIV; + + return div; +} + +static long clk_avb_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned int div = clk_avb_calc_div(rate, *parent_rate); + + return (*parent_rate * 32) / div; +} + +static int clk_avb_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_avb *avb = hw_to_avb(hw); + unsigned int div = clk_avb_calc_div(rate, parent_rate); + u32 val; + + val = clk_avb_div_read(avb->adg, avb->idx); + clk_avb_div_write(avb->adg, val | div, avb->idx); + + return 0; +} + +static const struct clk_ops clk_avb_ops = { + .enable = clk_avb_enable, + .disable = clk_avb_disable, + .is_enabled = clk_avb_is_enabled, + .recalc_rate = clk_avb_recalc_rate, + .round_rate = clk_avb_round_rate, + .set_rate = clk_avb_set_rate, +}; + +static struct clk *clk_register_avb(struct device *dev, struct rsnd_adg *adg, + unsigned int id, spinlock_t *lock) +{ + struct clk_init_data init; + struct clk_avb *avb; + struct clk *clk; + char name[8]; + const char *parent_name; + + if (IS_ERR(adg->clkadg)) + return ERR_PTR(-ENODEV); + + avb = devm_kzalloc(dev, sizeof(*avb), GFP_KERNEL); + if (!avb) + return ERR_PTR(-ENOMEM); + + parent_name = __clk_get_name(adg->clkadg); + + snprintf(name, sizeof(name), "%s%u", "avb", id); + + avb->idx = id; + avb->lock = lock; + avb->adg = adg; + + /* Register the clock. */ + init.name = name; + init.ops = &clk_avb_ops; + init.flags = CLK_IS_BASIC; + init.parent_names = &parent_name; + init.num_parents = 1; + + avb->hw.init = &init; + + /* init DIV to a valid state */ + clk_avb_div_write(avb->adg, AVB_MAX_DIV, avb->idx); + + clk = devm_clk_register(dev, &avb->hw); + + return clk; +} + static void rsnd_adg_get_clkin(struct rsnd_priv *priv, struct rsnd_adg *adg) { @@ -436,6 +697,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, unsigned long req_48kHz_rate, req_441kHz_rate; int i, req_size; const char *parent_clk_name = NULL; + struct rsnd_mod *mod = rsnd_mod_get(adg); static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", [CLKOUT1] = "audio_clkout1", @@ -540,21 +802,23 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ of_property_read_u32(np, "#clock-cells", &count); - /* - * for clkout - */ - if (!count) { + + switch (count) { + case 0: + /* + * for clkout + */ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], 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); } - } - /* - * for clkout0/1/2/3 - */ - else { + break; + case 1: + /* + * for clkout0/1/2/3 + */ for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, @@ -566,6 +830,33 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->onecell.clk_num = CLKOUTMAX; of_clk_add_provider(np, of_clk_src_onecell_get, &adg->onecell); + break; + case 2: + /* + * for clkout0/1/2/3 and avb clocks + */ + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, 0, + req_rate[0]); + if (!IS_ERR(clk)) + adg->clkout[i] = clk; + } + + for (i = 0; i < AVB_CLK_NUM; i++) { + clk = clk_register_avb(dev, adg, i, &adg->avb_lock); + if (!IS_ERR(clk)) + adg->clkavb[i] = clk; + } + + of_clk_add_provider(np, rsnd_adg_clk_src_twocell_get, adg); + + rsnd_mod_write(mod, AVB_CLK_CONFIG, AVB_DIV_EN_COM); + + break; + default: + dev_err(dev, "Invalid clock-cell %d\n", count); + break; } rsnd_adg_get_clkout_end: @@ -606,17 +897,24 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) return -ENOMEM; + spin_lock_init(&adg->avb_lock); + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, 0, 0); if (ret) return ret; + clk = devm_clk_get(dev, "adg"); + if (!IS_ERR(clk)) + adg->clkadg = clk; + rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); rsnd_adg_clk_dbg_info(priv, adg); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca639404f2cd..1b000d03b76e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -355,6 +355,15 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + RSND_GEN_S_REG(AVB_CLK_DIV0, 0x11c), + RSND_GEN_S_REG(AVB_CLK_DIV1, 0x120), + RSND_GEN_S_REG(AVB_CLK_DIV2, 0x124), + RSND_GEN_S_REG(AVB_CLK_DIV3, 0x128), + RSND_GEN_S_REG(AVB_CLK_DIV4, 0x12c), + RSND_GEN_S_REG(AVB_CLK_DIV5, 0x130), + RSND_GEN_S_REG(AVB_CLK_DIV6, 0x134), + RSND_GEN_S_REG(AVB_CLK_DIV7, 0x138), + RSND_GEN_S_REG(AVB_CLK_CONFIG, 0x13c), }; static const struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129af6d1..d31b8a65985f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -148,6 +148,15 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AVB_CLK_DIV0, + RSND_REG_AVB_CLK_DIV1, + RSND_REG_AVB_CLK_DIV2, + RSND_REG_AVB_CLK_DIV3, + RSND_REG_AVB_CLK_DIV4, + RSND_REG_AVB_CLK_DIV5, + RSND_REG_AVB_CLK_DIV6, + RSND_REG_AVB_CLK_DIV7, + RSND_REG_AVB_CLK_CONFIG, /* SSIU */ RSND_REG_SSI_MODE, -- 2.19.2

2 6

[alsa-devel] alsa-utils: axfer: add manuals
by GitHub pull_request - opened 05 Dec '18

05 Dec '18

alsa-project/alsa-utils pull request #6 was opened from takaswie: alsa-utils: axfer: add manuals This patchset adds manuals. As a result, users can refer to documents by man(1): ``` $ man 1 axfer $ man 1 axfer-list $ man 1 axfer-transfer ``` The manuals are written according to groff_man(7) and man-pages(7), however in some points they don't follow the guideline; e.g. usage of hyphen-minus ('-') without backslash '\' in pre-formatted text. I'll welcome any fixes. In this time, description of each backend in transfer subcommand is not yet added. It will be added in later work. ``` Takashi Sakamoto (5): axfer: add a manual for entry point axfer: add a manual for list subcommand axfer: add a manual for transfer subcommand axfer: add a section to describe compatibility to aplay(1) axfer: add a section about design of transfer subcommand axfer/Makefile.am | 10 + axfer/axfer-list.1 | 86 +++++++ axfer/axfer-transfer.1 | 500 +++++++++++++++++++++++++++++++++++++++++ axfer/axfer.1 | 132 +++++++++++ 4 files changed, 728 insertions(+) create mode 100644 axfer/axfer-list.1 create mode 100644 axfer/axfer-transfer.1 create mode 100644 axfer/axfer.1 ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/6 Patch URL : https://github.com/alsa-project/alsa-utils/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 05 Dec '18

05 Dec '18

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- include/sound/cs35l36.h | 58 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36-tables.c | 315 +++++++ sound/soc/codecs/cs35l36.c | 1736 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 451 ++++++++++ 6 files changed, 2567 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36-tables.c create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..6061e40 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,58 @@ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright (c) 2018 Cirrus Logic Inc. + * + * 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 __CS35L36_H +#define __CS35L36_H + +/* INT/GPIO pin configuration */ +struct irq_cfg { + bool is_present; + int irq_drv_sel; + int irq_pol; + int irq_gpio_sel; + int irq_out_en; + int irq_src_sel; +}; + +struct vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool sclk_frc; + bool lrclk_frc; + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_gain_zc; + bool amp_pcm_inv; + bool pdm_ldm_exit; + bool pdm_ldm_enter; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + struct vpbr_cfg vpbr_config; + struct irq_cfg irq_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f18..77cb861 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -465,6 +466,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c..7a6009a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o cs35l36-tables.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -310,6 +311,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36-tables.c b/sound/soc/codecs/cs35l36-tables.c new file mode 100644 index 0000000..4a93182 --- /dev/null +++ b/sound/soc/codecs/cs35l36-tables.c @@ -0,0 +1,315 @@ +/* + * cs35l36-tables.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include "cs35l36.h" + +struct reg_default cs35l36_reg[CS35L36_MAX_CACHE_REG] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; +EXPORT_SYMBOL_GPL(cs35l36_reg); + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} +EXPORT_SYMBOL_GPL(cs35l36_readable_reg); + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} +EXPORT_SYMBOL_GPL(cs35l36_volatile_reg); diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..4fa1531 --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1736 @@ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int prev_clksrc; + int extclk_freq; + int extclk_cfg; + int fll_igain; + int sclk; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; + struct completion global_pup_done; + struct completion global_pdn_done; +}; + +struct cs35l36_pll_sysclk_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_sysclk_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, + CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("AMP PCM Gain", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + if (reg & CS35L36_PLL_UNLOCK_MASK) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + return ret; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + } + return 0; +} +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux[] = { + SOC_DAPM_ENUM("Input Mux", chansel_enum), +}; + +static const char * const cs35l36_boost_text[] = { + "On", + "Off", +}; + +static SOC_ENUM_SINGLE_DECL(boost_enum, SND_SOC_NOPM, 0, + cs35l36_boost_text); + +static const struct snd_kcontrol_new cs35l36_boost_mux[] = { + SOC_DAPM_ENUM("Boost Enable", boost_enum), +}; + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", + "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, + CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, + CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, + CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, + CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, + CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, + CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, + asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_MUX_E("BOOST Mux", SND_SOC_NOPM, 0, 0, cs35l36_boost_mux, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Mux", "On", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Mux"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + int i; + unsigned int global_fs = params_rate(params); + unsigned int asp_width; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1_val = 0; + int fs2_val = 0; + + /* Need the SCLK Frequency regardless of sysclk source */ + cs35l36->sclk = freq; + + if (cs35l36->sclk > 6000000) { + fs1_val = 3 * 4 + 4; + fs2_val = 8 * 4 + 4; + } + + if (cs35l36->sclk <= 6000000) { + fs1_val = 3 * ((24000000 + cs35l36->sclk - 1) / cs35l36->sclk) + + 4; + fs2_val = 5 * ((24000000 + cs35l36->sclk - 1) / cs35l36->sclk) + + 4; + } + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK, fs1_val); + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS2_WINDOW_MASK, fs2_val << + CS35L36_FS2_WINDOW_SHIFT); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static int cs35l36_get_clk_config(struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) { + cs35l36->extclk_cfg = cs35l36_pll_sysclk[i].clk_cfg; + cs35l36->fll_igain = cs35l36_pll_sysclk[i].fll_igain; + return i; + } + } + + return -EINVAL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + if (!substream->runtime) + return 0; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret; + + cs35l36->extclk_freq = freq; + cs35l36->prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + ret = cs35l36_get_clk_config(cs35l36, freq); + + if (ret < 0) { + dev_err(component->dev, + "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + cs35l36->extclk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + cs35l36->fll_igain); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc != CS35L36_PLLSRC_PDMCLK) + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + } + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc != CS35L36_PLLSRC_PDMCLK) { + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + } + } else { + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc == CS35L36_PLLSRC_PDMCLK) + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + } + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + if (cs35l36->pdata.ldm_mode_sel) { + if (cs35l36->prev_clksrc == CS35L36_PLLSRC_PDMCLK) { + regmap_update_bits(cs35l36->regmap, + CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + } + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if (cs35l36->pdata.sclk_frc) + regmap_update_bits(cs35l36->regmap, + CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, + cs35l36->pdata.sclk_frc << + CS35L36_SCLK_FRC_SHIFT); + + if (cs35l36->pdata.lrclk_frc) + regmap_update_bits(cs35l36->regmap, + CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, + cs35l36->pdata.lrclk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + if (cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_update_bits(cs35l36->regmap, + CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, + CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + } + + if (cs35l36->pdata.amp_gain_zc) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_MASK, + CS35L36_AMP_ZC_MASK); + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.ldm_mode_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, + CS35L36_NG_AMP_EN_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.pdm_ldm_enter) + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_MASK, + CS35L36_PDM_LDM_ENTER_MASK); + + if (cs35l36->pdata.pdm_ldm_exit) + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_MASK, + CS35L36_PDM_LDM_EXIT_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, + status, ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, + masks, ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, + CS35L36_TEMP_WARN); + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, + CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, + CS35L36_TEMP_ERR); + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, + CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, + CS35L36_BST_OVP_ERR); + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, + CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, + CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + } + + return IRQ_HANDLED; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct irq_cfg *irq_gpio_config = &pdata->irq_config; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *irq_gpio, *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->sclk_frc = of_property_read_bool(np, + "cirrus,sclk-force-output"); + + pdata->lrclk_frc = of_property_read_bool(np, + "cirrus,lrclk-force-output"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_gain_zc = of_property_read_bool(np, + "cirrus,amp-gain-zc"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + ret = of_property_read_u32(np, "cirrus,ldm-mode-select", &val); + if (!ret) + pdata->ldm_mode_sel = val; + + pdata->pdm_ldm_exit = of_property_read_bool(np, + "cirrus,pdm-ldm-exit"); + + pdata->pdm_ldm_enter = of_property_read_bool(np, + "cirrus,pdm-ldm-enter"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + /* INT/GPIO Pin Config */ + irq_gpio = of_get_child_by_name(np, "cirrus,irq-config"); + irq_gpio_config->is_present = irq_gpio ? true : false; + if (irq_gpio_config->is_present) { + if (of_property_read_u32(irq_gpio, "cirrus,irq-drive-select", + &val) >= 0) + irq_gpio_config->irq_drv_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-polarity", + &val) >= 0) + irq_gpio_config->irq_pol = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-gpio-select", + &val) >= 0) + irq_gpio_config->irq_gpio_sel = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-output-enable", + &val) >= 0) + irq_gpio_config->irq_out_en = val; + if (of_property_read_u32(irq_gpio, "cirrus,irq-src-select", + &val) >= 0) + irq_gpio_config->irq_src_sel = val; + } + of_node_put(irq_gpio); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id == CS35L36_REV_B0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + return 0; +} + +static int cs35l36_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); + + return 0; +} + +static int cs35l36_irq_gpio_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct irq_cfg *irq_config = &pdata->irq_config; + int irq_polarity; + + /* setup the Interrupt Pin (INT | GPIO) */ + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, + irq_config->irq_out_en); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + irq_config->irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + irq_config->irq_pol << + CS35L36_INT_POL_SEL_SHIFT); + if (cs35l36->rev_id == CS35L36_REV_A0) + regmap_update_bits(cs35l36->regmap, + CS35L36_PAD_INTERFACE, + CS35L36_IRQ_SRC_MASK, + irq_config->irq_src_sel << + CS35L36_IRQ_SRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + irq_config->irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (irq_config->irq_pol) + irq_polarity = IRQF_TRIGGER_HIGH; + else + irq_polarity = IRQF_TRIGGER_LOW; + + return irq_polarity; +} + +static const struct reg_sequence cs35l36_pac_int_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + + int i; + int ret; + u32 reg_id, reg_revid, l37_id_reg; + int irq_pol = IRQF_TRIGGER_HIGH; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) + cs35l36->pdata = *pdata; + else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) + dev_info(dev, + "Reset line busy, assuming shared reset\n"); + else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err; + } + } + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", + reg_id, CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_pac_int_patch, + ARRAY_SIZE(cs35l36_pac_int_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0PAC errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_vpbr_config(cs35l36); + + if (pdata->irq_config.is_present) + irq_pol = cs35l36_irq_gpio_config(cs35l36); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | + irq_pol, + "cs35l36", cs35l36); + + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + snd_soc_unregister_component(&client->dev); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..5e1dc84 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,451 @@ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_MAX_CACHE_REG 0x00007E44 +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_AMP_EN_SHIFT 8 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_MASK 0x400 +#define CS35L36_PDM_LDM_ENTER_MASK 0x08 +#define CS35L36_PDM_LDM_EXIT_MASK 0x10 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg); +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg); + +extern struct reg_default cs35l36_reg[CS35L36_MAX_CACHE_REG]; +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

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[alsa-devel] USB speaker with CM103+ chip has left and right audio inverted
by Helge Deller 05 Dec '18

05 Dec '18

Hello, When using my Logitech LogiLink USB Speaker with Linux, the left and right audio channel is inverted. lsusb shows Bus 001 Device 016: ID 0d8c:0104 C-Media Electronics, Inc. CM103+ Audio Controller dmesg gives: kernel: usb 1-3: new full-speed USB device number 16 using xhci_hcd kernel: usb 1-3: New USB device found, idVendor=0d8c, idProduct=0104, bcdDevice= 0.10 kernel: usb 1-3: New USB device strings: Mfr=1, Product=2, SerialNumber=0 kernel: usb 1-3: Product: USB Sound Device kernel: usb 1-3: Manufacturer: C-Media INC. kernel: usb 1-3: Warning! Unlikely big volume range (=9472), cval->res is probably wrong. kernel: usb 1-3: [13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1 kernel: input: C-Media INC. USB Sound Device as /devices/pci0000:00/0000:00:14.0/usb1/1-3/1-3:1.2/0003:0D8C:0104.0034/input/input59 kernel: hid-generic 0003:0D8C:0104.0034: input,hidraw4: USB HID v1.00 Device [C-Media INC. USB Sound Device ] on usb-0000:00:14.0-3/input2 Is there a quirk possible? Any idea/patch what I could test? This is on Fedora29 with kernel 4.19.5-300.fc29.x86_64. Thanks, Helge

1 0

[alsa-devel] [resend PATCH] dt-bindings: sound: Add documentation for pcm3060 property out-single-ended
by Kirill Marinushkin 05 Dec '18

05 Dec '18

Output of pcm3060 codec may be configured as single-ended or differential Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Cc: devicetree(a)vger.kernel.org --- Hello Mark, yesterday there was a misunderstanding: when I wrote you > I think you forgot one patch in the series you accidently applied the already-applied patch instead of the forgotten one. To avoid the confusion, in this e-mail I resend you the proper forgotten patch in a separate mailing thread. Please apply this patch for-next. Best Regards, Kirill --- Documentation/devicetree/bindings/sound/pcm3060.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt index 90fcb8523099..97de66932d44 100644 --- a/Documentation/devicetree/bindings/sound/pcm3060.txt +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -9,9 +9,15 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + Examples: pcm3060: pcm3060@46 { compatible = "ti,pcm3060"; reg = <0x46>; + ti,out-single-ended = "true"; }; -- 2.13.6

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Re: [alsa-devel] [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880
by Takashi Iwai 05 Dec '18

05 Dec '18

On Wed, 05 Dec 2018 13:37:43 +0100, Kailang wrote: > > Hi Takashi, > > You should apply alc286 platform first. > I think they had test passed already. > ALC294 platform just wait for test via change model. OK, now applied this series for ALC286. For ALC294, I'll wait for the test result. thanks, Takashi > > BR, > Kailang > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2018年12月5日下午 05:32 > 至: Kailang > 副本: Jian-Hong Pan; Jaroslav Kysela; alsa-devel(a)alsa-project.org; Hui Wang; Chris Chiu; Daniel Drake; linux(a)endlessm.com; linux-kernel(a)vger.kernel.org > 主旨: Re: [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880 > > On Wed, 05 Dec 2018 10:06:14 +0100, > Kailang wrote: > > > > Sorry!! > > Don't change this. > > Hm, should I apply these series or not? > > > thanks, > > Takashi > > > > > -----Original Message----- > > From: Kailang > > Sent: Wednesday, December 5, 2018 4:52 PM > > To: 'Jian-Hong Pan' <jian-hong(a)endlessm.com>; Jaroslav Kysela <perex(a)perex.cz>; Takashi Iwai <tiwai(a)suse.com> > > Cc: Hui Wang <hui.wang(a)canonical.com>; alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; linux(a)endlessm.com; Chris Chiu <chiu(a)endlessm.com>; Daniel Drake <drake(a)endlessm.com> > > Subject: RE: [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880 > > > > Hi Jian Hong, > > > > Please also try to change model to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC. > > > > + .chain_id = ALC269_FIXUP_HEADSET_MIC ==> change to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC > > > > BR, > > Kailang > > -----Original Message----- > > From: Jian-Hong Pan <jian-hong(a)endlessm.com> > > Sent: Wednesday, December 5, 2018 2:49 PM > > To: Jaroslav Kysela <perex(a)perex.cz>; Takashi Iwai <tiwai(a)suse.com> > > Cc: Kailang <kailang(a)realtek.com>; Hui Wang <hui.wang(a)canonical.com>; alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org; linux(a)endlessm.com; Chris Chiu <chiu(a)endlessm.com>; Jian-Hong Pan <jian-hong(a)endlessm.com>; Daniel Drake <drake(a)endlessm.com> > > Subject: [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880 > > > > From: Chris Chiu <chiu(a)endlessm.com> > > > > Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. > > > > Unfortunately jack sensing for the headset mic is still not working. > > > > Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> > > Signed-off-by: Daniel Drake <drake(a)endlessm.com> > > Signed-off-by: Chris Chiu <chiu(a)endlessm.com> > > --- > > sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ > > 1 file changed, 14 insertions(+) > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0b289ba397f..f21d52eb2ed3 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -5516,6 +5516,7 @@ enum { > > ALC221_FIXUP_HP_HEADSET_MIC, > > ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, > > ALC295_FIXUP_HP_AUTO_MUTE, > > + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, > > }; > > > > static const struct hda_fixup alc269_fixups[] = { @@ -6393,6 +6394,15 @@ static const struct hda_fixup alc269_fixups[] = { > > .type = HDA_FIXUP_FUNC, > > .v.func = alc_fixup_auto_mute_via_amp, > > }, > > + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { > > + .type = HDA_FIXUP_PINS, > > + .v.pins = (const struct hda_pintbl[]) { > > + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ > > + { } > > + }, > > + .chained = true, > > + .chain_id = ALC269_FIXUP_HEADSET_MIC > > + }, > > }; > > > > static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7071,6 +7081,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > > {0x14, 0x90170110}, > > {0x19, 0x04a11040}, > > {0x21, 0x04211020}), > > + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, > > + {0x12, 0x90a60130}, > > + {0x17, 0x90170110}, > > + {0x21, 0x02211020}), > > SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, > > {0x12, 0x90a60120}, > > {0x14, 0x90170110}, > > -- > > 2.11.0 > > > > > > ------Please consider the environment before printing this e-mail.

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[alsa-devel] [PATCH linux-next v3 7/7] clk: renesas: Add binding document for ADG
by Jiada Wang 05 Dec '18

05 Dec '18

Add device tree bindings for Audio Clock Generator (ADG) of R-Car Socs. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- .../clock/renesas,rcar-adg-clocks.txt | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt diff --git a/Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt b/Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt new file mode 100644 index 000000000000..76fc4f8964e9 --- /dev/null +++ b/Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt @@ -0,0 +1,24 @@ +* Renesas R-Car Audio Clock Generator (ADG) + +The Audio Clock Generator (ADG) is part of R-Car Audio module, it +selects and supplies the necessary clock for the SSIU, EAVB-IF, SCU, +ADSP or DTCP module. It also divides the frequency of the selected +clock and sends it outside the chip. + +Required Properties: + + - compatible: Must be one of + - "renesas,rcar_sound-gen1" for the R-Car GEN1 SoCs + - "renesas,rcar_sound-gen2" for the R-Car GEN2 SoCs + - "renesas,rcar_sound-gen3" for the R-Car GEN3 SoCs + + - reg: Base address and length of the memory resource used by the ADG + + - clocks: References to the parent clock S0D1. + - clock-names: ADG refer to its parent clock by name "adg". + - #clock-cells: Can be 0, 1 or 2 + - When clock-cells = 0, ADG registers one fixed clock out + - When clock-cells = 1, ADG registers 3 fixed clock out + - When clock-cells = 2, ADG registers 3 fixed clock out and 8 AVB clocks + second clock specifier need to be 0 to refer to fixed clock out, need + to be 1 to refer to AVB clocks -- 2.19.2

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Re: [alsa-devel] No audio on iMac18,1 (A1418) with kernel 4.15.14
by Andrey L 05 Dec '18

05 Dec '18

Good day. I'm yet another owner of 2018 iMac who can't get sound to work. I was wondering if someone with the expertise can comment whether it is possible at all. Relevant earlier threads: https://www.spinics.net/lists/alsa-devel/msg76308.html https://bbs.archlinux.org/viewtopic.php?id=233449 Like others before I tried pretty much all options of snd-hda-intel with no success. The sound however works on windows, so I decided to look there. The codec chip is Cirrus CS4208, and without installing chip drivers, there was no audio in windows as well. It turns out that windows driver is a generic chip driver, where all hardware configuration is stored in registry (binary config file). These settings are configured in installation inf file (text config file) , and what is important – they are heavily commented. Unfortunately I do lack domain knowledge about hda-intel / codec / endpoint or alsa architecture, nor have I any desire to learn this area. I would appreciate if anyone can look at my findings and tell whether current linux drivers can be hacked to support this configuration. I’ve copied and sorted all relevant sections from inf file here - https://pastebin.com/p74qSdzc Here is what INF file defines: The chip is CS4208, but it is also referenced as "Cirrus Logic CS8409 (AB 16)" . There are 4 interfaces: speaker, headphones, mic and d-mic. Headphones are stereo. Speakers are quad(!!!) “quad (FL/FR - woofers, RL/RR – tweeters)”. The INF file contains following sections: CONF_0910.InitVerbs – apparently configuration initialization sequence, lists register-value pairs written by driver at startup. SettingsCS8409Exit – same for shutdown CONF_0900.Gpio, CONF_0910.Gpio – configuration of GPIO with comments CONF_0900.TXRX – Widgets,pins and amps caps (I have no clue what it is), but from comments this section seems to define all chip connections. CONF_09xx.I2C – raw i2c commands for various scenarios (chip power up, start/stop streaming, etc). So the question is, is this data of any help? Can this be somehow plugged into existing hda-intel or cs4207 drivers to get the sound? Thanks.

1 0

[alsa-devel] remove bogus GFP_DMA32 flags for dma allocations
by Christoph Hellwig 05 Dec '18

05 Dec '18

dma_alloc_* internally selects the zone to allocate from based on the DMA mask. Remove all explicit uses of GFP_DMA32 with dma_alloc_*.

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[alsa-devel] [PATCH v2] ASoC: DA7219: Implement error check on reg read and write
by Agrawal, Akshu 05 Dec '18

05 Dec '18

Failed i2c transaction can lead to failure in reg read or write. Need to have error check for each read write operation. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: replaced snd_soc_component_read32 by snd_soc_component_read Since, snd_soc_component_read32 implementation has error, in coming patches we can remove snd_soc_component_read32 from soc-io and replace all codecs by snd_soc_component_read sound/soc/codecs/da7219.c | 349 +++++++++++++++++++++++++++++++++++----------- sound/soc/codecs/da7219.h | 2 +- 2 files changed, 265 insertions(+), 86 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index e46e9f4..c65e734 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -312,69 +312,105 @@ static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, } /* ALC */ -static void da7219_alc_calib(struct snd_soc_component *component) +static int da7219_alc_calib(struct snd_soc_component *component) { - u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + unsigned int mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + int ret; /* Save current state of mic control register */ - mic_ctrl = snd_soc_component_read32(component, DA7219_MIC_1_CTRL); + ret = snd_soc_component_read(component, DA7219_MIC_1_CTRL, &mic_ctrl); + if (ret) + return ret; /* Save current state of input mixer control register */ - mixin_ctrl = snd_soc_component_read32(component, DA7219_MIXIN_L_CTRL); + ret = snd_soc_component_read(component, DA7219_MIXIN_L_CTRL, + &mixin_ctrl); + if (ret) + return ret; /* Save current state of input ADC control register */ - adc_ctrl = snd_soc_component_read32(component, DA7219_ADC_L_CTRL); + ret = snd_soc_component_read(component, DA7219_ADC_L_CTRL, &adc_ctrl); + if (ret) + return ret; /* Enable then Mute MIC PGAs */ - snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK, + ret = snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_MASK, DA7219_MIC_1_AMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_MUTE_EN_MASK, DA7219_MIC_1_AMP_MUTE_EN_MASK); + if (ret < 0) + return ret; /* Enable input mixers unmuted */ - snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, DA7219_MIXIN_L_AMP_EN_MASK | DA7219_MIXIN_L_AMP_MUTE_EN_MASK, DA7219_MIXIN_L_AMP_EN_MASK); + if (ret < 0) + return ret; /* Enable input filters unmuted */ - snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK, DA7219_ADC_L_EN_MASK); + if (ret < 0) + return ret; /* Perform auto calibration */ - snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, + ret = snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, DA7219_ALC_AUTO_CALIB_EN_MASK, DA7219_ALC_AUTO_CALIB_EN_MASK); + if (ret < 0) + return ret; do { - calib_ctrl = snd_soc_component_read32(component, DA7219_ALC_CTRL1); + ret = snd_soc_component_read(component, DA7219_ALC_CTRL1, + &calib_ctrl); + if (ret) + return ret; } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK); /* If auto calibration fails, disable DC offset, hybrid ALC */ if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) { dev_warn(component->dev, "ALC auto calibration failed with overflow\n"); - snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, + ret = snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, DA7219_ALC_OFFSET_EN_MASK | DA7219_ALC_SYNC_MODE_MASK, 0); + if (ret < 0) + return ret; } else { /* Enable DC offset cancellation, hybrid mode */ - snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, + ret = snd_soc_component_update_bits(component, DA7219_ALC_CTRL1, DA7219_ALC_OFFSET_EN_MASK | DA7219_ALC_SYNC_MODE_MASK, DA7219_ALC_OFFSET_EN_MASK | DA7219_ALC_SYNC_MODE_MASK); + if (ret < 0) + return ret; } /* Restore input filter control register to original state */ - snd_soc_component_write(component, DA7219_ADC_L_CTRL, adc_ctrl); + ret = snd_soc_component_write(component, DA7219_ADC_L_CTRL, adc_ctrl); + if (ret) + return ret; /* Restore input mixer control registers to original state */ - snd_soc_component_write(component, DA7219_MIXIN_L_CTRL, mixin_ctrl); + ret = snd_soc_component_write(component, DA7219_MIXIN_L_CTRL, + mixin_ctrl); + if (ret) + return ret; /* Restore MIC control registers to original states */ - snd_soc_component_write(component, DA7219_MIC_1_CTRL, mic_ctrl); + ret = snd_soc_component_write(component, DA7219_MIC_1_CTRL, mic_ctrl); + if (ret) + return ret; + + return 0; } static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, @@ -391,7 +427,7 @@ static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, * make sure calibrated offsets are updated. */ if ((ret == 1) && (da7219->alc_en)) - da7219_alc_calib(component); + ret = da7219_alc_calib(component); return ret; } @@ -401,11 +437,14 @@ static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + int ret; /* Force ALC offset calibration if enabling ALC */ if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { - da7219_alc_calib(component); + ret = da7219_alc_calib(component); + if (ret) + return ret; da7219->alc_en = true; } else { da7219->alc_en = false; @@ -797,7 +836,7 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); - u8 pll_ctrl, pll_status; + unsigned int pll_ctrl, pll_status; int i = 0, ret; bool srm_lock = false; @@ -813,25 +852,35 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, return ret; } } else { - snd_soc_component_update_bits(component, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, DA7219_DAI_CLK_EN_MASK); + if (ret < 0) + return ret; } } /* PC synchronised to DAI */ - snd_soc_component_update_bits(component, DA7219_PC_COUNT, + ret = snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, 0); + if (ret < 0) + return ret; /* Slave mode, if SRM not enabled no need for status checks */ - pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL); + ret = snd_soc_component_read(component, DA7219_PLL_CTRL, + &pll_ctrl); + if (ret) + return ret; if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) return 0; /* Check SRM has locked */ do { - pll_status = snd_soc_component_read32(component, DA7219_PLL_SRM_STS); + ret = snd_soc_component_read(component, + DA7219_PLL_SRM_STS, &pll_status); + if (ret) + return ret; if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { srm_lock = true; } else { @@ -846,19 +895,23 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, return 0; case SND_SOC_DAPM_POST_PMD: /* PC free-running */ - snd_soc_component_update_bits(component, DA7219_PC_COUNT, + ret = snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, DA7219_PC_FREERUN_MASK); + if (ret < 0) + return ret; /* Disable DAI clks if in master mode */ if (da7219->master) { if (da7219->dai_clks) clk_disable_unprepare(da7219->dai_clks); else - snd_soc_component_update_bits(component, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, 0); + if (ret < 0) + return ret; } return 0; @@ -887,6 +940,7 @@ static int da7219_mixout_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); u8 hp_ctrl, min_gain_mask; + int ret; switch (w->reg) { case DA7219_MIXOUT_L_CTRL: @@ -904,15 +958,20 @@ static int da7219_mixout_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMD: /* Enable minimum gain on HP to avoid pops */ - snd_soc_component_update_bits(component, hp_ctrl, min_gain_mask, - min_gain_mask); + ret = snd_soc_component_update_bits(component, hp_ctrl, + min_gain_mask, min_gain_mask); + if (ret < 0) + return ret; msleep(DA7219_MIN_GAIN_DELAY); break; case SND_SOC_DAPM_POST_PMU: /* Remove minimum gain on HP */ - snd_soc_component_update_bits(component, hp_ctrl, min_gain_mask, 0); + ret = snd_soc_component_update_bits(component, hp_ctrl, + min_gain_mask, 0); + if (ret < 0) + return ret; break; } @@ -925,21 +984,30 @@ static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_PRE_PMU: case SND_SOC_DAPM_PRE_PMD: /* Ensure nominal gain ramping for DAPM sequence */ - da7219->gain_ramp_ctrl = - snd_soc_component_read32(component, DA7219_GAIN_RAMP_CTRL); - snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, - DA7219_GAIN_RAMP_RATE_NOMINAL); + ret = snd_soc_component_read(component, + DA7219_GAIN_RAMP_CTRL, + &da7219->gain_ramp_ctrl); + if (ret) + return ret; + ret = snd_soc_component_write(component, + DA7219_GAIN_RAMP_CTRL, + DA7219_GAIN_RAMP_RATE_NOMINAL); + if (ret) + return ret; break; case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_POST_PMD: /* Restore previous gain ramp settings */ - snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, + ret = snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, da7219->gain_ramp_ctrl); + if (ret) + return ret; break; } @@ -1177,13 +1245,17 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case DA7219_CLKSRC_MCLK_SQR: - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_MCLK_SQR_EN_MASK, DA7219_PLL_MCLK_SQR_EN_MASK); + if (ret < 0) + return ret; break; case DA7219_CLKSRC_MCLK: - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_MCLK_SQR_EN_MASK, 0); + if (ret < 0) + return ret; break; default: dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); @@ -1219,6 +1291,7 @@ int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int u8 pll_frac_top, pll_frac_bot, pll_integer; u32 freq_ref; u64 frac_div; + int ret; /* Verify 2MHz - 54MHz MCLK provided, and set input divider */ if (da7219->mclk_rate < 2000000) { @@ -1252,9 +1325,11 @@ int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int switch (source) { case DA7219_SYSCLK_MCLK: pll_ctrl |= DA7219_PLL_MODE_BYPASS; - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, pll_ctrl); + if (ret < 0) + return ret; return 0; case DA7219_SYSCLK_PLL: pll_ctrl |= DA7219_PLL_MODE_NORMAL; @@ -1275,12 +1350,23 @@ int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int pll_frac_bot = (frac_div) & DA7219_BYTE_MASK; /* Write PLL config & dividers */ - snd_soc_component_write(component, DA7219_PLL_FRAC_TOP, pll_frac_top); - snd_soc_component_write(component, DA7219_PLL_FRAC_BOT, pll_frac_bot); - snd_soc_component_write(component, DA7219_PLL_INTEGER, pll_integer); - snd_soc_component_update_bits(component, DA7219_PLL_CTRL, + ret = snd_soc_component_write(component, DA7219_PLL_FRAC_TOP, + pll_frac_top); + if (ret) + return ret; + ret = snd_soc_component_write(component, DA7219_PLL_FRAC_BOT, + pll_frac_bot); + if (ret) + return ret; + ret = snd_soc_component_write(component, DA7219_PLL_INTEGER, + pll_integer); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_PLL_CTRL, DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, pll_ctrl); + if (ret < 0) + return ret; return 0; } @@ -1304,6 +1390,7 @@ static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_component *component = codec_dai->component; struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 dai_clk_mode = 0, dai_ctrl = 0; + int ret; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -1379,12 +1466,16 @@ static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* By default 64 BCLKs per WCLK is supported */ dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64; - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_BCLKS_PER_WCLK_MASK | DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK, dai_clk_mode); - snd_soc_component_update_bits(component, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK, - dai_ctrl); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_DAI_CTRL, + DA7219_DAI_FORMAT_MASK, dai_ctrl); + if (ret < 0) + return ret; return 0; } @@ -1398,15 +1489,22 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, u8 dai_bclks_per_wclk; u16 offset; u32 frame_size; + int ret; /* No channels enabled so disable TDM, revert to 64-bit frames */ if (!tx_mask) { - snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, + ret = snd_soc_component_update_bits(component, + DA7219_DAI_TDM_CTRL, DA7219_DAI_TDM_CH_EN_MASK | DA7219_DAI_TDM_MODE_EN_MASK, 0); - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, + DA7219_DAI_CLK_MODE, DA7219_DAI_BCLKS_PER_WCLK_MASK, DA7219_DAI_BCLKS_PER_WCLK_64); + if (ret < 0) + return ret; return 0; } @@ -1444,19 +1542,25 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, return -EINVAL; } - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_BCLKS_PER_WCLK_MASK, dai_bclks_per_wclk); + if (ret < 0) + return ret; offset = cpu_to_le16(rx_mask); - regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, + ret = regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, &offset, sizeof(offset)); + if (ret) + return ret; - snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, DA7219_DAI_TDM_CH_EN_MASK | DA7219_DAI_TDM_MODE_EN_MASK, (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | DA7219_DAI_TDM_MODE_EN_MASK); + if (ret < 0) + return ret; return 0; } @@ -1468,6 +1572,7 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; u8 dai_ctrl = 0, fs; unsigned int channels; + int ret; switch (params_width(params)) { case 16: @@ -1533,11 +1638,15 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_component_update_bits(component, DA7219_DAI_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CTRL, DA7219_DAI_WORD_LENGTH_MASK | DA7219_DAI_CH_NUM_MASK, dai_ctrl); - snd_soc_component_write(component, DA7219_SR, fs); + if (ret < 0) + return ret; + ret = snd_soc_component_write(component, DA7219_SR, fs); + if (ret) + return ret; return 0; } @@ -1692,9 +1801,12 @@ static int da7219_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_STANDBY: if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) /* Master bias */ - snd_soc_component_update_bits(component, DA7219_REFERENCES, + ret = snd_soc_component_update_bits(component, + DA7219_REFERENCES, DA7219_BIAS_EN_MASK, DA7219_BIAS_EN_MASK); + if (ret < 0) + return ret; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_PREPARE) { /* Remove MCLK */ @@ -1705,8 +1817,11 @@ static int da7219_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_OFF: /* Only disable master bias if we're not a wake-up source */ if (!da7219->wakeup_source) - snd_soc_component_update_bits(component, DA7219_REFERENCES, + ret = snd_soc_component_update_bits(component, + DA7219_REFERENCES, DA7219_BIAS_EN_MASK, 0); + if (ret < 0) + return ret; break; } @@ -1754,10 +1869,16 @@ static int da7219_handle_supplies(struct snd_soc_component *component) } /* Ensure device in active mode */ - snd_soc_component_write(component, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); + ret = snd_soc_component_write(component, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK); + if (ret) + return ret; /* Update IO voltage level range */ - snd_soc_component_write(component, DA7219_IO_CTRL, io_voltage_lvl); + ret = snd_soc_component_write(component, DA7219_IO_CTRL, + io_voltage_lvl); + if (ret) + return ret; return 0; } @@ -1768,10 +1889,13 @@ static int da7219_dai_clks_prepare(struct clk_hw *hw) struct da7219_priv *da7219 = container_of(hw, struct da7219_priv, dai_clks_hw); struct snd_soc_component *component = da7219->aad->component; + int ret; - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, DA7219_DAI_CLK_EN_MASK); + if (ret < 0) + return ret; return 0; } @@ -1781,9 +1905,12 @@ static void da7219_dai_clks_unprepare(struct clk_hw *hw) struct da7219_priv *da7219 = container_of(hw, struct da7219_priv, dai_clks_hw); struct snd_soc_component *component = da7219->aad->component; + int ret; - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + ret = snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, 0); + if (ret < 0) + dev_err(component->dev, "Failed to disable clk: %d\n", ret); } static int da7219_dai_clks_is_prepared(struct clk_hw *hw) @@ -1791,9 +1918,12 @@ static int da7219_dai_clks_is_prepared(struct clk_hw *hw) struct da7219_priv *da7219 = container_of(hw, struct da7219_priv, dai_clks_hw); struct snd_soc_component *component = da7219->aad->component; - u8 clk_reg; + unsigned int clk_reg; + int ret; - clk_reg = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); + ret = snd_soc_component_read(component, DA7219_DAI_CLK_MODE, &clk_reg); + if (ret) + return ret; return !!(clk_reg & DA7219_DAI_CLK_EN_MASK); } @@ -1844,10 +1974,11 @@ static void da7219_register_dai_clks(struct snd_soc_component *component) static inline void da7219_register_dai_clks(struct snd_soc_component *component) {} #endif /* CONFIG_COMMON_CLK */ -static void da7219_handle_pdata(struct snd_soc_component *component) +static int da7219_handle_pdata(struct snd_soc_component *component) { struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); struct da7219_pdata *pdata = da7219->pdata; + int ret; if (pdata) { u8 micbias_lvl = 0; @@ -1869,7 +2000,10 @@ static void da7219_handle_pdata(struct snd_soc_component *component) break; } - snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + ret = snd_soc_component_write(component, DA7219_MICBIAS_CTRL, + micbias_lvl); + if (ret) + return ret; /* * Calculate delay required to compensate for DC offset in @@ -1884,11 +2018,15 @@ static void da7219_handle_pdata(struct snd_soc_component *component) case DA7219_MIC_AMP_IN_SEL_DIFF: case DA7219_MIC_AMP_IN_SEL_SE_P: case DA7219_MIC_AMP_IN_SEL_SE_N: - snd_soc_component_write(component, DA7219_MIC_1_SELECT, - pdata->mic_amp_in_sel); + ret = snd_soc_component_write(component, + DA7219_MIC_1_SELECT, + pdata->mic_amp_in_sel); + if (ret) + return ret; break; } } + return 0; } static struct reg_sequence da7219_rev_aa_patch[] = { @@ -1934,7 +2072,9 @@ static int da7219_probe(struct snd_soc_component *component) if (!da7219->pdata) da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); + ret = da7219_handle_pdata(component); + if (ret) + return ret; /* Check if MCLK provided */ da7219->mclk = devm_clk_get(component->dev, "mclk"); @@ -1948,36 +2088,57 @@ static int da7219_probe(struct snd_soc_component *component) } /* Default PC counter to free-running */ - snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, - DA7219_PC_FREERUN_MASK); + ret = snd_soc_component_update_bits(component, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, DA7219_PC_FREERUN_MASK); + if (ret < 0) + return ret; /* Default gain ramping */ - snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_MIXIN_L_CTRL, DA7219_MIXIN_L_AMP_RAMP_EN_MASK, DA7219_MIXIN_L_AMP_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK, - DA7219_ADC_L_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK, - DA7219_DAC_L_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK, - DA7219_DAC_R_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_ADC_L_CTRL, + DA7219_ADC_L_RAMP_EN_MASK, DA7219_ADC_L_RAMP_EN_MASK); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_DAC_L_CTRL, + DA7219_DAC_L_RAMP_EN_MASK, DA7219_DAC_L_RAMP_EN_MASK); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_DAC_R_CTRL, + DA7219_DAC_R_RAMP_EN_MASK, DA7219_DAC_R_RAMP_EN_MASK); + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_RAMP_EN_MASK, DA7219_HP_L_AMP_RAMP_EN_MASK); - snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_RAMP_EN_MASK, DA7219_HP_R_AMP_RAMP_EN_MASK); + if (ret < 0) + return ret; /* Default minimum gain on HP to avoid pops during DAPM sequencing */ - snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, + ret = snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_MIN_GAIN_EN_MASK, DA7219_HP_L_AMP_MIN_GAIN_EN_MASK); - snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_MIN_GAIN_EN_MASK, DA7219_HP_R_AMP_MIN_GAIN_EN_MASK); + if (ret < 0) + return ret; /* Default infinite tone gen, start/stop by Kcontrol */ - snd_soc_component_write(component, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); + ret = snd_soc_component_write(component, DA7219_TONE_GEN_CYCLES, + DA7219_BEEP_CYCLES_MASK); + if (ret) + return ret; /* Initialise AAD block */ ret = da7219_aad_init(component); @@ -2221,16 +2382,28 @@ static int da7219_i2c_probe(struct i2c_client *i2c, regcache_cache_bypass(da7219->regmap, true); /* Disable audio paths if still active from previous start */ - regmap_read(da7219->regmap, DA7219_SYSTEM_ACTIVE, &system_active); + ret = regmap_read(da7219->regmap, DA7219_SYSTEM_ACTIVE, &system_active); + if (ret) + return ret; if (system_active) { - regmap_write(da7219->regmap, DA7219_GAIN_RAMP_CTRL, + ret = regmap_write(da7219->regmap, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_NOMINAL); - regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_INPUT, 0x00); - regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_OUTPUT, 0x01); + if (ret) + return ret; + ret = regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_INPUT, + 0x00); + if (ret) + return ret; + ret = regmap_write(da7219->regmap, DA7219_SYSTEM_MODES_OUTPUT, + 0x01); + if (ret) + return ret; for (i = 0; i < DA7219_SYS_STAT_CHECK_RETRIES; ++i) { - regmap_read(da7219->regmap, DA7219_SYSTEM_STATUS, + ret = regmap_read(da7219->regmap, DA7219_SYSTEM_STATUS, &system_status); + if (ret) + return ret; if (!system_status) break; @@ -2239,13 +2412,19 @@ static int da7219_i2c_probe(struct i2c_client *i2c, } /* Soft reset component */ - regmap_write_bits(da7219->regmap, DA7219_ACCDET_CONFIG_1, + ret = regmap_write_bits(da7219->regmap, DA7219_ACCDET_CONFIG_1, DA7219_ACCDET_EN_MASK, 0); - regmap_write_bits(da7219->regmap, DA7219_CIF_CTRL, + if (ret) + return ret; + ret = regmap_write_bits(da7219->regmap, DA7219_CIF_CTRL, DA7219_CIF_REG_SOFT_RESET_MASK, DA7219_CIF_REG_SOFT_RESET_MASK); - regmap_write_bits(da7219->regmap, DA7219_SYSTEM_ACTIVE, + if (ret) + return ret; + ret = regmap_write_bits(da7219->regmap, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK, 0); + if (ret) + return ret; regcache_cache_bypass(da7219->regmap, false); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 3a00686..78d2194 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -832,7 +832,7 @@ struct da7219_priv { bool alc_en; bool micbias_on_event; unsigned int mic_pga_delay; - u8 gain_ramp_ctrl; + unsigned int gain_ramp_ctrl; }; int da7219_set_pll(struct snd_soc_component *component, int source, unsigned int fout); -- 1.9.1

1 0

[alsa-devel] [PATCH 0/3] Add ALC294 quirks for ASUS laptops
by Jian-Hong Pan 05 Dec '18

05 Dec '18

Add Realtek ALC294 quirks for ASUS X542UN, UX533FD and UX433FN laptops. Chris Chiu (1): ALSA: hda/realtek: ALC294 mic and headset-mode fixups for ASUS X542UN Jian-Hong Pan (2): ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294 ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN with ALC294 sound/pci/hda/patch_realtek.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) -- 2.11.0

2 5

[alsa-devel] [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880
by Jian-Hong Pan 05 Dec '18

05 Dec '18

From: Chris Chiu <chiu(a)endlessm.com> Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. Unfortunately jack sensing for the headset mic is still not working. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0b289ba397f..f21d52eb2ed3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5516,6 +5516,7 @@ enum { ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, ALC295_FIXUP_HP_AUTO_MUTE, + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6393,6 +6394,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_auto_mute_via_amp, }, + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7071,6 +7081,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, -- 2.11.0

3 6

[alsa-devel] [PATCH linux-next v3 5/7] clk: renesas: r8a77965: Add ADG clock
by Jiada Wang 05 Dec '18

05 Dec '18

From: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> This patch adds ADG clock to the R8A77965 SoC. Signed-off-by: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- drivers/clk/renesas/r8a77965-cpg-mssr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/clk/renesas/r8a77965-cpg-mssr.c b/drivers/clk/renesas/r8a77965-cpg-mssr.c index 1fcc411502da..3ce8b0b31647 100644 --- a/drivers/clk/renesas/r8a77965-cpg-mssr.c +++ b/drivers/clk/renesas/r8a77965-cpg-mssr.c @@ -211,6 +211,7 @@ static const struct mssr_mod_clk r8a77965_mod_clks[] __initconst = { DEF_MOD("can-if0", 916, R8A77965_CLK_S3D4), DEF_MOD("i2c6", 918, R8A77965_CLK_S0D6), DEF_MOD("i2c5", 919, R8A77965_CLK_S0D6), + DEF_MOD("adg", 922, R8A77965_CLK_S0D1), DEF_MOD("i2c-dvfs", 926, R8A77965_CLK_CP), DEF_MOD("i2c4", 927, R8A77965_CLK_S0D6), DEF_MOD("i2c3", 928, R8A77965_CLK_S0D6), -- 2.19.2

1 0

[alsa-devel] [PATCH linux-next v3 0/7] clk: renesas: adg: add AVB Clock
by Jiada Wang 05 Dec '18

05 Dec '18

on R-Car SoCs there are AVB Counter Clocks, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch-set adds 'adg' clock to R-Car Soc, and changes adg driver to register avb clocks when clock-cells of rcar_sound node is 2. --- v3: - Removed clock id header file, added device tree bindings documentation instead to describe clock id - Instead of hardcode parent clock name in adg driver, refer to parent clock via clock specifier in device tree with 'adg' name - Added patch to add ADG in r8a77965 - Some other fixes v2: - expends adg register size and register avb clocks instead of add new clk-avb driver - Add adg clock v1: initial version Jiada Wang (2): ASoC: rsnd: add avb clocks clk: renesas: Add binding document for ADG Takeshi Kihara (5): clk: renesas: r8a7795: Add ADG clock clk: renesas: r8a7796: Add ADG clock clk: renesas: r8a77990: Add ADG clock clk: renesas: r8a77995: Add ADG clock clk: renesas: r8a77965: Add ADG clock .../clock/renesas,rcar-adg-clocks.txt | 24 ++ drivers/clk/renesas/r8a7795-cpg-mssr.c | 1 + drivers/clk/renesas/r8a7796-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77965-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77990-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77995-cpg-mssr.c | 1 + sound/soc/sh/rcar/adg.c | 316 +++++++++++++++++- sound/soc/sh/rcar/gen.c | 9 + sound/soc/sh/rcar/rsnd.h | 9 + 9 files changed, 354 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt -- 2.19.2

1 4

[alsa-devel] axfer: check whether a terminal is referred for stdio
by GitHub pull_request - opened 05 Dec '18

05 Dec '18

alsa-project/alsa-utils pull request #5 was opened from takaswie: A reference to a terminal for standard input/output brings some troubles. For capture transmission, it can bring some terminal control codes and it's possible to make the terminal in disorder. For playback transmission, it can bring endless loop to read data for detection of type of container. This commit checks whether a terminal is referred for the standard input/output by a call isatty(3). When detecting a terminal, axfer run time prints message and goes to finish. Suggested-by: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-utils/pull/5 Patch URL : https://github.com/alsa-project/alsa-utils/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: pxa: remove raumfeld machine driver
by Daniel Mack 04 Dec '18

04 Dec '18

These boards are now fully ported to devicetree and make use of the simple-card driver, so the platform specific machine driver can be removed. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/Kconfig | 10 -- sound/soc/pxa/Makefile | 1 - sound/soc/pxa/raumfeld.c | 318 --------------------------------------- 3 files changed, 329 deletions(-) delete mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 943b44de1464..a6968bfe1d2d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -169,16 +169,6 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. -config SND_SOC_RAUMFELD - tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - depends on I2C && SPI_MASTER - select SND_PXA_SOC_SSP - select SND_SOC_CS4270 - select SND_SOC_AK4104 - help - Say Y if you want to add support for SoC audio on Raumfeld devices - config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" depends on SND_PXA2XX_SOC && MACH_H4700 && I2C diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..0ab2a9dcb720 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -49,6 +49,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o -obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c deleted file mode 100644 index 111a907c4eb9..000000000000 --- a/sound/soc/pxa/raumfeld.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * raumfeld_audio.c -- SoC audio for Raumfeld audio devices - * - * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> - * - * based on code from: - * - * Wolfson Microelectronics PLC. - * Openedhand Ltd. - * Liam Girdwood <lrg(a)slimlogic.co.uk> - * Richard Purdie <richard(a)openedhand.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/module.h> -#include <linux/i2c.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "pxa-ssp.h" - -#define GPIO_SPDIF_RESET (38) -#define GPIO_MCLK_RESET (111) -#define GPIO_CODEC_RESET (120) - -static struct i2c_client *max9486_client; -static struct i2c_board_info max9486_hwmon_info = { - I2C_BOARD_INFO("max9485", 0x63), -}; - -#define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 -#define MAX9485_MCLK_FREQ_225792 0x32 -#define MAX9485_MCLK_FREQ_245760 0x33 - -static void set_max9485_clk(char clk) -{ - i2c_master_send(max9486_client, &clk, 1); -} - -static void raumfeld_enable_audio(bool en) -{ - if (en) { - gpio_set_value(GPIO_MCLK_RESET, 1); - - /* wait some time to let the clocks become stable */ - msleep(100); - - gpio_set_value(GPIO_SPDIF_RESET, 1); - gpio_set_value(GPIO_CODEC_RESET, 1); - } else { - gpio_set_value(GPIO_MCLK_RESET, 0); - gpio_set_value(GPIO_SPDIF_RESET, 0); - gpio_set_value(GPIO_CODEC_RESET, 0); - } -} - -/* CS4270 */ -static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); - if (ret < 0) - return ret; - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops raumfeld_cs4270_ops = { - .startup = raumfeld_cs4270_startup, - .shutdown = raumfeld_cs4270_shutdown, - .hw_params = raumfeld_cs4270_hw_params, -}; - -static int raumfeld_analog_suspend(struct snd_soc_card *card) -{ - raumfeld_enable_audio(false); - return 0; -} - -static int raumfeld_analog_resume(struct snd_soc_card *card) -{ - raumfeld_enable_audio(true); - return 0; -} - -/* AK4104 */ - -static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, clk = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops raumfeld_ak4104_ops = { - .hw_params = raumfeld_ak4104_hw_params, -}; - -#define DAI_LINK_CS4270 \ -{ \ - .name = "CS4270", \ - .stream_name = "CS4270", \ - .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ - .codec_dai_name = "cs4270-hifi", \ - .codec_name = "cs4270.0-0048", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_cs4270_ops, \ -} - -#define DAI_LINK_AK4104 \ -{ \ - .name = "ak4104", \ - .stream_name = "Playback", \ - .cpu_dai_name = "pxa-ssp-dai.1", \ - .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_ak4104_ops, \ - .codec_name = "spi0.0", \ -} - -static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - DAI_LINK_CS4270, - DAI_LINK_AK4104, -}; - -static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - DAI_LINK_CS4270, -}; - -static struct snd_soc_card snd_soc_raumfeld_connector = { - .name = "Raumfeld Connector", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_connector_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct snd_soc_card snd_soc_raumfeld_speaker = { - .name = "Raumfeld Speaker", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_speaker_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct platform_device *raumfeld_audio_device; - -static int __init raumfeld_audio_init(void) -{ - int ret; - - if (!machine_is_raumfeld_speaker() && - !machine_is_raumfeld_connector()) - return 0; - - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); - - if (!max9486_client) - return -ENOMEM; - - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - - /* Register analog device */ - raumfeld_audio_device = platform_device_alloc("soc-audio", 0); - if (!raumfeld_audio_device) - return -ENOMEM; - - if (machine_is_raumfeld_speaker()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); - - ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) { - platform_device_put(raumfeld_audio_device); - return ret; - } - - raumfeld_enable_audio(true); - return 0; -} - -static void __exit raumfeld_audio_exit(void) -{ - raumfeld_enable_audio(false); - - platform_device_unregister(raumfeld_audio_device); - - i2c_unregister_device(max9486_client); - - gpio_free(GPIO_MCLK_RESET); - gpio_free(GPIO_CODEC_RESET); - gpio_free(GPIO_SPDIF_RESET); -} - -module_init(raumfeld_audio_init); -module_exit(raumfeld_audio_exit); - -/* Module information */ -MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("Raumfeld audio SoC"); -MODULE_LICENSE("GPL"); -- 2.19.2

2 1

[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Clapper
by Hans de Goede 04 Dec '18

04 Dec '18

The Clapper model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3 for the mclk, just like the Swanky model. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Clapper that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 9d9f6e41d81c..ad0c98383853 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -389,6 +389,13 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Clapper model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Clapper"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: indicates Channel and Mode for debug
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> For TDM debug purpose, indicating Channel and Mode is very useful. This patch indicate it if it has #define DEBUG Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6439a74..45ef295 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -329,7 +329,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, ssi->rate = rate; ssi->chan = chan; - dev_dbg(dev, "%s outputs %u Hz\n", rsnd_mod_name(mod), rate); + dev_dbg(dev, "%s outputs %d chan %u Hz\n", + rsnd_mod_name(mod), chan, rate); return 0; } @@ -360,6 +361,8 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); u32 cr_own = ssi->cr_own; @@ -371,6 +374,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); + if (is_tdm) + dev_dbg(dev, "TDM mode\n"); + if (is_tdm_split) + dev_dbg(dev, "TDM Split mode\n"); + cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai); if (rdai->bit_clk_inv) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup mod ID for CTU regmap read/write
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c16015f36cc12824 ("ASoC: rsnd: add .get_id/.get_id_sub") add new .get_id/.get_id_sub to indicate module ID/subID. It is used for SSIU and CTU. In SSIU case, subID indicates BUSIF, but register settings is based on SSIU ID. OTOH, in CTU case, subID indicates CTU channel, and register settings is based on it. This means regmap read/write function needs to care it. This patch fixup this issue. It can't play MIXed sound without this patch. Fixes: c16015f36cc12824 ("ASoC: rsnd: add .get_id/.get_id_sub") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 1 + sound/soc/sh/rcar/gen.c | 14 +++++++++++--- sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 4dda7d9..ff41a5e 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -317,6 +317,7 @@ static struct rsnd_mod_ops rsnd_ctu_ops = { .get_status = rsnd_mod_get_status, .id = rsnd_ctu_id, .id_sub = rsnd_ctu_id_sub, + .id_cmd = rsnd_mod_id_raw, }; struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca63940..3283d67 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -71,6 +71,14 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, return 1; } +static int rsnd_mod_id_cmd(struct rsnd_mod *mod) +{ + if (mod->ops->id_cmd) + return mod->ops->id_cmd(mod); + + return rsnd_mod_id(mod); +} + u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { @@ -81,7 +89,7 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + regmap_fields_read(gen->regs[reg], rsnd_mod_id_cmd(mod), &val); dev_dbg(dev, "r %s - %-18s (%4d) : %08x\n", rsnd_mod_name(mod), @@ -100,7 +108,7 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); + regmap_fields_force_write(gen->regs[reg], rsnd_mod_id_cmd(mod), data); dev_dbg(dev, "w %s - %-18s (%4d) : %08x\n", rsnd_mod_name(mod), @@ -117,7 +125,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, return; regmap_fields_force_update_bits(gen->regs[reg], - rsnd_mod_id(mod), mask, data); + rsnd_mod_id_cmd(mod), mask, data); dev_dbg(dev, "b %s - %-18s (%4d) : %08x/%08x\n", rsnd_mod_name(mod), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129..94d51fe 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -306,6 +306,7 @@ struct rsnd_mod_ops { enum rsnd_mod_type type); int (*id)(struct rsnd_mod *mod); int (*id_sub)(struct rsnd_mod *mod); + int (*id_cmd)(struct rsnd_mod *mod); }; struct rsnd_dai_stream; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: add missing TDM Split mode support for simple-card
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f69f452243e4e1 ("ASoC: rsnd: add TDM Split mode support") added TDM Split mode support for rsnd driver. But, it cares audio-graph-card style only. We can't use TDM Split mode on simple-card style now. This patch fixup this issue. Fixes: f69f452243e4e1 ("ASoC: rsnd: add TDM Split mode support") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 51 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 031cd5b..59e250c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1031,6 +1031,49 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { .prepare = rsnd_soc_dai_prepare, }; +static void rsnd_parse_connect_simple(struct rsnd_priv *priv, + struct device_node *dai_np, + int dai_i, int is_play) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i); + struct rsnd_dai_stream *io = is_play ? + &rdai->playback : + &rdai->capture; + struct device_node *ssiu_np = rsnd_ssiu_of_node(priv); + struct device_node *np; + int i, j; + + if (!ssiu_np) + return; + + if (!rsnd_io_to_mod_ssi(io)) + return; + + /* + * This driver assumes that it is TDM Split mode + * if it includes ssiu node + */ + for (i = 0;; i++) { + struct device_node *node = is_play ? + of_parse_phandle(dai_np, "playback", i) : + of_parse_phandle(dai_np, "capture", i); + + if (!node) + break; + + j = 0; + for_each_child_of_node(ssiu_np, np) { + if (np == node) { + rsnd_flags_set(io, RSND_STREAM_TDM_SPLIT); + dev_dbg(dev, "%s is part of TDM Split\n", io->name); + } + j++; + } + + } +} + static void rsnd_parse_connect_graph(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct device_node *endpoint) @@ -1246,8 +1289,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i++; } } else { - for_each_child_of_node(dai_node, dai_np) - __rsnd_dai_probe(priv, dai_np, dai_i++); + for_each_child_of_node(dai_node, dai_np) { + __rsnd_dai_probe(priv, dai_np, dai_i); + if (rsnd_is_gen3(priv)) { + rsnd_parse_connect_simple(priv, dai_np, dai_i, 1); + rsnd_parse_connect_simple(priv, dai_np, dai_i, 0); + } + dai_i++; + } } return 0; -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: merge audio-graph-scu into audio-graph - step3
by Kuninori Morimoto 04 Dec '18

04 Dec '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is v2 of step3 patches. I mentioned that step4 patch-set will be based on OF-graph posted patch, but, I could solve it without posted patch. There is no blocker anymore. Thus, I included blocked patch on step3. Kuninori Morimoto (5): ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props sound/soc/generic/audio-graph-card.c | 57 ++++++---- sound/soc/generic/audio-graph-scu-card.c | 180 +++++++++++++++++++++++-------- sound/soc/generic/simple-card-utils.c | 22 +--- 3 files changed, 173 insertions(+), 86 deletions(-) -- 2.7.4

2 7

[alsa-devel] Applied "ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a1af9858edae12eca8f77b152d17ee049835e57d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:15 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge audio-graph-card and audio-graph-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on audio-graph-scu-card. It is same logic with audio-graph-card, thus easy merging. This is prepare for merging audio card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-scu-card.c | 72 ++++++++++++++++-------- 1 file changed, 49 insertions(+), 23 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index a0a28671948a..ed128d430060 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -26,12 +26,14 @@ struct graph_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; struct graph_dai_props { - struct asoc_simple_dai dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; } *dai_props; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; }; @@ -47,8 +49,17 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - return asoc_simple_card_clk_enable(&dai_props->dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + return ret; } static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) @@ -57,7 +68,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); } static const struct snd_soc_ops asoc_graph_card_ops = { @@ -68,18 +81,20 @@ static const struct snd_soc_ops asoc_graph_card_ops = { static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - int num = rtd->num; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - dai_link = graph_priv_to_link(priv, num); - dai_props = graph_priv_to_props(priv, num); - dai = dai_link->dynamic ? - rtd->cpu_dai : - rtd->codec_dai; + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; - return asoc_simple_card_init_dai(dai, &dai_props->dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; } static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -99,14 +114,15 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, - int idx, int is_fe) + int *dai_idx, int link_idx, int is_fe) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *ep = is_fe ? cpu_ep : codec_ep; struct device_node *node = of_graph_get_port_parent(ep); + struct asoc_simple_dai *dai; int ret; if (is_fe) { @@ -122,11 +138,14 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; + dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); if (ret < 0) return ret; @@ -149,11 +168,14 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; + dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai); + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); if (ret < 0) return ret; @@ -178,7 +200,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai); + ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; @@ -210,7 +232,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - int dai_idx, ret; + int dai_idx, link_idx, ret; int rc, codec; if (!node) @@ -232,6 +254,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) * see asoc_graph_get_dais_count */ + link_idx = 0; dai_idx = 0; codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { @@ -260,8 +283,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, dai_idx++, - !codec); + priv, &dai_idx, + link_idx++, !codec); if (ret < 0) goto parse_of_err; } @@ -364,6 +387,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; struct graph_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; int lnum = 0, dnum = 0, cnum = 0; @@ -380,7 +404,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -397,6 +422,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = graph_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: care multi DPCM codec_conf" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-scu-card: care multi DPCM codec_conf has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7c2ae7981d73f9e4f10ed9d1dcb5bc6056189b30 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:32 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: care multi DPCM codec_conf Current audio-graph-scu-card didn't care about codec_conf for multi DPCM case. This patch cares it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-scu-card.c | 40 +++++++++++++++++------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index ed128d430060..e1b192ea147b 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -24,17 +24,18 @@ struct graph_card_data { struct snd_soc_card snd_card; - struct snd_soc_codec_conf codec_conf; struct graph_dai_props { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; struct snd_soc_dai_link_component platform; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; }; #define graph_priv_to_card(priv) (&(priv)->snd_card) @@ -114,7 +115,8 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, struct device_node *codec_ep, struct graph_card_data *priv, - int *dai_idx, int link_idx, int is_fe) + int *dai_idx, int link_idx, + int *conf_idx, int is_fe) { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); @@ -159,6 +161,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); } else { + struct snd_soc_codec_conf *cconf; + /* FE is dummy */ dai_link->cpu_of_node = NULL; dai_link->cpu_dai_name = "snd-soc-dummy-dai"; @@ -171,6 +175,9 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, dai = dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + cconf = + dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; + ret = asoc_simple_card_parse_graph_codec(ep, dai_link); if (ret < 0) return ret; @@ -186,14 +193,12 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, return ret; /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, - &priv->codec_conf, + snd_soc_of_parse_audio_prefix(card, cconf, dai_link->codecs->of_node, "prefix"); /* check "prefix" from each node if top doesn't have */ - if (!priv->codec_conf.of_node) - snd_soc_of_parse_node_prefix(node, - &priv->codec_conf, + if (!cconf->of_node) + snd_soc_of_parse_node_prefix(node, cconf, dai_link->codecs->of_node, PREFIX "prefix"); } @@ -232,7 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - int dai_idx, link_idx, ret; + int dai_idx, link_idx, conf_idx, ret; int rc, codec; if (!node) @@ -256,6 +261,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) link_idx = 0; dai_idx = 0; + conf_idx = 0; codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* @@ -284,7 +290,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, priv, &dai_idx, - link_idx++, !codec); + link_idx++, &conf_idx, + !codec); if (ret < 0) goto parse_of_err; } @@ -294,6 +301,14 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) if (ret) goto parse_of_err; + if ((card->num_links != link_idx) || + (card->num_configs != conf_idx)) { + dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", + card->num_links, link_idx, card->num_configs, conf_idx); + ret = -EINVAL; + goto parse_of_err; + } + ret = 0; parse_of_err: @@ -390,6 +405,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; + struct snd_soc_codec_conf *cconf; int lnum = 0, dnum = 0, cnum = 0; int ret, i; @@ -405,6 +421,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; @@ -423,6 +440,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; priv->dais = dais; + priv->codec_conf = cconf; /* Init snd_soc_card */ card = graph_priv_to_card(priv); @@ -430,8 +448,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = lnum; - card->codec_conf = &priv->codec_conf; - card->num_configs = 1; + card->codec_conf = cconf; + card->num_configs = cnum; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0e3460bceedc98176684602a34659d51cd8bd4ce Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 30 Nov 2018 02:07:48 +0000 Subject: [PATCH] ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and non DPCM case, it uses CPU-Codec dai links. Now, we want to merge audio-graph-card and audio-graph-scu-card. These sound cards are using silimar but not same logic on each functions. Then, of course we want to share same logic. To compromise, this patch uses cpu/codec pointer on audio-graph-card. It is same logic with audio-graph-scu-card, thus easy merging. This is prepare for merging audio card Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 57 +++++++++++++++++----------- 1 file changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1b158cc2df88..1da9532ef897 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -23,8 +23,8 @@ struct graph_card_data { struct snd_soc_card snd_card; struct graph_dai_props { - struct asoc_simple_dai cpu_dai; - struct asoc_simple_dai codec_dai; + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ struct snd_soc_dai_link_component platform; unsigned int mclk_fs; @@ -33,6 +33,7 @@ struct graph_card_data { struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; struct gpio_desc *pa_gpio; }; @@ -75,13 +76,13 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(&dai_props->cpu_dai); + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(&dai_props->codec_dai); + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); return ret; } @@ -92,9 +93,9 @@ static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(&dai_props->cpu_dai); + asoc_simple_card_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(&dai_props->codec_dai); + asoc_simple_card_clk_disable(dai_props->codec_dai); } static int asoc_graph_card_hw_params(struct snd_pcm_substream *substream, @@ -139,17 +140,16 @@ static const struct snd_soc_ops asoc_graph_card_ops = { static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) { struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - struct graph_dai_props *dai_props = - graph_priv_to_props(priv, rtd->num); - int ret; + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + int ret = 0; - ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai); + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(cpu, &dai_props->cpu_dai); + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); if (ret < 0) return ret; @@ -158,17 +158,22 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_card_data *priv, - int idx) + int *dai_idx, int link_idx) { struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); - struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; - struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; + struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; + cpu_dai = + dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; + codec_dai = + dai_props->codec_dai = &priv->dais[(*dai_idx)++]; + ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -231,8 +236,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device *dev = graph_priv_to_dev(priv); struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *node = dev->of_node; - int rc, idx = 0; - int ret; + int rc, ret; + int link_idx, dai_idx; ret = asoc_simple_card_of_parse_widgets(card, NULL); if (ret < 0) @@ -245,8 +250,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* Factor to mclk, used in hw_params() */ of_property_read_u32(node, "mclk-fs", &priv->mclk_fs); + link_idx = 0; + dai_idx = 0; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { - ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); + ret = asoc_graph_card_dai_link_of(it.node, priv, + &dai_idx, link_idx++); if (ret < 0) { of_node_put(it.node); @@ -291,6 +299,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; struct graph_dai_props *dai_props; + struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; int num, ret, i; @@ -306,7 +315,8 @@ static int asoc_graph_card_probe(struct platform_device *pdev) dai_props = devm_kcalloc(dev, num, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, num, sizeof(*dai_link), GFP_KERNEL); - if (!dai_props || !dai_link) + dais = devm_kcalloc(dev, num * 2, sizeof(*dais), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) return -ENOMEM; /* @@ -330,6 +340,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) priv->dai_props = dai_props; priv->dai_link = dai_link; + priv->dais = dais; /* Init snd_soc_card */ card = graph_priv_to_card(priv); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max98373: Added max98373_reset for stable amp reset" to the asoc tree
by Mark Brown 04 Dec '18

04 Dec '18

The patch ASoC: max98373: Added max98373_reset for stable amp reset has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 20f2ab247d3b787af91c1aa5eb27c5061744c154 Mon Sep 17 00:00:00 2001 From: Ryan Lee <RyanS.Lee(a)maximintegrated.com> Date: Fri, 30 Nov 2018 03:21:09 +0000 Subject: [PATCH] ASoC: max98373: Added max98373_reset for stable amp reset This patch added max98373_reset function to avoid amp software reset failure and code duplication. Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed. Chip revision ID is available when the amp is in the idle state which means software reset is completed well. Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d01318f79..9c8616a7b61c 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v9 0/3] Huawei laptops
by Ayman Bagabas 04 Dec '18

04 Dec '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v8: * Explicitly use NULL in acpi_get_handle * Drop __initconst from huawei_wmi_keymap Changes from v7: * Switch to wmi_driver * Use devm to allocate and manage devices * Skip registering LED subsystem if a ACPI controller method was not found Changes from v6: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes from v5: * Review tags Changes from v4: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes from v3: * Code formatting Changes from v2: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 16 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 226 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 265 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.2

5 8

[alsa-devel] [PATCH linux-next v2 6/6] ASoC: rsnd: add avb clocks
by jiada_wang＠mentor.com 04 Dec '18

04 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> There are AVB Counter Clocks in ADG, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch registers 8 AVB Counter Clocks when clock-cells of rcar_sound node is 2, Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/adg.c | 306 +++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 9 ++ sound/soc/sh/rcar/rsnd.h | 9 ++ 3 files changed, 315 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66588eb..2c03d420ae76 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -5,6 +5,8 @@ // Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/clk-provider.h> +#include <linux/of_address.h> +#include <dt-bindings/clock/renesas-adg.h> #include "rsnd.h" #define CLKA 0 @@ -21,13 +23,33 @@ #define BRRx_MASK(x) (0x3FF & x) +#define ADG_CLK_NAME "adg" +#define AVB_CLK_NAME "avb" +#define AVB_CLK_NUM 8 +#define AVB_CLK_NAME_SIZE 10 +#define AVB_MAX_RATE 25000000 +#define AVB_DIV_EN_COM BIT(31) +#define AVB_DIV_MASK 0x3ffff +#define AVB_MAX_DIV 0x3ffc0 + static struct rsnd_mod_ops adg_ops = { .name = "adg", }; +struct clk_avb { + struct clk_hw hw; + unsigned int idx; + struct rsnd_mod *mod; + /* lock reg access */ + spinlock_t *lock; +}; + +#define to_clk_avb(_hw) container_of(_hw, struct clk_avb, hw) + struct rsnd_adg { struct clk *clk[CLKMAX]; struct clk *clkout[CLKOUTMAX]; + struct clk *clkavb[AVB_CLK_NUM]; struct clk_onecell_data onecell; struct rsnd_mod mod; u32 flags; @@ -37,6 +59,7 @@ struct rsnd_adg { int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ + spinlock_t avb_lock; }; #define LRCLK_ASYNC (1 << 0) @@ -408,6 +431,239 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) } } +static struct clk *rsnd_adg_clk_src_twocell_get(struct of_phandle_args *clkspec, + void *data) +{ + unsigned int clkidx = clkspec->args[1]; + struct rsnd_adg *adg = data; + const char *type; + struct clk *clk; + + switch (clkspec->args[0]) { + case ADG_FIX: + type = "fixed"; + if (clkidx >= CLKOUTMAX) { + pr_err("Invalid %s clock index %u\n", type, + clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkout[clkidx]; + break; + case ADG_AVB: + type = "avb"; + if (clkidx >= AVB_CLK_NUM) { + pr_err("Invalid %s clock index %u\n", type, + clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkavb[clkidx]; + break; + default: + pr_err("Invalid ADG clock type %u\n", clkspec->args[0]); + return ERR_PTR(-EINVAL); + } + + return clk; +} + +static void clk_avb_div_write(struct rsnd_mod *mod, u32 data, int idx) +{ + switch (idx) { + case 0: + rsnd_mod_write(mod, AVB_CLK_DIV0, data); + break; + case 1: + rsnd_mod_write(mod, AVB_CLK_DIV1, data); + break; + case 2: + rsnd_mod_write(mod, AVB_CLK_DIV2, data); + break; + case 3: + rsnd_mod_write(mod, AVB_CLK_DIV3, data); + break; + case 4: + rsnd_mod_write(mod, AVB_CLK_DIV4, data); + break; + case 5: + rsnd_mod_write(mod, AVB_CLK_DIV5, data); + break; + case 6: + rsnd_mod_write(mod, AVB_CLK_DIV6, data); + break; + case 7: + rsnd_mod_write(mod, AVB_CLK_DIV7, data); + break; + } +} + +static u32 clk_avb_div_read(struct rsnd_mod *mod, int idx) +{ + u32 val = 0; + + switch (idx) { + case 0: + val = rsnd_mod_read(mod, AVB_CLK_DIV0); + break; + case 1: + val = rsnd_mod_read(mod, AVB_CLK_DIV1); + break; + case 2: + val = rsnd_mod_read(mod, AVB_CLK_DIV2); + break; + case 3: + val = rsnd_mod_read(mod, AVB_CLK_DIV3); + break; + case 4: + val = rsnd_mod_read(mod, AVB_CLK_DIV4); + break; + case 5: + val = rsnd_mod_read(mod, AVB_CLK_DIV5); + break; + case 6: + val = rsnd_mod_read(mod, AVB_CLK_DIV6); + break; + case 7: + val = rsnd_mod_read(mod, AVB_CLK_DIV7); + break; + } + + return val; +} + +static int clk_avb_is_enabled(struct clk_hw *hw) +{ + struct clk_avb *avb = to_clk_avb(hw); + + return rsnd_mod_read(avb->mod, AVB_CLK_CONFIG) & BIT(avb->idx); +} + +static int clk_avb_enabledisable(struct clk_hw *hw, int enable) +{ + struct clk_avb *avb = to_clk_avb(hw); + u32 val; + + spin_lock(avb->lock); + + val = rsnd_mod_read(avb->mod, AVB_CLK_CONFIG); + if (enable) + val |= BIT(avb->idx); + else + val &= ~BIT(avb->idx); + rsnd_mod_write(avb->mod, AVB_CLK_CONFIG, val); + + spin_unlock(avb->lock); + + return 0; +} + +static int clk_avb_enable(struct clk_hw *hw) +{ + return clk_avb_enabledisable(hw, 1); +} + +static void clk_avb_disable(struct clk_hw *hw) +{ + clk_avb_enabledisable(hw, 0); +} + +static unsigned long clk_avb_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_avb *avb = to_clk_avb(hw); + u32 div; + + div = clk_avb_div_read(avb->mod, avb->idx) & AVB_DIV_MASK; + if (!div) + return parent_rate; + + return parent_rate * 32 / div; +} + +static unsigned int clk_avb_calc_div(unsigned long rate, + unsigned long parent_rate) +{ + unsigned int div; + + if (!rate) + rate = 1; + + if (rate > AVB_MAX_RATE) + rate = AVB_MAX_RATE; + + div = DIV_ROUND_CLOSEST(parent_rate * 32, rate); + + if (div > AVB_MAX_DIV) + div = AVB_MAX_DIV; + + return div; +} + +static long clk_avb_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned int div = clk_avb_calc_div(rate, *parent_rate); + + return (*parent_rate * 32) / div; +} + +static int clk_avb_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_avb *avb = to_clk_avb(hw); + unsigned int div = clk_avb_calc_div(rate, parent_rate); + u32 val; + + val = clk_avb_div_read(avb->mod, avb->idx) & ~AVB_DIV_MASK; + clk_avb_div_write(avb->mod, val | div, avb->idx); + + return 0; +} + +static const struct clk_ops clk_avb_ops = { + .enable = clk_avb_enable, + .disable = clk_avb_disable, + .is_enabled = clk_avb_is_enabled, + .recalc_rate = clk_avb_recalc_rate, + .round_rate = clk_avb_round_rate, + .set_rate = clk_avb_set_rate, +}; + +static struct clk *clk_register_avb(struct device *dev, struct rsnd_mod *mod, + unsigned int id, spinlock_t *lock) +{ + struct clk_init_data init; + struct clk_avb *avb; + struct clk *clk; + char name[AVB_CLK_NAME_SIZE]; + const char *parent_name = ADG_CLK_NAME; + + avb = devm_kzalloc(dev, sizeof(*avb), GFP_KERNEL); + if (!avb) + return ERR_PTR(-ENOMEM); + + snprintf(name, AVB_CLK_NAME_SIZE, "%s%u", AVB_CLK_NAME, id); + + avb->idx = id; + avb->lock = lock; + avb->mod = mod; + + /* Register the clock. */ + init.name = name; + init.ops = &clk_avb_ops; + init.flags = CLK_IS_BASIC; + init.parent_names = &parent_name; + init.num_parents = 1; + + avb->hw.init = &init; + + /* init DIV to a valid state */ + clk_avb_div_write(avb->mod, avb->idx, AVB_MAX_DIV); + + clk = devm_clk_register(dev, &avb->hw); + + return clk; +} + static void rsnd_adg_get_clkin(struct rsnd_priv *priv, struct rsnd_adg *adg) { @@ -436,6 +692,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, unsigned long req_48kHz_rate, req_441kHz_rate; int i, req_size; const char *parent_clk_name = NULL; + struct rsnd_mod *mod = rsnd_mod_get(adg); static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", [CLKOUT1] = "audio_clkout1", @@ -540,21 +797,23 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ of_property_read_u32(np, "#clock-cells", &count); - /* - * for clkout - */ - if (!count) { + + switch (count) { + case 0: + /* + * for clkout + */ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], 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); } - } - /* - * for clkout0/1/2/3 - */ - else { + break; + case 1: + /* + * for clkout0/1/2/3 + */ for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, @@ -566,6 +825,33 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->onecell.clk_num = CLKOUTMAX; of_clk_add_provider(np, of_clk_src_onecell_get, &adg->onecell); + break; + case 2: + /* + * for clkout0/1/2/3 and avb clocks + */ + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, 0, + req_rate[0]); + if (!IS_ERR(clk)) + adg->clkout[i] = clk; + } + + for (i = 0; i < AVB_CLK_NUM; i++) { + clk = clk_register_avb(dev, mod, i, &adg->avb_lock); + if (!IS_ERR(clk)) + adg->clkavb[i] = clk; + } + + of_clk_add_provider(np, rsnd_adg_clk_src_twocell_get, adg); + + rsnd_mod_write(mod, AVB_CLK_CONFIG, AVB_DIV_EN_COM); + + break; + default: + dev_err(dev, "Invalid clock-cell %d\n", count); + break; } rsnd_adg_get_clkout_end: @@ -612,6 +898,8 @@ int rsnd_adg_probe(struct rsnd_priv *priv) if (!adg) return -ENOMEM; + spin_lock_init(&adg->avb_lock); + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, 0, 0); if (ret) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca639404f2cd..1b000d03b76e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -355,6 +355,15 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + RSND_GEN_S_REG(AVB_CLK_DIV0, 0x11c), + RSND_GEN_S_REG(AVB_CLK_DIV1, 0x120), + RSND_GEN_S_REG(AVB_CLK_DIV2, 0x124), + RSND_GEN_S_REG(AVB_CLK_DIV3, 0x128), + RSND_GEN_S_REG(AVB_CLK_DIV4, 0x12c), + RSND_GEN_S_REG(AVB_CLK_DIV5, 0x130), + RSND_GEN_S_REG(AVB_CLK_DIV6, 0x134), + RSND_GEN_S_REG(AVB_CLK_DIV7, 0x138), + RSND_GEN_S_REG(AVB_CLK_CONFIG, 0x13c), }; static const struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129af6d1..d31b8a65985f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -148,6 +148,15 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AVB_CLK_DIV0, + RSND_REG_AVB_CLK_DIV1, + RSND_REG_AVB_CLK_DIV2, + RSND_REG_AVB_CLK_DIV3, + RSND_REG_AVB_CLK_DIV4, + RSND_REG_AVB_CLK_DIV5, + RSND_REG_AVB_CLK_DIV6, + RSND_REG_AVB_CLK_DIV7, + RSND_REG_AVB_CLK_CONFIG, /* SSIU */ RSND_REG_SSI_MODE, -- 2.17.0

4 5

[alsa-devel] [PATCH linux-next v2 0/6] clk: renesas: adg: add AVB Clock
by jiada_wang＠mentor.com 04 Dec '18

04 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> on R-Car SoCs there are AVB Counter Clocks, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch-set adds 'adg' clock to R-Car Soc, and changes adg driver to register avb clocks when clock-cells of rcar_sound node is 2. --- v2: - expends adg register size and register avb clocks instead of add new clk-avb driver - Add adg clock v1: initial version Jiada Wang (2): dt-bindings: clock: add clock id for renesas adg clocks ASoC: rsnd: add avb clocks Takeshi Kihara (4): clk: renesas: r8a7795: Add ADG clock clk: renesas: r8a7796: Add ADG clock clk: renesas: r8a77990: Add ADG clocks clk: renesas: r8a77995: Add ADG clock drivers/clk/renesas/r8a7795-cpg-mssr.c | 1 + drivers/clk/renesas/r8a7796-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77990-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77995-cpg-mssr.c | 1 + include/dt-bindings/clock/renesas-adg.h | 11 + sound/soc/sh/rcar/adg.c | 306 +++++++++++++++++++++++- sound/soc/sh/rcar/gen.c | 9 + sound/soc/sh/rcar/rsnd.h | 9 + 8 files changed, 330 insertions(+), 9 deletions(-) create mode 100644 include/dt-bindings/clock/renesas-adg.h -- 2.17.0

3 6

[alsa-devel] How to inform application about sample rate changed from device.
by Chakravarthi Pradeep 04 Dec '18

04 Dec '18

Hi All, I'm working in FPGA device, which can provide raw video and pcm data. I'm using v4l2 and alsa framework for video and audio respectively. Audio flow path is using ALSA framework and video flow path is using v4l2 framework. Device can generate different audio sample rate such as 44.1KHz, 48KHz and 96KHz. VLC can render video and audio smoothly with 48KHz of sample rate, 2 channel and video with FHD(1920x1080). When device is set with 44.1KHz and 96KHz then vlc audio is breaking.I'm using vlc version as 2.2.8 and there is no way to inform how to set the sample rate for audio device. Below are queries: 1) When device is generating audio sample rate of 44.1KHz then how to inform VLC that use 44.1KHz sample rate(same is for 96KHz) 2) what is command for vlc with video and audio device setting for width=1920 height= 1080 colorspace = RV24, channel =2,sample rate 44.1KHz. Assume video device is (/dev/video0 and audio device is hw:1.0) -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

3 3

[alsa-devel] [PATCH 1/2] ALSA: hda/tegra - Probe up to 8 codecs
by Thierry Reding 03 Dec '18

03 Dec '18

From: Thierry Reding <treding(a)nvidia.com> Recent devices support more than the 4 codecs that the AZX core will probe by default. Probe up to 8 codecs to make sure all of them are enumerated. Suggested-by: Sameer Pujar <spujar(a)nvidia.com> Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 84ea6069d28b..83befd8d43e8 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -531,7 +531,7 @@ static void hda_tegra_probe_work(struct work_struct *work) goto out_free; /* create codec instances */ - err = azx_probe_codecs(chip, 0); + err = azx_probe_codecs(chip, 8); if (err < 0) goto out_free; -- 2.19.1

2 3

[alsa-devel] [PATCH v8 0/3] Huawei laptops
by Ayman Bagabas 03 Dec '18

03 Dec '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v7: * Switch to wmi_driver * Use devm to allocate and register devices * Skip registering LED subsystem if a ACPI controller method was not found Changes from v6: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Changes from v5: * Review tags Changes from v4: * Consistency in file names * How module would be enabled (Kconfig) * Match license in SPDX and MODULE_LICENSE Changes from v3: * Code formatting Changes from v2: * Support for Huawei MBX * Style and formatting issues Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 16 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 238 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 277 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.1

4 12

[alsa-devel] UAF write in usb_audio_probe
by Mathias Payer 03 Dec '18

03 Dec '18

Hi there, We are submitting a patch for an UAF write in usb_audio_probe. A malicious USB device can send a broken device configuration that will trigger a free of the underlying object before it is decremented. I'm sending the patch and KASAN report after discussing with Takashi on the security mailing list so that he can merge. The attacker needs local access to plug in a malicious USB device that replays the trace (e.g., through FaceDancer) to get read/write primitives in the kernel. For, e.g., Android or locked Desktops this becomes security critical. This bug is present in 4.14.81 through to HEAD (i.e., all versions we tested). The place where UAF is triggered is in `usb_audio_probe` in `sound/usb/card.c`. In the error handling code in `usb_audio_probe`, if the number of interfaces read from the device side is zero, `chip->card` object is freed. Unfortunately, the `chip` object is part of the `card` object, thus, when `chip->card` is freed, `chip` is also freed. However, in the error handling code, the `active` field will be written to by an `atomic_dec` operation, resulting in an UAF write. ``` static int usb_audio_probe(struct usb_interface *intf, const struct usb_device_id *usb_id) ..... __error: if (chip) { if (!chip->num_interfaces) snd_card_free(chip->card); // UAF write atomic_dec(&chip->active); } mutex_unlock(&register_mutex); return err; ``` The attacker can race the snd_card_free and the atomic_dec by attaching new USB devices (the attacker can time when what parts of the first malicious device are read then play with attaching the second device; the race is arbitrarily repeatable). The new USB device descriptors will use the recently freed memory which is then modified by the atomic_dec, resulting in an attacker-controlled decrement operation to, e.g., a USB data structure under the attacker's control. We have also attached the KASAN report from when the bug was triggered. The patch for card.c is attached. It moves the atomic_dec to above the free of the chip memory object, ensuring that, if no cards are left, the decrement happens before the free. We've also added a comment to clarify that the two objects are dependent. Thanks, Hui Peng Mathias Payer

2 1

[alsa-devel] cplay: Always write whole buffer requested by user
by GitHub pull_request - opened 03 Dec '18

03 Dec '18

alsa-project/tinycompress pull request #1 was opened from dbaluta: cplay first writes frag * fragment_size and then it only writes one fragment at a time. This means for example than if the user supplied a buffer_size it will only be used for the first write. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/1 Patch URL : https://github.com/alsa-project/tinycompress/pull/1.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[alsa-devel] [PATCH linux-next v2 5/6] dt-bindings: clock: add clock id for renesas adg clocks
by jiada_wang＠mentor.com 03 Dec '18

03 Dec '18

From: Jiada Wang <jiada_wang(a)mentor.com> This patch adds clock ID for renesas adg clocks Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- include/dt-bindings/clock/renesas-adg.h | 11 +++++++++++ 1 file changed, 11 insertions(+) create mode 100644 include/dt-bindings/clock/renesas-adg.h diff --git a/include/dt-bindings/clock/renesas-adg.h b/include/dt-bindings/clock/renesas-adg.h new file mode 100644 index 000000000000..fe30c186cef7 --- /dev/null +++ b/include/dt-bindings/clock/renesas-adg.h @@ -0,0 +1,11 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (C) 2018 Mentor Graphics inc. + */ +#ifndef __DT_BINDINGS_RENESAS_ADG_H__ +#define __DT_BINDINGS_RENESAS_ADG_H__ + +#define ADG_FIX 0 +#define ADG_AVB 1 + +#endif /* __DT_BINDINGS_RENESAS_ADG_H__ */ -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: use a generic node name for rt5631
by Lubomir Rintel 03 Dec '18

03 Dec '18

The example should follow the practice or using a generic node name instead of the precise programming model, as recommended by the DTSpec. Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> --- Documentation/devicetree/bindings/sound/rt5631.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt index 92b986ca337b..56bc85232c49 100644 --- a/Documentation/devicetree/bindings/sound/rt5631.txt +++ b/Documentation/devicetree/bindings/sound/rt5631.txt @@ -35,14 +35,14 @@ Pins on the device (for linking into audio routes): Example: -alc5631: alc5631@1a { +alc5631: audio-codec@1a { compatible = "realtek,alc5631"; reg = <0x1a>; }; or -rt5631: rt5631@1a { +rt5631: audio-codec@1a { compatible = "realtek,rt5631"; reg = <0x1a>; }; -- 2.19.1

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[alsa-devel] [PATCH 0/2] SOF dependencies
by Pierre-Louis Bossart 03 Dec '18

03 Dec '18

Two minor patches that are needed to build the rest of the SOF patches which I will post next week after the last round of internal reviews. Submitting them first. Keyon Jie (1): ALSA: HDA: export process_unsol_events() Pierre-Louis Bossart (1): ASoC: Intel: common: add ACPI matching tables for ICL include/sound/hdaudio.h | 1 + include/sound/soc-acpi-intel-match.h | 1 + sound/hda/hdac_bus.c | 7 ++-- sound/soc/intel/common/Makefile | 2 +- .../intel/common/soc-acpi-intel-icl-match.c | 32 +++++++++++++++++++ 5 files changed, 38 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-icl-match.c -- 2.17.1

2 4

[alsa-devel] alsa-utils: axfer: use second argument for transmission direction
by GitHub pull_request - opened 03 Dec '18

03 Dec '18

alsa-project/alsa-utils pull request #4 was opened from takaswie: This patchset is for axfer command in alsa-utils to use second argument for transmission direction. As a result, syntax of its command line is axfer subcommand direction options For example: $ axfer transfer playback [options] $ axfer list capture [options] This fashion requires more typing to users, however in my taste this looks good because it's easier to understand than long/short options. Additionally, this patchset applies refactoring to list subcommand. Takashi Sakamoto (5): axfer: apply refactoring to list subcommand for backward compatibility to aplay(1) axfer: apply refactoring in list subcommand for new command system axfer: use second argument in command line for transmission direction axfer: use transfer subcommand as a default for compatibility mode to aplay(1) axfer: truncate parsed arguments before operating subcommand ``` axfer/main.c | 100 +++++++++++++++++++++++++++++----------- axfer/subcmd-list.c | 85 +++++++++++++++++++++++----------- axfer/subcmd-transfer.c | 7 --- 3 files changed, 133 insertions(+), 59 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/4 Patch URL : https://github.com/alsa-project/alsa-utils/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] Regression: very quiet speakers on Thinkpad T570s
by Jeremy Cline 03 Dec '18

03 Dec '18

Hi, Some folks have reported on the Fedora bug tracker[0] that the laptop speaker volume is very low on the Thinkpad T570 when running a kernel that includes commit 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform"). alsa-info.sh from v4.15.4 (just before commit 61fcf8ece9b6 arrived in stable) and v4.19.4 with the issue present are attached to the bugzilla. I've also Cc'd Tim, who uploaded them and has the laptop in question. [0] https://bugzilla.redhat.com/show_bug.cgi?id=1554304 Regards, Jeremy

2 5

[alsa-devel] [Xen-devel][PATCH 1/3] xen: Introduce shared buffer helpers for page directory...
by Oleksandr Andrushchenko 30 Nov '18

30 Nov '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> based frontends. Currently the frontends which implement similar code for sharing big buffers between frontend and backend are para-virtualized DRM and sound drivers. Both define the same way to share grant references of a data buffer with the corresponding backend with little differences. Move shared code into a helper module, so there is a single implementation of the same functionality for all. Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> --- drivers/xen/Kconfig | 3 + drivers/xen/Makefile | 1 + drivers/xen/xen-front-pgdir-shbuf.c | 553 ++++++++++++++++++++++++++++ include/xen/xen-front-pgdir-shbuf.h | 89 +++++ 4 files changed, 646 insertions(+) create mode 100644 drivers/xen/xen-front-pgdir-shbuf.c create mode 100644 include/xen/xen-front-pgdir-shbuf.h diff --git a/drivers/xen/Kconfig b/drivers/xen/Kconfig index 815b9e9bb975..838b66a9a0e7 100644 --- a/drivers/xen/Kconfig +++ b/drivers/xen/Kconfig @@ -340,4 +340,7 @@ config XEN_SYMS config XEN_HAVE_VPMU bool +config XEN_FRONT_PGDIR_SHBUF + tristate + endmenu diff --git a/drivers/xen/Makefile b/drivers/xen/Makefile index 3e542f60f29f..c48927a58e10 100644 --- a/drivers/xen/Makefile +++ b/drivers/xen/Makefile @@ -44,3 +44,4 @@ xen-gntdev-y := gntdev.o xen-gntdev-$(CONFIG_XEN_GNTDEV_DMABUF) += gntdev-dmabuf.o xen-gntalloc-y := gntalloc.o xen-privcmd-y := privcmd.o privcmd-buf.o +obj-$(CONFIG_XEN_FRONT_PGDIR_SHBUF) += xen-front-pgdir-shbuf.o diff --git a/drivers/xen/xen-front-pgdir-shbuf.c b/drivers/xen/xen-front-pgdir-shbuf.c new file mode 100644 index 000000000000..48a658dc7ccf --- /dev/null +++ b/drivers/xen/xen-front-pgdir-shbuf.c @@ -0,0 +1,553 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen frontend/backend page directory based shared buffer + * helper module. + * + * Copyright (C) 2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/mm.h> + +#include <asm/xen/hypervisor.h> +#include <xen/balloon.h> +#include <xen/xen.h> +#include <xen/xenbus.h> +#include <xen/interface/io/ring.h> + +#include <xen/xen-front-pgdir-shbuf.h> + +#ifndef GRANT_INVALID_REF +/* + * FIXME: usage of grant reference 0 as invalid grant reference: + * grant reference 0 is valid, but never exposed to a PV driver, + * because of the fact it is already in use/reserved by the PV console. + */ +#define GRANT_INVALID_REF 0 +#endif + +/** + * This structure represents the structure of a shared page + * that contains grant references to the pages of the shared + * buffer. This structure is common to many Xen para-virtualized + * protocols at include/xen/interface/io/ + */ +struct xen_page_directory { + grant_ref_t gref_dir_next_page; + grant_ref_t gref[1]; /* Variable length */ +}; + +/** + * Shared buffer ops which are differently implemented + * depending on the allocation mode, e.g. if the buffer + * is allocated by the corresponding backend or frontend. + * Some of the operations. + */ +struct xen_front_pgdir_shbuf_ops { + /* + * Calculate number of grefs required to handle this buffer, + * e.g. if grefs are required for page directory only or the buffer + * pages as well. + */ + void (*calc_num_grefs)(struct xen_front_pgdir_shbuf *buf); + + /* Fill page directory according to para-virtual display protocol. */ + void (*fill_page_dir)(struct xen_front_pgdir_shbuf *buf); + + /* Claim grant references for the pages of the buffer. */ + int (*grant_refs_for_buffer)(struct xen_front_pgdir_shbuf *buf, + grant_ref_t *priv_gref_head, int gref_idx); + + /* Map grant references of the buffer. */ + int (*map)(struct xen_front_pgdir_shbuf *buf); + + /* Unmap grant references of the buffer. */ + int (*unmap)(struct xen_front_pgdir_shbuf *buf); +}; + +/** + * Get granted reference to the very first page of the + * page directory. Usually this is passed to the backend, + * so it can find/fill the grant references to the buffer's + * pages. + * + * \param buf shared buffer which page directory is of interest. + * \return granted reference to the very first page of the + * page directory. + */ +grant_ref_t +xen_front_pgdir_shbuf_get_dir_start(struct xen_front_pgdir_shbuf *buf) +{ + if (!buf->grefs) + return GRANT_INVALID_REF; + + return buf->grefs[0]; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_get_dir_start); + +/** + * Map granted references of the shared buffer. + * + * Depending on the shared buffer mode of allocation + * (be_alloc flag) this can either do nothing (for buffers + * shared by the frontend itself) or map the provided granted + * references onto the backing storage (buf->pages). + * + * \param buf shared buffer which grants to be maped. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_map(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->ops && buf->ops->map) + return buf->ops->map(buf); + + /* No need to map own grant references. */ + return 0; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_map); + +/** + * Unmap granted references of the shared buffer. + * + * Depending on the shared buffer mode of allocation + * (be_alloc flag) this can either do nothing (for buffers + * shared by the frontend itself) or unmap the provided granted + * references. + * + * \param buf shared buffer which grants to be unmaped. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_unmap(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->ops && buf->ops->unmap) + return buf->ops->unmap(buf); + + /* No need to unmap own grant references. */ + return 0; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_unmap); + +/** + * Free all the resources of the shared buffer. + * + * \param buf shared buffer which resources to be freed. + */ +void xen_front_pgdir_shbuf_free(struct xen_front_pgdir_shbuf *buf) +{ + if (buf->grefs) { + int i; + + for (i = 0; i < buf->num_grefs; i++) + if (buf->grefs[i] != GRANT_INVALID_REF) + gnttab_end_foreign_access(buf->grefs[i], + 0, 0UL); + } + kfree(buf->grefs); + kfree(buf->directory); +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_free); + +/* + * Number of grefs a page can hold with respect to the + * struct xen_page_directory header. + */ +#define XEN_NUM_GREFS_PER_PAGE ((PAGE_SIZE - \ + offsetof(struct xen_page_directory, \ + gref)) / sizeof(grant_ref_t)) + +/** + * Get the number of pages the page directory consumes itself. + * + * \param buf shared buffer. + */ +static int get_num_pages_dir(struct xen_front_pgdir_shbuf *buf) +{ + return DIV_ROUND_UP(buf->num_pages, XEN_NUM_GREFS_PER_PAGE); +} + +/** + * Calculate the number of grant references needed to share the buffer + * and its pages when backend allocates the buffer. + * + * \param buf shared buffer. + */ +static void backend_calc_num_grefs(struct xen_front_pgdir_shbuf *buf) +{ + /* Only for pages the page directory consumes itself. */ + buf->num_grefs = get_num_pages_dir(buf); +} + +/** + * Calculate the number of grant references needed to share the buffer + * and its pages when frontend allocates the buffer. + * + * \param buf shared buffer. + */ +static void guest_calc_num_grefs(struct xen_front_pgdir_shbuf *buf) +{ + /* + * Number of pages the page directory consumes itself + * plus grefs for the buffer pages. + */ + buf->num_grefs = get_num_pages_dir(buf) + buf->num_pages; +} + +#define xen_page_to_vaddr(page) \ + ((uintptr_t)pfn_to_kaddr(page_to_xen_pfn(page))) + +/** + * Unmap the buffer previously mapped with grant references + * provided by the backend. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int backend_unmap(struct xen_front_pgdir_shbuf *buf) +{ + struct gnttab_unmap_grant_ref *unmap_ops; + int i, ret; + + if (!buf->pages || !buf->backend_map_handles || !buf->grefs) + return 0; + + unmap_ops = kcalloc(buf->num_pages, sizeof(*unmap_ops), + GFP_KERNEL); + if (!unmap_ops) + return -ENOMEM; + + for (i = 0; i < buf->num_pages; i++) { + phys_addr_t addr; + + addr = xen_page_to_vaddr(buf->pages[i]); + gnttab_set_unmap_op(&unmap_ops[i], addr, GNTMAP_host_map, + buf->backend_map_handles[i]); + } + + ret = gnttab_unmap_refs(unmap_ops, NULL, buf->pages, + buf->num_pages); + + for (i = 0; i < buf->num_pages; i++) { + if (unlikely(unmap_ops[i].status != GNTST_okay)) + dev_err(&buf->xb_dev->dev, + "Failed to unmap page %d: %d\n", + i, unmap_ops[i].status); + } + + if (ret) + dev_err(&buf->xb_dev->dev, + "Failed to unmap grant references, ret %d", ret); + + kfree(unmap_ops); + kfree(buf->backend_map_handles); + buf->backend_map_handles = NULL; + return ret; +} + +/** + * Map the buffer with grant references provided by the backend. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int backend_map(struct xen_front_pgdir_shbuf *buf) +{ + struct gnttab_map_grant_ref *map_ops = NULL; + unsigned char *ptr; + int ret, cur_gref, cur_dir_page, cur_page, grefs_left; + + map_ops = kcalloc(buf->num_pages, sizeof(*map_ops), GFP_KERNEL); + if (!map_ops) + return -ENOMEM; + + buf->backend_map_handles = kcalloc(buf->num_pages, + sizeof(*buf->backend_map_handles), + GFP_KERNEL); + if (!buf->backend_map_handles) { + kfree(map_ops); + return -ENOMEM; + } + + /* + * Read page directory to get grefs from the backend: for external + * buffer we only allocate buf->grefs for the page directory, + * so buf->num_grefs has number of pages in the page directory itself. + */ + ptr = buf->directory; + grefs_left = buf->num_pages; + cur_page = 0; + for (cur_dir_page = 0; cur_dir_page < buf->num_grefs; cur_dir_page++) { + struct xen_page_directory *page_dir = + (struct xen_page_directory *)ptr; + int to_copy = XEN_NUM_GREFS_PER_PAGE; + + if (to_copy > grefs_left) + to_copy = grefs_left; + + for (cur_gref = 0; cur_gref < to_copy; cur_gref++) { + phys_addr_t addr; + + addr = xen_page_to_vaddr(buf->pages[cur_page]); + gnttab_set_map_op(&map_ops[cur_page], addr, + GNTMAP_host_map, + page_dir->gref[cur_gref], + buf->xb_dev->otherend_id); + cur_page++; + } + + grefs_left -= to_copy; + ptr += PAGE_SIZE; + } + ret = gnttab_map_refs(map_ops, NULL, buf->pages, buf->num_pages); + + /* Save handles even if error, so we can unmap. */ + for (cur_page = 0; cur_page < buf->num_pages; cur_page++) { + buf->backend_map_handles[cur_page] = map_ops[cur_page].handle; + if (unlikely(map_ops[cur_page].status != GNTST_okay)) + dev_err(&buf->xb_dev->dev, + "Failed to map page %d: %d\n", + cur_page, map_ops[cur_page].status); + } + + if (ret) { + dev_err(&buf->xb_dev->dev, + "Failed to map grant references, ret %d", ret); + backend_unmap(buf); + } + + kfree(map_ops); + return ret; +} + +/** + * Fill page directory with grant references to the pages of the + * page directory itself. + * + * The grant references to the buffer pages are provided by the + * backend in this case. + * + * \param buf shared buffer. + */ +static void backend_fill_page_dir(struct xen_front_pgdir_shbuf *buf) +{ + struct xen_page_directory *page_dir; + unsigned char *ptr; + int i, num_pages_dir; + + ptr = buf->directory; + num_pages_dir = get_num_pages_dir(buf); + + /* Fill only grefs for the page directory itself. */ + for (i = 0; i < num_pages_dir - 1; i++) { + page_dir = (struct xen_page_directory *)ptr; + + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + ptr += PAGE_SIZE; + } + /* Last page must say there is no more pages. */ + page_dir = (struct xen_page_directory *)ptr; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; +} + +/** + * Fill page directory with grant references to the pages of the + * page directory and the buffer we share with the backend. + * + * \param buf shared buffer. + */ +static void guest_fill_page_dir(struct xen_front_pgdir_shbuf *buf) +{ + unsigned char *ptr; + int cur_gref, grefs_left, to_copy, i, num_pages_dir; + + ptr = buf->directory; + num_pages_dir = get_num_pages_dir(buf); + + /* + * While copying, skip grefs at start, they are for pages + * granted for the page directory itself. + */ + cur_gref = num_pages_dir; + grefs_left = buf->num_pages; + for (i = 0; i < num_pages_dir; i++) { + struct xen_page_directory *page_dir = + (struct xen_page_directory *)ptr; + + if (grefs_left <= XEN_NUM_GREFS_PER_PAGE) { + to_copy = grefs_left; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; + } else { + to_copy = XEN_NUM_GREFS_PER_PAGE; + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + } + memcpy(&page_dir->gref, &buf->grefs[cur_gref], + to_copy * sizeof(grant_ref_t)); + ptr += PAGE_SIZE; + grefs_left -= to_copy; + cur_gref += to_copy; + } +} + +/** + * Grant references to the frontend's buffer pages. + * + * These will be shared with the backend, so it can + * access the buffer's data. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int guest_grant_refs_for_buffer(struct xen_front_pgdir_shbuf *buf, + grant_ref_t *priv_gref_head, + int gref_idx) +{ + int i, cur_ref, otherend_id; + + otherend_id = buf->xb_dev->otherend_id; + for (i = 0; i < buf->num_pages; i++) { + cur_ref = gnttab_claim_grant_reference(priv_gref_head); + if (cur_ref < 0) + return cur_ref; + + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, + xen_page_to_gfn(buf->pages[i]), + 0); + buf->grefs[gref_idx++] = cur_ref; + } + return 0; +} + +/** + * Grant all the references needed to share the buffer. + * + * Grant references to the page directory pages and, if + * needed, also to the pages of the shared buffer data. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int grant_references(struct xen_front_pgdir_shbuf *buf) +{ + grant_ref_t priv_gref_head; + int ret, i, j, cur_ref; + int otherend_id, num_pages_dir; + + ret = gnttab_alloc_grant_references(buf->num_grefs, &priv_gref_head); + if (ret < 0) { + dev_err(&buf->xb_dev->dev, + "Cannot allocate grant references\n"); + return ret; + } + + otherend_id = buf->xb_dev->otherend_id; + j = 0; + num_pages_dir = get_num_pages_dir(buf); + for (i = 0; i < num_pages_dir; i++) { + unsigned long frame; + + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) + return cur_ref; + + frame = xen_page_to_gfn(virt_to_page(buf->directory + + PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + if (buf->ops->grant_refs_for_buffer) { + ret = buf->ops->grant_refs_for_buffer(buf, &priv_gref_head, j); + if (ret) + return ret; + } + + gnttab_free_grant_references(priv_gref_head); + return 0; +} + +/** + * Allocate all required structures to mange shared buffer. + * + * \param buf shared buffer. + * \return zero on success or a negative number on failure. + */ +static int alloc_storage(struct xen_front_pgdir_shbuf *buf) +{ + buf->grefs = kcalloc(buf->num_grefs, sizeof(*buf->grefs), GFP_KERNEL); + if (!buf->grefs) + return -ENOMEM; + + buf->directory = kcalloc(get_num_pages_dir(buf), PAGE_SIZE, GFP_KERNEL); + if (!buf->directory) + return -ENOMEM; + + return 0; +} + +/* + * For backend allocated buffers we don't need grant_refs_for_buffer + * as those grant references are allocated at backend side. + */ +static const struct xen_front_pgdir_shbuf_ops backend_ops = { + .calc_num_grefs = backend_calc_num_grefs, + .fill_page_dir = backend_fill_page_dir, + .map = backend_map, + .unmap = backend_unmap +}; + +/* + * For locally granted references we do not need to map/unmap + * the references. + */ +static const struct xen_front_pgdir_shbuf_ops local_ops = { + .calc_num_grefs = guest_calc_num_grefs, + .fill_page_dir = guest_fill_page_dir, + .grant_refs_for_buffer = guest_grant_refs_for_buffer, +}; + +/** + * Allocate a new instance of a shared buffer. + * + * \param cfg configuration to be used while allocating a new shared buffer. + * \return zero on success or a negative number on failure. + */ +int xen_front_pgdir_shbuf_alloc(struct xen_front_pgdir_shbuf_cfg *cfg) +{ + struct xen_front_pgdir_shbuf *buf = cfg->pgdir; + int ret; + + if (cfg->be_alloc) + buf->ops = &backend_ops; + else + buf->ops = &local_ops; + buf->xb_dev = cfg->xb_dev; + buf->num_pages = cfg->num_pages; + buf->pages = cfg->pages; + + buf->ops->calc_num_grefs(buf); + + ret = alloc_storage(buf); + if (ret) + goto fail; + + ret = grant_references(buf); + if (ret) + goto fail; + + buf->ops->fill_page_dir(buf); + + return 0; + +fail: + xen_front_pgdir_shbuf_free(buf); + return ret; +} +EXPORT_SYMBOL_GPL(xen_front_pgdir_shbuf_alloc); + +MODULE_DESCRIPTION("Xen frontend/backend page directory based " + "shared buffer handling"); +MODULE_AUTHOR("Oleksandr Andrushchenko"); +MODULE_LICENSE("GPL"); diff --git a/include/xen/xen-front-pgdir-shbuf.h b/include/xen/xen-front-pgdir-shbuf.h new file mode 100644 index 000000000000..150ef7ec51ec --- /dev/null +++ b/include/xen/xen-front-pgdir-shbuf.h @@ -0,0 +1,89 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen frontend/backend page directory based shared buffer + * helper module. + * + * Copyright (C) 2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> + */ + +#ifndef __XEN_FRONT_PGDIR_SHBUF_H_ +#define __XEN_FRONT_PGDIR_SHBUF_H_ + +#include <linux/kernel.h> + +#include <xen/grant_table.h> + +struct xen_front_pgdir_shbuf_ops; + +struct xen_front_pgdir_shbuf { + /* + * Number of references granted for the backend use: + * + * - for frontend allocated/imported buffers this holds the number + * of grant references for the page directory and the pages + * of the buffer + * + * - for the buffer provided by the backend this only holds the number + * of grant references for the page directory itself as grant + * references for the buffer will be provided by the backend. + */ + int num_grefs; + grant_ref_t *grefs; + /* Page directory backing storage. */ + u8 *directory; + + /* + * Number of pages for the shared buffer itself (excluding the page + * directory). + */ + int num_pages; + /* + * Backing storage of the shared buffer: these are the pages being + * shared. + */ + struct page **pages; + + struct xenbus_device *xb_dev; + + /* These are the ops used internally depending on be_alloc mode. */ + const struct xen_front_pgdir_shbuf_ops *ops; + + /* Xen map handles for the buffer allocated by the backend. */ + grant_handle_t *backend_map_handles; +}; + +struct xen_front_pgdir_shbuf_cfg { + struct xenbus_device *xb_dev; + + /* Number of pages of the buffer backing storage. */ + int num_pages; + /* Pages of the buffer to be shared. */ + struct page **pages; + + /* + * This is allocated outside because there are use-cases when + * the buffer structure is allocated as a part of a bigger one. + */ + struct xen_front_pgdir_shbuf *pgdir; + /* + * Mode of grant reference sharing: if set then backend will share + * grant references to the buffer with the frontend. + */ + int be_alloc; +}; + +int xen_front_pgdir_shbuf_alloc(struct xen_front_pgdir_shbuf_cfg *cfg); + +grant_ref_t +xen_front_pgdir_shbuf_get_dir_start(struct xen_front_pgdir_shbuf *buf); + +int xen_front_pgdir_shbuf_map(struct xen_front_pgdir_shbuf *buf); + +int xen_front_pgdir_shbuf_unmap(struct xen_front_pgdir_shbuf *buf); + +void xen_front_pgdir_shbuf_free(struct xen_front_pgdir_shbuf *buf); + +#endif /* __XEN_FRONT_PGDIR_SHBUF_H_ */ -- 2.19.1

4 10

[alsa-devel] [V3] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 30 Nov '18

30 Nov '18

This patch added max98373_reset function to avoid amp software reset failure and code duplication. Reset verification step has been added for stable amp reset and it repeats verification maximum 3 times when it is failed. Chip revision ID is available when the amp is in the idle state which means software reset is completed well. Additional 10ms delay was added for every retrial and maximum 30ms delay can be applied. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v2: - Added commit message. Changes since v1: - Removed unusual repeat for amp software reset and verification. - Amp software reset will be performed once and it repeats verification maximum 3 times if it is failed. - Wait 10ms before every verification trial. Maximum 30ms delay will be applied to wait AMP idle state. Changes: - Created max98373_reset function to minimize code duplication. - Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. - Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. - Revision ID is available when the amp is in the idle state which means software reset is completed. - Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. - sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..9c8616a 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

1 0

[alsa-devel] [PATCH V2] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 30 Nov '18

30 Nov '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v1 : Removed unusual repeat for amp software reset and verification. Amp software reset will be performed once and it repeats verification maximum 3 times if it is failed. Wait 10ms before every verification trial. Maximum 30ms delay will be applied to wait AMP idle state. Changes : Created max98373_reset function to minimize code duplication. Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. Revision ID is available when the amp is in the idle state which means software reset is completed. Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 35 +++++++++++++++++++++++++++++------ 1 file changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..9c8616a 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,39 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +843,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

3 6

[alsa-devel] [PATCH 0/4] ASoC: merge audio-graph-scu into audio-graph - step3
by Kuninori Morimoto 30 Nov '18

30 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. This is step3 patches. After step3, the remaining patch is "merge audio-graph and audio-graph-scu" and "remove audio-graph-scu" patches, and, +1 more patch which is based on OF graph patches. I'm posting it to OF ML, but no response so far. Thus, step4 patch will be *after* OF graph patch. Kuninori Morimoto (4): ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props sound/soc/generic/audio-graph-card.c | 57 ++++++----- sound/soc/generic/audio-graph-scu-card.c | 163 ++++++++++++++++++++++--------- 2 files changed, 153 insertions(+), 67 deletions(-) -- 2.7.4

2 7

[alsa-devel] [PATCH] ALSA: pcm: Fix starvation on down_write_nonblock()
by Chanho Min 30 Nov '18

30 Nov '18

Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream") fixes deadlock for non-atomic PCM stream. But, This patch causes antother stuck. If writer is RT thread and reader is a normal thread, the reader thread will be difficult to get scheduled. It may not give chance to release readlocks and writer gets stuck for a long time if they are pinned to single cpu. The deadlock described in the previous commit is because the linux rwsem queues like a FIFO. So, we might need non-FIFO writelock, not non-block one. My suggestion is that the writer gives reader a chance to be scheduled by using the minimum msleep() instaed of spinning without blocking by writer. Also, The *_nonblock may be changed to *_nonfifo appropriately to this concept. In terms of performance, when trylock is failed, this minimum periodic msleep will have the same performance as the tick-based schedule()/wake_up_q(). Suggested-by: Wonmin Jung <wonmin.jung(a)lge.com> Signed-off-by: Chanho Min <chanho.min(a)lge.com> --- sound/core/pcm_native.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f4..bdca0e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -36,6 +36,7 @@ #include <sound/timer.h> #include <sound/minors.h> #include <linux/uio.h> +#include <linux/delay.h> #include "pcm_local.h" @@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to - * spin until it gets the lock. + * sleep until all the readers are completed without blocking by writer. */ -static inline void down_write_nonblock(struct rw_semaphore *lock) +static inline void down_write_nonfifo(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 @@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -ENOMEM; goto _nolock; } - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || substream->runtime->status->state != substream1->runtime->status->state || @@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_substream *s; int res = 0; - down_write_nonblock(&snd_pcm_link_rwsem); + down_write_nonfifo(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (!snd_pcm_stream_linked(substream)) { res = -EALREADY; -- 2.1.4

2 7

[alsa-devel] [PATCH] ALSA: pcm: Fix interval evaluation with openmin/max
by Takashi Iwai 29 Nov '18

29 Nov '18

As addressed in alsa-lib (commit b420056604f0), we need to fix the case where the evaluation of PCM interval "(x x+1]" leading to -EINVAL. After applying rules, such an interval may be translated as "(x x+1)". Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm_params.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 2dd37cada7c0..888a833d3b00 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -254,11 +254,13 @@ static inline int snd_interval_empty(const struct snd_interval *i) static inline int snd_interval_single(const struct snd_interval *i) { return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } static inline int snd_interval_value(const struct snd_interval *i) { + if (i->openmin && !i->openmax) + return i->max; return i->min; } -- 2.19.1

1 0

[alsa-devel] [PATCH v2 0/6] Introduce audio-mute LED trigger (and conversions to it)
by Takashi Iwai 29 Nov '18

29 Nov '18

Hi, this is revised patch series for adding a new LED trigger audio-mute and audio-micmute, and converting the HD-audio driver and the platform drivers to use the LED trigger instead of the ugly direct dynamic symbol binding. v2 series addressed a couple of issues Andy suggested, and changed the led class dev names to "platform::xxx" as discussed in the previous thread. The latest version of patches are found in topic/leds-trigger branch in my sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git As these are cross-tree patches, the branch above is based cleanly on v4.20-rc3. From now on, it'll stay as an immutable branch, so feel free to merge into other trees. thanks, Takashi === Takashi Iwai (6): leds: trigger: Introduce audio mute LED trigger platform/x86: dell-laptop: Add micmute LED trigger support platform/x86: thinkpad_acpi: Add audio mute LED classdev support ALSA: hda - Support led audio trigger platform/x86: dell-laptop: Drop superfluous exported function platform/x86: thinkpad_acpi: Drop superfluous exported function drivers/leds/trigger/Kconfig | 7 +++ drivers/leds/trigger/Makefile | 1 + drivers/leds/trigger/ledtrig-audio.c | 44 ++++++++++++++++++ drivers/platform/x86/Kconfig | 4 ++ drivers/platform/x86/dell-laptop.c | 27 ++++++++--- drivers/platform/x86/thinkpad_acpi.c | 67 +++++++++++++++++++++------- include/linux/dell-led.h | 7 --- include/linux/leds.h | 20 +++++++++ include/linux/thinkpad_acpi.h | 16 ------- sound/pci/hda/dell_wmi_helper.c | 48 -------------------- sound/pci/hda/hda_generic.c | 31 +++++++++++++ sound/pci/hda/hda_generic.h | 2 + sound/pci/hda/patch_realtek.c | 17 +++---- sound/pci/hda/thinkpad_helper.c | 43 +++--------------- 14 files changed, 194 insertions(+), 140 deletions(-) create mode 100644 drivers/leds/trigger/ledtrig-audio.c delete mode 100644 include/linux/dell-led.h delete mode 100644 include/linux/thinkpad_acpi.h delete mode 100644 sound/pci/hda/dell_wmi_helper.c -- 2.19.1

1 7

[alsa-devel] [PATCH] ASoc: au8540: use 64-bit arithmetic instead of 32-bit
by Yang Xiao 29 Nov '18

29 Nov '18

From: Young_X <YangX92(a)hotmail.com> Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Signed-off-by: Young_X <YangX92(a)hotmail.com> --- sound/soc/codecs/nau8540.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index e3c8cd1..4dd1a60 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -585,7 +585,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in, 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; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by niklas.morberg＠axis.com 29 Nov '18

29 Nov '18

From: Niklas Carlsson <niklasc(a)axis.com> DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

2 2

[alsa-devel] [PATCH] ALSA: hda: Add support for AMD Stoney Ridge
by Kai-Heng Feng 29 Nov '18

29 Nov '18

It's similar to other AMD audio devices, it also supports D3, which can save some power drain. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8eb2b5f51ae..3f5659064585 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2496,6 +2496,10 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD Stoney */ + { PCI_DEVICE(0x1022, 0x157a), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.17.1

2 1

[alsa-devel] User-friendly dynamic HDA pin reconfiguration
by Daniel Drake 29 Nov '18

29 Nov '18

Hi, We are working with a new consumer All-in-One platform with HDA audio (Realtek codec). Upon plugging in an audio device into one of the audio jacks, under Windows, a prompt appears which says: Which device did you plug in? And you can select from several options: Mic In / Headphone / Speaker Out Screenshot: https://imgur.com/a/mrUolHN The system vendor is asking if we can implement the same under Linux. At first glance, this sounds quite similar to what happens on other models when you plug in a headset under GNOME: a dialog pops up asking if you connected a mic, headphones, or a headset. Under the surface, such headset jacks seems to correspond to 2 separate HDA pins (one input and one output) and this dialog seems to manage which streams are enabled or muted. However, this case is different. We have experimented, and the jack corresponds to just a single HDA pin. We can manually quirk or retask this pin to be an input or an output and it works in the direction that we choose. But that's a far cry from having a nice GUI that automatically comes up asking the user how the pin should be configured. Is there any current solution or work in progress with the goal of offering this functionality? If not, brainstorming, possible solutions could include: 1. Taking the hdajackretask idea but making it user-accessible and user-friendly (currently it goes through debugfs and must be done by root). This would require the creation of a new kernel interface to expose pin reconfiguration in a nicer way, which would have to be plumbed through pulseaudio and GNOME. 2. Could ALSA ignore the pin config here and provide both input and output streams for this pin? Then pulseaudio + GUI could pick which one to use - similar to the existing Headset dialog. This would still require some work on pulseaudio and the corresponding UI, but it might be a less invasive project. Thanks, Daniel

2 2

[alsa-devel] [PATCH] ucm: Set default include path
by Takashi Iwai 29 Nov '18

29 Nov '18

Many UCM profiles include the UCM profile components under ucm/* subdirectories and thusly put <searchdir:ucm> at each place. This is rather cumbersome. This patch makes the UCM parser to set the default include path, so that each profile no longer needs to set searchdir. All the <searchdir:ucm> lines currently found in the profiles are removed gracefully, too. For the needed implementation, a new helper, _snd_config_load_with_include() is introduced. It's not exported, only for the use inside alsa-lib. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/local.h | 3 ++ src/conf.c | 32 +++++++++++++++++-- .../HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 1 - .../bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 1 - .../bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 1 - src/conf/ucm/bytcr-rt5640/HiFi.conf | 1 - .../bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 1 - .../bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 1 - .../HiFi.conf | 1 - .../bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 1 - src/conf/ucm/bytcr-rt5651/HiFi.conf | 1 - src/conf/ucm/chtnau8824/HiFi.conf | 1 - src/conf/ucm/chtrt5645/HiFi.conf | 1 - .../HiFi.conf | 1 - src/ucm/parser.c | 3 -- src/ucm/ucm_local.h | 3 ++ src/ucm/utils.c | 7 +++- 23 files changed, 41 insertions(+), 25 deletions(-) diff --git a/include/local.h b/include/local.h index b5e7b6f00ff5..03d8f0ccdc58 100644 --- a/include/local.h +++ b/include/local.h @@ -354,6 +354,9 @@ int snd_config_search_alias_hooks(snd_config_t *config, int _snd_conf_generic_id(const char *id); +int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + const char *default_include_path); + /* convenience macros */ #define ARRAY_SIZE(x) (sizeof(x) / sizeof(x[0])) diff --git a/src/conf.c b/src/conf.c index 254c485868b5..1f9bad7b3a97 100644 --- a/src/conf.c +++ b/src/conf.c @@ -1832,7 +1832,8 @@ int snd_config_top(snd_config_t **config) return _snd_config_make(config, 0, SND_CONFIG_TYPE_COMPOUND); } -static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) +static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override, + char *default_include_path) { int err; input_t input; @@ -1847,6 +1848,11 @@ static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) fd->column = 0; fd->next = NULL; INIT_LIST_HEAD(&fd->include_paths); + if (default_include_path) { + err = add_include_path(fd, default_include_path); + if (err < 0) + goto _end; + } input.current = fd; input.unget = 0; err = parse_defs(config, &input, 0, override); @@ -1915,9 +1921,29 @@ static int snd_config_load1(snd_config_t *config, snd_input_t *in, int override) */ int snd_config_load(snd_config_t *config, snd_input_t *in) { - return snd_config_load1(config, in, 0); + return snd_config_load1(config, in, 0, NULL); } +#ifndef DOC_HIDDEN +/* load config with the default include path; used internally for UCM parser */ +int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + const char *default_include_path) +{ + int err; + char *s = NULL; + + if (default_include_path) { + s = strdup(default_include_path); + if (!s) + return -ENOMEM; + } + err = snd_config_load1(config, in, 0, s); + if (err < 0) + free(s); + return err; +} +#endif + /** * \brief Loads a configuration tree and overrides existing configuration nodes. * \param config Handle to a top level configuration node. @@ -1930,7 +1956,7 @@ int snd_config_load(snd_config_t *config, snd_input_t *in) */ int snd_config_load_override(snd_config_t *config, snd_input_t *in) { - return snd_config_load1(config, in, 1); + return snd_config_load1(config, in, 1, NULL); } /** diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index facc73ac3b2b..b931b7914ce7 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index c78cdfb80a44..0e21d6e1abf9 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index 34a5d53d363c..d8ca499a5799 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 0aa0c0266412..e4fb28df546f 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index e358d4f23f12..bb31bf531633 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index 097560d02d12..9474a64e84ff 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index 8a018f63c9c6..478130e67ae4 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index 20ebe2dd36e2..54dd2517dbd7 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -1,6 +1,5 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index ca8a7d9078fb..b93e0be33cbd 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 60ef6b332ea2..ef048626732e 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index e75210d4aa7e..c324c14ad542 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index 2b7675bce831..aa1a1cde7856 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index 7a8c94bc5d6b..343aed211bb4 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index 45c7837c6fb9..6f6d74cf2e68 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index d3928e9b9976..fddd8a3ca19a 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -1,6 +1,5 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index 97a7ac7ba366..845cbcaffe5f 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index 6a3c6875f301..422f3b7eef4f 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { # ALSA PCM diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index facc73ac3b2b..b931b7914ce7 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,3 @@ -<searchdir:ucm> SectionVerb { diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 2b6f1159aa7d..5e1a8862fac1 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -34,9 +34,6 @@ #include <dirent.h> #include <limits.h> -/** The name of the environment variable containing the UCM directory */ -#define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" - /* Directories to store UCM configuration files for components, like * off-soc codecs or embedded DSPs. Components can define their own * devices and sequences, to be reused by sound cards/machines. UCM diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index ce2fc6ea18eb..8e7ac7ea48e9 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -243,3 +243,6 @@ void uc_mgr_free_sequence_element(struct sequence_element *seq); void uc_mgr_free_transition_element(struct transition_sequence *seq); void uc_mgr_free_verb(snd_use_case_mgr_t *uc_mgr); void uc_mgr_free(snd_use_case_mgr_t *uc_mgr); + +/** The name of the environment variable containing the UCM directory */ +#define ALSA_CONFIG_UCM_VAR "ALSA_CONFIG_UCM" diff --git a/src/ucm/utils.c b/src/ucm/utils.c index 18785488db07..ea5ac5dd1b01 100644 --- a/src/ucm/utils.c +++ b/src/ucm/utils.c @@ -54,6 +54,7 @@ int uc_mgr_config_load(const char *file, snd_config_t **cfg) FILE *fp; snd_input_t *in; snd_config_t *top; + const char *default_path; int err; fp = fopen(file, "r"); @@ -70,7 +71,11 @@ int uc_mgr_config_load(const char *file, snd_config_t **cfg) err = snd_config_top(&top); if (err < 0) return err; - err = snd_config_load(top, in); + + default_path = getenv(ALSA_CONFIG_UCM_VAR); + if (!default_path || !*default_path) + default_path = ALSA_CONFIG_DIR "/ucm"; + err = _snd_config_load_with_include(top, in, default_path); if (err < 0) { uc_error("could not load configuration file %s", file); snd_config_delete(top); -- 2.19.1

1 0

[alsa-devel] [PATCH v2] conf: Move UCM profile snippets into components subdirectory
by Takashi Iwai 29 Nov '18

29 Nov '18

We have placed UCM profile snippets to be included by the main config files also in the same directory, src/conf/ucm, it confuses alsaucm program that scans over all subdirectories. It thinks such a file is also the main config file, and spews errors like: % alsaucm ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr alsaucm: unable to obtain card list: No such file or directory Actually we already defined the subdirectory for such components, and they are skipped at parsing the main configs. So we just need to move the files there -- this is what's done here. One more thing done here is to add a new component subdirectory, platforms, for definitions bytcr/* that don't match with neither the existing ones (codecs nor dsps). Suggested-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 10 ++++++---- src/conf/ucm/Makefile.am | 6 ++---- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 14 ++++++------- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 14 ++++++------- src/conf/ucm/bytcr-rt5640/HiFi.conf | 20 +++++++++---------- .../bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 14 ++++++------- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 14 ++++++------- .../bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 14 ++++++------- .../HiFi.conf | 14 ++++++------- .../bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 14 ++++++------- src/conf/ucm/bytcr-rt5651/HiFi.conf | 20 +++++++++---------- src/conf/ucm/chtnau8824/HiFi.conf | 14 ++++++------- src/conf/ucm/chtrt5645/HiFi.conf | 2 +- src/conf/ucm/codecs/Makefile.am | 4 ++++ .../ucm/{ => codecs}/nau8824/EnableSeq.conf | 0 .../ucm/{ => codecs}/nau8824/HeadPhones.conf | 0 .../ucm/{ => codecs}/nau8824/HeadsetMic.conf | 0 .../ucm/{ => codecs}/nau8824/InternalMic.conf | 0 src/conf/ucm/{ => codecs}/nau8824/Makefile.am | 2 +- .../ucm/{ => codecs}/nau8824/MonoSpeaker.conf | 0 .../ucm/{ => codecs}/nau8824/Speaker.conf | 0 .../ucm/{ => codecs}/rt5640/DigitalMics.conf | 0 .../ucm/{ => codecs}/rt5640/EnableSeq.conf | 0 .../ucm/{ => codecs}/rt5640/HeadPhones.conf | 0 .../ucm/{ => codecs}/rt5640/HeadsetMic.conf | 0 .../{ => codecs}/rt5640/IN1-InternalMic.conf | 0 .../{ => codecs}/rt5640/IN3-InternalMic.conf | 0 src/conf/ucm/{ => codecs}/rt5640/Makefile.am | 2 +- .../ucm/{ => codecs}/rt5640/MonoSpeaker.conf | 0 src/conf/ucm/{ => codecs}/rt5640/Speaker.conf | 0 .../ucm/{ => codecs}/rt5651/EnableSeq.conf | 0 .../rt5651/HeadPhones-swapped.conf | 0 .../ucm/{ => codecs}/rt5651/HeadPhones.conf | 0 .../{ => codecs}/rt5651/IN1-InternalMic.conf | 0 .../{ => codecs}/rt5651/IN12-InternalMic.conf | 0 .../{ => codecs}/rt5651/IN2-InternalMic.conf | 0 .../{ => codecs}/rt5651/IN3-HeadsetMic.conf | 0 src/conf/ucm/{ => codecs}/rt5651/Makefile.am | 2 +- .../ucm/{ => codecs}/rt5651/MonoSpeaker.conf | 0 src/conf/ucm/{ => codecs}/rt5651/Speaker.conf | 0 .../HiFi.conf | 14 ++++++------- src/conf/ucm/platforms/Makefile.am | 2 ++ .../ucm/{ => platforms}/bytcr/Makefile.am | 2 +- .../bytcr/PlatformDisableSeq.conf | 0 .../bytcr/PlatformEnableSeq.conf | 0 src/ucm/parser.c | 1 + 52 files changed, 145 insertions(+), 138 deletions(-) create mode 100644 src/conf/ucm/codecs/Makefile.am rename src/conf/ucm/{ => codecs}/nau8824/EnableSeq.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/HeadPhones.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/HeadsetMic.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/Makefile.am (79%) rename src/conf/ucm/{ => codecs}/nau8824/MonoSpeaker.conf (100%) rename src/conf/ucm/{ => codecs}/nau8824/Speaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/DigitalMics.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/EnableSeq.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/HeadPhones.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/HeadsetMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/IN1-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/IN3-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/Makefile.am (82%) rename src/conf/ucm/{ => codecs}/rt5640/MonoSpeaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5640/Speaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/EnableSeq.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/HeadPhones-swapped.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/HeadPhones.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN1-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN12-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN2-InternalMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/IN3-HeadsetMic.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/Makefile.am (85%) rename src/conf/ucm/{ => codecs}/rt5651/MonoSpeaker.conf (100%) rename src/conf/ucm/{ => codecs}/rt5651/Speaker.conf (100%) create mode 100644 src/conf/ucm/platforms/Makefile.am rename src/conf/ucm/{ => platforms}/bytcr/Makefile.am (71%) rename src/conf/ucm/{ => platforms}/bytcr/PlatformDisableSeq.conf (100%) rename src/conf/ucm/{ => platforms}/bytcr/PlatformEnableSeq.conf (100%) diff --git a/configure.ac b/configure.ac index 4c9d860fd0e5..d0cab2d6cf78 100644 --- a/configure.ac +++ b/configure.ac @@ -720,7 +720,6 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/broxton-rt298/Makefile \ - src/conf/ucm/bytcr/Makefile \ src/conf/ucm/bytcr-rt5640/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/Makefile \ @@ -747,18 +746,21 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \ - src/conf/ucm/nau8824/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/PAZ00/Makefile \ src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/Makefile \ - src/conf/ucm/rt5640/Makefile \ - src/conf/ucm/rt5651/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/TECLAST-X80Pro-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ + src/conf/ucm/codecs/Makefile \ + src/conf/ucm/codecs/nau8824/Makefile \ + src/conf/ucm/codecs/rt5640/Makefile \ + src/conf/ucm/codecs/rt5651/Makefile \ + src/conf/ucm/platforms/Makefile \ + src/conf/ucm/platforms/bytcr/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 7cdc6176584a..85c15090951d 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1,8 +1,9 @@ SUBDIRS=\ +codecs \ +platforms \ ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN \ broadwell-rt286 \ broxton-rt298 \ -bytcr \ bytcr-rt5640 \ bytcr-rt5640-mono-spk-dmic1-mic \ bytcr-rt5640-mono-spk-in1-mic \ @@ -29,13 +30,10 @@ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ HDAudio-Lenovo-DualCodecs \ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ -nau8824 \ PandaBoard \ PandaBoardES \ PAZ00 \ PIPO-W2S-Defaultstring-CherryTrailCR \ -rt5640 \ -rt5651 \ SDP4430 \ skylake-rt286 \ TECLAST-X80Pro-Defaultstring-CherryTrailCR \ diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..facc73ac3b2b 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -11,18 +11,18 @@ SectionVerb { EnableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformEnableSeq.conf> - <nau8824/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/nau8824/EnableSeq.conf> ] DisableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] } -<nau8824/MonoSpeaker.conf> -<nau8824/HeadPhones.conf> +<codecs/nau8824/MonoSpeaker.conf> +<codecs/nau8824/HeadPhones.conf> -<nau8824/InternalMic.conf> -<nau8824/HeadsetMic.conf> +<codecs/nau8824/InternalMic.conf> +<codecs/nau8824/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index 938b359290d4..c78cdfb80a44 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/DigitalMics.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/DigitalMics.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index f36b245e5092..34a5d53d363c 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN1-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN1-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 1a08ededc40c..0aa0c0266412 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN3-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN3-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index 4587b2e1294e..e358d4f23f12 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/DigitalMics.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/DigitalMics.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index decd396047aa..097560d02d12 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN1-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN1-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index d243bf823511..8a018f63c9c6 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,8 +21,8 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/IN3-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/IN3-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index c833fe74845b..20ebe2dd36e2 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -6,13 +6,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformEnableSeq.conf> - <rt5640/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5640/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5640" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -21,11 +21,11 @@ SectionVerb { } } -<rt5640/Speaker.conf> -<rt5640/MonoSpeaker.conf> -<rt5640/HeadPhones.conf> +<codecs/rt5640/Speaker.conf> +<codecs/rt5640/MonoSpeaker.conf> +<codecs/rt5640/HeadPhones.conf> -<rt5640/DigitalMics.conf> -<rt5640/IN1-InternalMic.conf> -<rt5640/IN3-InternalMic.conf> -<rt5640/HeadsetMic.conf> +<codecs/rt5640/DigitalMics.conf> +<codecs/rt5640/IN1-InternalMic.conf> +<codecs/rt5640/IN3-InternalMic.conf> +<codecs/rt5640/HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index dd5b35fc02cb..ca8a7d9078fb 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN1-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN1-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 777ed7836ee5..60ef6b332ea2 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones-swapped.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones-swapped.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index 9f56c077b65a..e75210d4aa7e 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index e2a579ab2261..2b7675bce831 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN1-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN1-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index ae52e7196110..7a8c94bc5d6b 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN12-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN12-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index a73ff5f86b8f..45c7837c6fb9 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,8 +20,8 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index 8733403cf28d..d3928e9b9976 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -5,13 +5,13 @@ SectionVerb { EnableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformEnableSeq.conf> - <rt5651/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5651/EnableSeq.conf> ] DisableSequence [ cdev "hw:bytcrrt5651" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] Value { @@ -20,11 +20,11 @@ SectionVerb { } } -<rt5651/Speaker.conf> -<rt5651/MonoSpeaker.conf> -<rt5651/HeadPhones.conf> +<codecs/rt5651/Speaker.conf> +<codecs/rt5651/MonoSpeaker.conf> +<codecs/rt5651/HeadPhones.conf> -<rt5651/IN1-InternalMic.conf> -<rt5651/IN2-InternalMic.conf> -<rt5651/IN12-InternalMic.conf> -<rt5651/IN3-HeadsetMic.conf> +<codecs/rt5651/IN1-InternalMic.conf> +<codecs/rt5651/IN2-InternalMic.conf> +<codecs/rt5651/IN12-InternalMic.conf> +<codecs/rt5651/IN3-HeadsetMic.conf> diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index dc52800e5eaa..97a7ac7ba366 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -11,18 +11,18 @@ SectionVerb { EnableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformEnableSeq.conf> - <nau8824/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/nau8824/EnableSeq.conf> ] DisableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] } -<nau8824/Speaker.conf> -<nau8824/HeadPhones.conf> +<codecs/nau8824/Speaker.conf> +<codecs/nau8824/HeadPhones.conf> -<nau8824/InternalMic.conf> -<nau8824/HeadsetMic.conf> +<codecs/nau8824/InternalMic.conf> +<codecs/nau8824/HeadsetMic.conf> diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index d993f6ae9929..9a72dab9343c 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,7 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - <bytcr/PlatformEnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" diff --git a/src/conf/ucm/codecs/Makefile.am b/src/conf/ucm/codecs/Makefile.am new file mode 100644 index 000000000000..2b374acb09fd --- /dev/null +++ b/src/conf/ucm/codecs/Makefile.am @@ -0,0 +1,4 @@ +SUBDIRS=\ +rt5640 \ +rt5651 \ +nau8824 diff --git a/src/conf/ucm/nau8824/EnableSeq.conf b/src/conf/ucm/codecs/nau8824/EnableSeq.conf similarity index 100% rename from src/conf/ucm/nau8824/EnableSeq.conf rename to src/conf/ucm/codecs/nau8824/EnableSeq.conf diff --git a/src/conf/ucm/nau8824/HeadPhones.conf b/src/conf/ucm/codecs/nau8824/HeadPhones.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadPhones.conf rename to src/conf/ucm/codecs/nau8824/HeadPhones.conf diff --git a/src/conf/ucm/nau8824/HeadsetMic.conf b/src/conf/ucm/codecs/nau8824/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadsetMic.conf rename to src/conf/ucm/codecs/nau8824/HeadsetMic.conf diff --git a/src/conf/ucm/nau8824/InternalMic.conf b/src/conf/ucm/codecs/nau8824/InternalMic.conf similarity index 100% rename from src/conf/ucm/nau8824/InternalMic.conf rename to src/conf/ucm/codecs/nau8824/InternalMic.conf diff --git a/src/conf/ucm/nau8824/Makefile.am b/src/conf/ucm/codecs/nau8824/Makefile.am similarity index 79% rename from src/conf/ucm/nau8824/Makefile.am rename to src/conf/ucm/codecs/nau8824/Makefile.am index b39b8bb4c934..fd3bafe135b7 100644 --- a/src/conf/ucm/nau8824/Makefile.am +++ b/src/conf/ucm/codecs/nau8824/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/nau8824 +ucmdir = $(alsaconfigdir)/ucm/codecs/nau8824 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadsetMic.conf InternalMic.conf \ MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/nau8824/MonoSpeaker.conf b/src/conf/ucm/codecs/nau8824/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/nau8824/MonoSpeaker.conf rename to src/conf/ucm/codecs/nau8824/MonoSpeaker.conf diff --git a/src/conf/ucm/nau8824/Speaker.conf b/src/conf/ucm/codecs/nau8824/Speaker.conf similarity index 100% rename from src/conf/ucm/nau8824/Speaker.conf rename to src/conf/ucm/codecs/nau8824/Speaker.conf diff --git a/src/conf/ucm/rt5640/DigitalMics.conf b/src/conf/ucm/codecs/rt5640/DigitalMics.conf similarity index 100% rename from src/conf/ucm/rt5640/DigitalMics.conf rename to src/conf/ucm/codecs/rt5640/DigitalMics.conf diff --git a/src/conf/ucm/rt5640/EnableSeq.conf b/src/conf/ucm/codecs/rt5640/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5640/EnableSeq.conf rename to src/conf/ucm/codecs/rt5640/EnableSeq.conf diff --git a/src/conf/ucm/rt5640/HeadPhones.conf b/src/conf/ucm/codecs/rt5640/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadPhones.conf rename to src/conf/ucm/codecs/rt5640/HeadPhones.conf diff --git a/src/conf/ucm/rt5640/HeadsetMic.conf b/src/conf/ucm/codecs/rt5640/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadsetMic.conf rename to src/conf/ucm/codecs/rt5640/HeadsetMic.conf diff --git a/src/conf/ucm/rt5640/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5640/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN1-InternalMic.conf rename to src/conf/ucm/codecs/rt5640/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5640/IN3-InternalMic.conf b/src/conf/ucm/codecs/rt5640/IN3-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN3-InternalMic.conf rename to src/conf/ucm/codecs/rt5640/IN3-InternalMic.conf diff --git a/src/conf/ucm/rt5640/Makefile.am b/src/conf/ucm/codecs/rt5640/Makefile.am similarity index 82% rename from src/conf/ucm/rt5640/Makefile.am rename to src/conf/ucm/codecs/rt5640/Makefile.am index a6fbdc572529..80fa8cfc758c 100644 --- a/src/conf/ucm/rt5640/Makefile.am +++ b/src/conf/ucm/codecs/rt5640/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5640 +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5640 ucm_DATA = DigitalMics.conf EnableSeq.conf HeadPhones.conf HeadsetMic.conf \ IN1-InternalMic.conf IN3-InternalMic.conf MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/rt5640/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5640/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5640/MonoSpeaker.conf rename to src/conf/ucm/codecs/rt5640/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5640/Speaker.conf b/src/conf/ucm/codecs/rt5640/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5640/Speaker.conf rename to src/conf/ucm/codecs/rt5640/Speaker.conf diff --git a/src/conf/ucm/rt5651/EnableSeq.conf b/src/conf/ucm/codecs/rt5651/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5651/EnableSeq.conf rename to src/conf/ucm/codecs/rt5651/EnableSeq.conf diff --git a/src/conf/ucm/rt5651/HeadPhones-swapped.conf b/src/conf/ucm/codecs/rt5651/HeadPhones-swapped.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones-swapped.conf rename to src/conf/ucm/codecs/rt5651/HeadPhones-swapped.conf diff --git a/src/conf/ucm/rt5651/HeadPhones.conf b/src/conf/ucm/codecs/rt5651/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones.conf rename to src/conf/ucm/codecs/rt5651/HeadPhones.conf diff --git a/src/conf/ucm/rt5651/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN1-InternalMic.conf rename to src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN12-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN12-InternalMic.conf rename to src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN2-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN2-InternalMic.conf rename to src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN3-HeadsetMic.conf rename to src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf diff --git a/src/conf/ucm/rt5651/Makefile.am b/src/conf/ucm/codecs/rt5651/Makefile.am similarity index 85% rename from src/conf/ucm/rt5651/Makefile.am rename to src/conf/ucm/codecs/rt5651/Makefile.am index acc8ea491043..a68bd7bb9fcf 100644 --- a/src/conf/ucm/rt5651/Makefile.am +++ b/src/conf/ucm/codecs/rt5651/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5651 +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5651 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadPhones-swapped.conf \ IN1-InternalMic.conf IN2-InternalMic.conf IN12-InternalMic.conf \ IN3-HeadsetMic.conf Speaker.conf MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5651/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5651/MonoSpeaker.conf rename to src/conf/ucm/codecs/rt5651/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/Speaker.conf b/src/conf/ucm/codecs/rt5651/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5651/Speaker.conf rename to src/conf/ucm/codecs/rt5651/Speaker.conf diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..facc73ac3b2b 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -11,18 +11,18 @@ SectionVerb { EnableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformEnableSeq.conf> - <nau8824/EnableSeq.conf> + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/nau8824/EnableSeq.conf> ] DisableSequence [ cdev "hw:chtnau8824" - <bytcr/PlatformDisableSeq.conf> + <platforms/bytcr/PlatformDisableSeq.conf> ] } -<nau8824/MonoSpeaker.conf> -<nau8824/HeadPhones.conf> +<codecs/nau8824/MonoSpeaker.conf> +<codecs/nau8824/HeadPhones.conf> -<nau8824/InternalMic.conf> -<nau8824/HeadsetMic.conf> +<codecs/nau8824/InternalMic.conf> +<codecs/nau8824/HeadsetMic.conf> diff --git a/src/conf/ucm/platforms/Makefile.am b/src/conf/ucm/platforms/Makefile.am new file mode 100644 index 000000000000..1bf252f0e0e3 --- /dev/null +++ b/src/conf/ucm/platforms/Makefile.am @@ -0,0 +1,2 @@ +SUBDIRS=\ +bytcr diff --git a/src/conf/ucm/bytcr/Makefile.am b/src/conf/ucm/platforms/bytcr/Makefile.am similarity index 71% rename from src/conf/ucm/bytcr/Makefile.am rename to src/conf/ucm/platforms/bytcr/Makefile.am index f7a54cbdedcd..c53b21438ad2 100644 --- a/src/conf/ucm/bytcr/Makefile.am +++ b/src/conf/ucm/platforms/bytcr/Makefile.am @@ -1,4 +1,4 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/bytcr +ucmdir = $(alsaconfigdir)/ucm/platforms/bytcr ucm_DATA = PlatformEnableSeq.conf PlatformDisableSeq.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr/PlatformDisableSeq.conf b/src/conf/ucm/platforms/bytcr/PlatformDisableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformDisableSeq.conf rename to src/conf/ucm/platforms/bytcr/PlatformDisableSeq.conf diff --git a/src/conf/ucm/bytcr/PlatformEnableSeq.conf b/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformEnableSeq.conf rename to src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 219edb9691a6..2b6f1159aa7d 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -53,6 +53,7 @@ static const char * const component_dir[] = { "codecs", /* for off-soc codecs */ "dsps", /* for DSPs embedded in SoC */ + "platforms", /* for common platform implementations */ NULL, /* terminator */ }; -- 2.19.1

3 8

[alsa-devel] [PATCH] ALSA: hda/tegra: compatible string as shortname
by Sameer Pujar 29 Nov '18

29 Nov '18

By default HDA sound card is registered with shortname "tegra-hda". Same driver is used across tegra platforms and it is necessary to distinguish between platforms to use platform specific settings from userspace. One such example is, hdmi port on different platforms use different alsa pcm device ID. For hdmi playback to work it should open correct pcm device depending on the platform. This patch applies shortname from first compatible string provided in root node of device tree. Userspace then can use this card name to apply specific settings. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index dd7d424..84ea606 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -31,6 +31,7 @@ #include <linux/of_device.h> #include <linux/slab.h> #include <linux/time.h> +#include <linux/string.h> #include <sound/core.h> #include <sound/initval.h> @@ -344,6 +345,8 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) int err; unsigned short gcap; int irq_id = platform_get_irq(pdev, 0); + const char *sname; + struct device_node *root; err = hda_tegra_init_chip(chip, pdev); if (err) @@ -401,8 +404,23 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) return -ENODEV; } + /* driver name */ strcpy(card->driver, "tegra-hda"); - strcpy(card->shortname, "tegra-hda"); + + root = of_find_node_by_path("/"); + sname = of_get_property(root, "compatible", NULL); + of_node_put(root); + if (!sname) { + dev_err(card->dev, + "failed to get compatible property from root node\n"); + return -ENODEV; + } + /* shortname for card */ + if (strlen(sname) > sizeof(card->shortname)) + dev_info(card->dev, "truncating shortname for card\n"); + strncpy(card->shortname, sname, sizeof(card->shortname)); + + /* longname for card */ snprintf(card->longname, sizeof(card->longname), "%s at 0x%lx irq %i", card->shortname, bus->addr, bus->irq); -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 1

[alsa-devel] Subwoofers aren't working on ALC298 (Lenovo Yoga C930 Laptop)
by Jonathan Reeve 29 Nov '18

29 Nov '18

Hi Alsa Devs, I've been struggling with a sound issue for a while now. I just bought a new laptop that's known for its high-quality audio. It's billed as a Dolby Atmos speaker system, and it appears to be using a ALC298 card (or so I gathered from `aplay -l`). From what I can tell, it has tweeters in the back, and subwoofers on the bottom of the laptop. It sounds really great in Windows, but tinny and flat in Linux. I'm guessing this is because the subwoofers aren't being engaged. Here are some things I've tried so far: - Tweaking what is possible to tweak in GNOME. - Running pavucontrol. No useful options here. - Unmuting everything in alsamixer. Unmuting only some things in alsamixer. - Running hdajackretask, from the alsaTools package. It seems to correctly recognize that there are some unassigned pins, and even gives me the option of assigning them to "internal speaker (LFE)," which sounds like it could be a bass output. But when I click "apply now" or "apply boot options," the scripts fail by saying they can't find tools like md or tee. I can run the scripts it generates manually, but then the changes don't really seem to take effect. - Following the instructions from the ALSA page on the wiki, and trying all kinds of different options - All of the above, and also restarting alsa / snd_hda / pulseaudio all manually. As far as I can tell, I'm running the latest version of ALSA, since I'm on NixOS unstable, and [[https://github.com/NixOS/nixpkgs/blob/master/pkgs/os-specific/linux/alsa-fi… build script]] seems to grab the latest version. [[http://www.alsa-project.org/db/?f=8c072b3fbb139d350cd0e2c4c98cbb7736c6455a]…'s the output from my alsa-info.sh.]] I don't know enough about ALSA, or sound hardware, to know what's going on there. Is it obvious from my alsa-info which pins I should try reassigning to which channels? And does anyone know of a good way of doing this? Preferably one that doesn't use hdajackretask, which doesn't appear to work on NixOS. [[https://unix.stackexchange.com/questions/482885/how-can-i-get-my-laptops-in… also posted a few more details about this on this StackExchange question.]] I hope that getting this to work will not only help me, but others with this same laptop or speaker configuration. Thanks in advance for any help you'd be able to provide! Best, Jonathan

3 5

[alsa-devel] [RFC PATCH 0/6] ASoC:Intel:Skylake: Enable HDaudio legacy fallback
by Pierre-Louis Bossart 28 Nov '18

28 Nov '18

This patchset is provided as an RFC and should not be merged as is (Turkey break in the USA and more validation needed). This is however a good time to gather comments. This work is the result of multiple email discussions to finally provide more flexibility for distributions to chose whether to stick with the legacy HDaudio driver or to enable the SST/Skylake driver for newer platforms (required e.g. for digital microphone support) The patches add support for CoffeeLake, simplify the probe for the Skylake driver, introduce more compile-time granularity so that platforms can be selected individually and last provide a dynamic fallback mechanism when two drivers are registered for the same PCI ID. When the SOF driver is released, the same mechanism will be used to enable the SOF-legacy fallback. There will be no plans to enable an SOF->SST falback. Pierre-Louis Bossart (4): ASoC: Intel: Skylake: stop init/probe if DSP is not present ASoC: Intel: Skylake: remove useless tests on DSP presence ASoC: Intel: Skylake: Add more platform granularity ALSA: hda: add fallback capabilities for SKL+ platforms Takashi Iwai (2): ASoC: Intel: Skylake: Add CFL-S support ALSA: hda: Allow fallback binding with legacy HD-audio for Intel SKL+ include/sound/hdaudio.h | 6 + sound/pci/hda/Kconfig | 40 +++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 51 ++++++-- sound/soc/intel/Kconfig | 86 ++++++++++++-- sound/soc/intel/boards/Kconfig | 16 ++- sound/soc/intel/skylake/skl-messages.c | 8 ++ sound/soc/intel/skylake/skl.c | 154 +++++++++++++++++++------ 8 files changed, 311 insertions(+), 52 deletions(-) -- 2.17.1

3 19

[alsa-devel] [PATCH 1/1] ASoC: pcm3168a: Don't disable pcm3168a when CONFIG_PM defined
by jiada_wang＠mentor.com 28 Nov '18

28 Nov '18

From: Jiada Wang <jiada_wang(a)mentor.com> pcm3168 codec support runtime_[resume|suspend], whenever it is not active, it enters suspend mode, and it's clock and regulators will be disabled. so there is no need to disable them again in remove callback. Otherwise we got following kernel warnings, when unload pcm3168a driver [ 222.257514] unbalanced disables for amp-en-regulator [ 222.262526] ------------[ cut here ]------------ [ 222.267158] WARNING: CPU: 0 PID: 2423 at drivers/regulator/core.c:2264 _regulator_disable+0x28/0x108 [ 222.276291] Modules linked in: [ 222.279343] snd_soc_pcm3168a_i2c(-) [ 222.282916] snd_aloop [ 222.285272] arc4 [ 222.287194] wl18xx [ 222.289289] wlcore [ 222.291385] mac80211 [ 222.293654] cfg80211 [ 222.295923] aes_ce_blk [ 222.298366] crypto_simd [ 222.300896] cryptd [ 222.302992] aes_ce_cipher [ 222.305696] crc32_ce [ 222.307965] ghash_ce [ 222.310234] aes_arm64 [ 222.312590] gf128mul [ 222.314860] snd_soc_rcar [ 222.317476] sha2_ce [ 222.319658] xhci_plat_hcd [ 222.322362] sha256_arm64 [ 222.324978] xhci_hcd [ 222.327247] sha1_ce [ 222.329430] renesas_usbhs [ 222.332133] evdev [ 222.334142] sha1_generic [ 222.336758] rcar_gen3_thermal [ 222.339810] cpufreq_dt [ 222.342253] ravb_streaming(C) [ 222.345304] wlcore_sdio [ 222.347834] thermal_sys [ 222.350363] udc_core [ 222.352632] mch_core(C) [ 222.355161] usb_dmac [ 222.357430] snd_soc_pcm3168a [ 222.360394] snd_soc_ak4613 [ 222.363184] gpio_keys [ 222.365540] virt_dma [ 222.367809] nfsd [ 222.369730] ipv6 [ 222.371652] autofs4 [ 222.373834] [last unloaded: snd_soc_pcm3168a_i2c] [ 222.378629] CPU: 0 PID: 2423 Comm: rmmod Tainted: G WC 4.14.63-04798-gd456126e4a42-dirty #457 [ 222.388196] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 222.396199] task: ffff8006fa8c6200 task.stack: ffff00000a0a0000 [ 222.402117] PC is at _regulator_disable+0x28/0x108 [ 222.406906] LR is at _regulator_disable+0x28/0x108 [ 222.411695] pc : [<ffff0000083bd89c>] lr : [<ffff0000083bd89c>] pstate: 00000145 [ 222.419089] sp : ffff00000a0a3c80 [ 222.422401] x29: ffff00000a0a3c80 [ 222.425799] x28: ffff8006fa8c6200 [ 222.429199] x27: ffff0000086f1000 [ 222.432597] x26: 000000000000006a [ 222.435997] x25: 0000000000000124 [ 222.439395] x24: 0000000000000018 [ 222.442795] x23: 0000000000000006 [ 222.446193] x22: ffff8006f925d490 [ 222.449592] x21: ffff8006f9ac2068 [ 222.452991] x20: ffff8006f9ac2000 [ 222.456390] x19: 0000000000000005 [ 222.459787] x18: 000000000000000a [ 222.463186] x17: 0000000000000000 [ 222.466584] x16: 0000000000000000 [ 222.469984] x15: 000000000d3f616a [ 222.473382] x14: 0720072007200720 [ 222.476781] x13: 0720072007200720 [ 222.480179] x12: 0720072007200720 [ 222.483578] x11: 0720072007200720 [ 222.486975] x10: 0720072007200720 [ 222.490375] x9 : 0720072007200720 [ 222.493773] x8 : 07200772076f0774 [ 222.497172] x7 : 0000000000000000 [ 222.500570] x6 : 0000000000000007 [ 222.503969] x5 : 0000000000000000 [ 222.507367] x4 : 0000000000000000 [ 222.510766] x3 : 0000000000000000 [ 222.514164] x2 : c790b852091e2600 [ 222.517563] x1 : 0000000000000000 [ 222.520961] x0 : 0000000000000028 [ 222.524361] Call trace: [ 222.526805] Exception stack(0xffff00000a0a3b40 to 0xffff00000a0a3c80) [ 222.533245] 3b40: 0000000000000028 0000000000000000 c790b852091e2600 0000000000000000 [ 222.541075] 3b60: 0000000000000000 0000000000000000 0000000000000007 0000000000000000 [ 222.548905] 3b80: 07200772076f0774 0720072007200720 0720072007200720 0720072007200720 [ 222.556735] 3ba0: 0720072007200720 0720072007200720 0720072007200720 000000000d3f616a [ 222.564564] 3bc0: 0000000000000000 0000000000000000 000000000000000a 0000000000000005 [ 222.572394] 3be0: ffff8006f9ac2000 ffff8006f9ac2068 ffff8006f925d490 0000000000000006 [ 222.580224] 3c00: 0000000000000018 0000000000000124 000000000000006a ffff0000086f1000 [ 222.588053] 3c20: ffff8006fa8c6200 ffff00000a0a3c80 ffff0000083bd89c ffff00000a0a3c80 [ 222.595883] 3c40: ffff0000083bd89c 0000000000000145 0000000000000000 0000000000000000 [ 222.603713] 3c60: 0000ffffffffffff ffff00000a0a3c30 ffff00000a0a3c80 ffff0000083bd89c [ 222.611543] [<ffff0000083bd89c>] _regulator_disable+0x28/0x108 [ 222.617375] [<ffff0000083bd9c4>] regulator_disable+0x48/0x68 [ 222.623033] [<ffff0000083be8e4>] regulator_bulk_disable+0x58/0xc0 [ 222.629134] [<ffff0000007d831c>] pcm3168a_remove+0x30/0x50 [snd_soc_pcm3168a] [ 222.636270] [<ffff0000007e5010>] pcm3168a_i2c_remove+0x10/0x1c [snd_soc_pcm3168a_i2c] [ 222.644106] [<ffff0000084b9d9c>] i2c_device_remove+0x38/0x70 [ 222.649766] [<ffff00000843cd5c>] device_release_driver_internal+0xd0/0x1c0 [ 222.656640] [<ffff00000843ced8>] driver_detach+0x70/0x7c [ 222.661951] [<ffff00000843bf68>] bus_remove_driver+0x74/0xa0 [ 222.667609] [<ffff00000843d7e4>] driver_unregister+0x48/0x4c [ 222.673268] [<ffff0000084ba8dc>] i2c_del_driver+0x24/0x30 [ 222.678666] [<ffff0000007e5078>] pcm3168a_i2c_driver_exit+0x10/0xf98 [snd_soc_pcm3168a_i2c] [ 222.687019] [<ffff00000811bd28>] SyS_delete_module+0x198/0x1d4 [ 222.692850] Exception stack(0xffff00000a0a3ec0 to 0xffff00000a0a4000) [ 222.699289] 3ec0: 0000aaaafeb4b268 0000000000000800 14453f6470497100 0000fffffaa520d8 [ 222.707119] 3ee0: 0000fffffaa520d9 000000000000000a 1999999999999999 0000000000000000 [ 222.714948] 3f00: 000000000000006a 0000ffffa8f7d1d8 000000000000000a 0000000000000005 [ 222.722778] 3f20: 0000000000000000 0000000000000000 000000000000002d 0000000000000000 [ 222.730607] 3f40: 0000aaaae19b9f68 0000ffffa8f411f0 0000000000000000 0000aaaae19b9000 [ 222.738436] 3f60: 0000fffffaa533b8 0000fffffaa531f0 0000000000000000 0000000000000001 [ 222.746266] 3f80: 0000fffffaa53ec6 0000000000000000 0000aaaafeb4b200 0000aaaafeb4a010 [ 222.754096] 3fa0: 0000000000000000 0000fffffaa53130 0000aaaae199f36c 0000fffffaa53130 [ 222.761926] 3fc0: 0000ffffa8f411f8 0000000000000000 0000aaaafeb4b268 000000000000006a [ 222.769755] 3fe0: 0000000000000000 0000000000000000 0000000000000000 0000000000000000 [ 222.777589] [<ffff0000080832c0>] el0_svc_naked+0x34/0x38 [ 222.782899] ---[ end trace eaf8939a3698b1a8 ]--- [ 222.787609] Failed to disable VCCDA2: -5 [ 222.791649] ------------[ cut here ]------------ [ 222.796283] WARNING: CPU: 0 PID: 2423 at drivers/clk/clk.c:595 clk_core_disable+0xc/0x1d8 [ 222.804460] Modules linked in: [ 222.807511] snd_soc_pcm3168a_i2c(-) [ 222.811083] snd_aloop [ 222.813439] arc4 [ 222.815360] wl18xx [ 222.817456] wlcore [ 222.819551] mac80211 [ 222.821820] cfg80211 [ 222.824088] aes_ce_blk [ 222.826531] crypto_simd [ 222.829060] cryptd [ 222.831155] aes_ce_cipher [ 222.833859] crc32_ce [ 222.836127] ghash_ce [ 222.838396] aes_arm64 [ 222.840752] gf128mul [ 222.843020] snd_soc_rcar [ 222.845637] sha2_ce [ 222.847818] xhci_plat_hcd [ 222.850522] sha256_arm64 [ 222.853138] xhci_hcd [ 222.855407] sha1_ce [ 222.857589] renesas_usbhs [ 222.860292] evdev [ 222.862300] sha1_generic [ 222.864917] rcar_gen3_thermal [ 222.867968] cpufreq_dt [ 222.870410] ravb_streaming(C) [ 222.873461] wlcore_sdio [ 222.875991] thermal_sys [ 222.878520] udc_core [ 222.880789] mch_core(C) [ 222.883318] usb_dmac [ 222.885587] snd_soc_pcm3168a [ 222.888551] snd_soc_ak4613 [ 222.891341] gpio_keys [ 222.893696] virt_dma [ 222.895965] nfsd [ 222.897886] ipv6 [ 222.899808] autofs4 [ 222.901990] [last unloaded: snd_soc_pcm3168a_i2c] [ 222.906783] CPU: 0 PID: 2423 Comm: rmmod Tainted: G WC 4.14.63-04798-gd456126e4a42-dirty #457 [ 222.916349] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 222.924351] task: ffff8006fa8c6200 task.stack: ffff00000a0a0000 [ 222.930270] PC is at clk_core_disable+0xc/0x1d8 [ 222.934799] LR is at clk_core_disable_lock+0x20/0x34 [ 222.939761] pc : [<ffff0000083ab9b8>] lr : [<ffff0000083acd28>] pstate: 800001c5 [ 222.947154] sp : ffff00000a0a3cf0 [ 222.950466] x29: ffff00000a0a3cf0 [ 222.953864] x28: ffff8006fa8c6200 [ 222.957263] x27: ffff0000086f1000 [ 222.960661] x26: 000000000000006a [ 222.964061] x25: 0000000000000124 [ 222.967458] x24: 0000000000000015 [ 222.970858] x23: ffff8006f9ffa8d0 [ 222.974256] x22: ffff8006faf16480 [ 222.977655] x21: ffff0000007e7040 [ 222.981053] x20: ffff8006faadd100 [ 222.984452] x19: 0000000000000140 [ 222.987850] x18: 000000000000000a [ 222.991249] x17: 0000000000000000 [ 222.994647] x16: 0000000000000000 [ 222.998046] x15: 000000000d477819 [ 223.001444] x14: 0720072007200720 [ 223.004843] x13: 0720072007200720 [ 223.008242] x12: 0720072007200720 [ 223.011641] x11: 0720072007200720 [ 223.015039] x10: 0720072007200720 [ 223.018438] x9 : 0720072007200720 [ 223.021837] x8 : 0720072007200720 [ 223.025236] x7 : 0000000000000000 [ 223.028634] x6 : 0000000000000007 [ 223.032034] x5 : 0000000000000000 [ 223.035432] x4 : 0000000000000000 [ 223.038831] x3 : 0000000000000000 [ 223.042229] x2 : 0000000004720471 [ 223.045628] x1 : 0000000000000000 [ 223.049026] x0 : ffff8006faadd100 [ 223.052426] Call trace: [ 223.054870] Exception stack(0xffff00000a0a3bb0 to 0xffff00000a0a3cf0) [ 223.061309] 3ba0: ffff8006faadd100 0000000000000000 [ 223.069139] 3bc0: 0000000004720471 0000000000000000 0000000000000000 0000000000000000 [ 223.076969] 3be0: 0000000000000007 0000000000000000 0720072007200720 0720072007200720 [ 223.084798] 3c00: 0720072007200720 0720072007200720 0720072007200720 0720072007200720 [ 223.092628] 3c20: 0720072007200720 000000000d477819 0000000000000000 0000000000000000 [ 223.100458] 3c40: 000000000000000a 0000000000000140 ffff8006faadd100 ffff0000007e7040 [ 223.108287] 3c60: ffff8006faf16480 ffff8006f9ffa8d0 0000000000000015 0000000000000124 [ 223.116117] 3c80: 000000000000006a ffff0000086f1000 ffff8006fa8c6200 ffff00000a0a3cf0 [ 223.123947] 3ca0: ffff0000083acd28 ffff00000a0a3cf0 ffff0000083ab9b8 00000000800001c5 [ 223.131777] 3cc0: ffff00000a0a3cf0 ffff0000083acd1c 0000ffffffffffff ffff8006faadd100 [ 223.139606] 3ce0: ffff00000a0a3cf0 ffff0000083ab9b8 [ 223.144483] [<ffff0000083ab9b8>] clk_core_disable+0xc/0x1d8 [ 223.150054] [<ffff0000083acd58>] clk_disable+0x1c/0x28 [ 223.155198] [<ffff0000007d8328>] pcm3168a_remove+0x3c/0x50 [snd_soc_pcm3168a] [ 223.162334] [<ffff0000007e5010>] pcm3168a_i2c_remove+0x10/0x1c [snd_soc_pcm3168a_i2c] [ 223.170167] [<ffff0000084b9d9c>] i2c_device_remove+0x38/0x70 [ 223.175826] [<ffff00000843cd5c>] device_release_driver_internal+0xd0/0x1c0 [ 223.182700] [<ffff00000843ced8>] driver_detach+0x70/0x7c [ 223.188012] [<ffff00000843bf68>] bus_remove_driver+0x74/0xa0 [ 223.193669] [<ffff00000843d7e4>] driver_unregister+0x48/0x4c [ 223.199329] [<ffff0000084ba8dc>] i2c_del_driver+0x24/0x30 [ 223.204726] [<ffff0000007e5078>] pcm3168a_i2c_driver_exit+0x10/0xf98 [snd_soc_pcm3168a_i2c] [ 223.213079] [<ffff00000811bd28>] SyS_delete_module+0x198/0x1d4 [ 223.218909] Exception stack(0xffff00000a0a3ec0 to 0xffff00000a0a4000) [ 223.225349] 3ec0: 0000aaaafeb4b268 0000000000000800 14453f6470497100 0000fffffaa520d8 [ 223.233179] 3ee0: 0000fffffaa520d9 000000000000000a 1999999999999999 0000000000000000 [ 223.241008] 3f00: 000000000000006a 0000ffffa8f7d1d8 000000000000000a 0000000000000005 [ 223.248838] 3f20: 0000000000000000 0000000000000000 000000000000002d 0000000000000000 [ 223.256668] 3f40: 0000aaaae19b9f68 0000ffffa8f411f0 0000000000000000 0000aaaae19b9000 [ 223.264497] 3f60: 0000fffffaa533b8 0000fffffaa531f0 0000000000000000 0000000000000001 [ 223.272327] 3f80: 0000fffffaa53ec6 0000000000000000 0000aaaafeb4b200 0000aaaafeb4a010 [ 223.280157] 3fa0: 0000000000000000 0000fffffaa53130 0000aaaae199f36c 0000fffffaa53130 [ 223.287986] 3fc0: 0000ffffa8f411f8 0000000000000000 0000aaaafeb4b268 000000000000006a [ 223.295816] 3fe0: 0000000000000000 0000000000000000 0000000000000000 0000000000000000 [ 223.303648] [<ffff0000080832c0>] el0_svc_naked+0x34/0x38 [ 223.308958] ---[ end trace eaf8939a3698b1a9 ]--- [ 223.313752] ------------[ cut here ]------------ [ 223.318383] WARNING: CPU: 0 PID: 2423 at drivers/clk/clk.c:477 clk_core_unprepare+0xc/0x1ac [ 223.326733] Modules linked in: [ 223.329784] snd_soc_pcm3168a_i2c(-) [ 223.333356] snd_aloop [ 223.335712] arc4 [ 223.337633] wl18xx [ 223.339728] wlcore [ 223.341823] mac80211 [ 223.344092] cfg80211 [ 223.346360] aes_ce_blk [ 223.348803] crypto_simd [ 223.351332] cryptd [ 223.353428] aes_ce_cipher [ 223.356131] crc32_ce [ 223.358400] ghash_ce [ 223.360668] aes_arm64 [ 223.363024] gf128mul [ 223.365293] snd_soc_rcar [ 223.367909] sha2_ce [ 223.370091] xhci_plat_hcd [ 223.372794] sha256_arm64 [ 223.375410] xhci_hcd [ 223.377679] sha1_ce [ 223.379861] renesas_usbhs [ 223.382564] evdev [ 223.384572] sha1_generic [ 223.387188] rcar_gen3_thermal [ 223.390239] cpufreq_dt [ 223.392682] ravb_streaming(C) [ 223.395732] wlcore_sdio [ 223.398261] thermal_sys [ 223.400790] udc_core [ 223.403059] mch_core(C) [ 223.405588] usb_dmac [ 223.407856] snd_soc_pcm3168a [ 223.410820] snd_soc_ak4613 [ 223.413609] gpio_keys [ 223.415965] virt_dma [ 223.418234] nfsd [ 223.420155] ipv6 [ 223.422076] autofs4 [ 223.424258] [last unloaded: snd_soc_pcm3168a_i2c] [ 223.429050] CPU: 0 PID: 2423 Comm: rmmod Tainted: G WC 4.14.63-04798-gd456126e4a42-dirty #457 [ 223.438616] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 223.446618] task: ffff8006fa8c6200 task.stack: ffff00000a0a0000 [ 223.452536] PC is at clk_core_unprepare+0xc/0x1ac [ 223.457239] LR is at clk_unprepare+0x28/0x3c [ 223.461506] pc : [<ffff0000083ab5a4>] lr : [<ffff0000083ace4c>] pstate: 60000145 [ 223.468900] sp : ffff00000a0a3d00 [ 223.472211] x29: ffff00000a0a3d00 [ 223.475609] x28: ffff8006fa8c6200 [ 223.479009] x27: ffff0000086f1000 [ 223.482407] x26: 000000000000006a [ 223.485807] x25: 0000000000000124 [ 223.489205] x24: 0000000000000015 [ 223.492604] x23: ffff8006f9ffa8d0 [ 223.496003] x22: ffff8006faf16480 [ 223.499402] x21: ffff0000007e7040 [ 223.502800] x20: ffff8006faf16420 [ 223.506199] x19: ffff8006faadd100 [ 223.509597] x18: 000000000000000a [ 223.512997] x17: 0000000000000000 [ 223.516395] x16: 0000000000000000 [ 223.519794] x15: 0000000000000000 [ 223.523192] x14: 00000033fe89076c [ 223.526591] x13: 0000000000000400 [ 223.529989] x12: 0000000000000400 [ 223.533388] x11: 0000000000000000 [ 223.536786] x10: 00000000000009e0 [ 223.540185] x9 : ffff00000a0a3be0 [ 223.543583] x8 : ffff8006fa8c6c40 [ 223.546982] x7 : ffff8006fa8c6400 [ 223.550380] x6 : 0000000000000001 [ 223.553780] x5 : 0000000000000000 [ 223.557178] x4 : ffff8006fa8c6200 [ 223.560577] x3 : 0000000000000000 [ 223.563975] x2 : ffff8006fa8c6200 [ 223.567374] x1 : 0000000000000000 [ 223.570772] x0 : ffff8006faadd100 [ 223.574170] Call trace: [ 223.576615] Exception stack(0xffff00000a0a3bc0 to 0xffff00000a0a3d00) [ 223.583054] 3bc0: ffff8006faadd100 0000000000000000 ffff8006fa8c6200 0000000000000000 [ 223.590884] 3be0: ffff8006fa8c6200 0000000000000000 0000000000000001 ffff8006fa8c6400 [ 223.598714] 3c00: ffff8006fa8c6c40 ffff00000a0a3be0 00000000000009e0 0000000000000000 [ 223.606544] 3c20: 0000000000000400 0000000000000400 00000033fe89076c 0000000000000000 [ 223.614374] 3c40: 0000000000000000 0000000000000000 000000000000000a ffff8006faadd100 [ 223.622204] 3c60: ffff8006faf16420 ffff0000007e7040 ffff8006faf16480 ffff8006f9ffa8d0 [ 223.630033] 3c80: 0000000000000015 0000000000000124 000000000000006a ffff0000086f1000 [ 223.637863] 3ca0: ffff8006fa8c6200 ffff00000a0a3d00 ffff0000083ace4c ffff00000a0a3d00 [ 223.645693] 3cc0: ffff0000083ab5a4 0000000060000145 0000000000000140 ffff8006faadd100 [ 223.653523] 3ce0: 0000ffffffffffff ffff0000083ace44 ffff00000a0a3d00 ffff0000083ab5a4 [ 223.661353] [<ffff0000083ab5a4>] clk_core_unprepare+0xc/0x1ac [ 223.667103] [<ffff0000007d8330>] pcm3168a_remove+0x44/0x50 [snd_soc_pcm3168a] [ 223.674239] [<ffff0000007e5010>] pcm3168a_i2c_remove+0x10/0x1c [snd_soc_pcm3168a_i2c] [ 223.682070] [<ffff0000084b9d9c>] i2c_device_remove+0x38/0x70 [ 223.687731] [<ffff00000843cd5c>] device_release_driver_internal+0xd0/0x1c0 [ 223.694604] [<ffff00000843ced8>] driver_detach+0x70/0x7c [ 223.699915] [<ffff00000843bf68>] bus_remove_driver+0x74/0xa0 [ 223.705572] [<ffff00000843d7e4>] driver_unregister+0x48/0x4c [ 223.711230] [<ffff0000084ba8dc>] i2c_del_driver+0x24/0x30 [ 223.716628] [<ffff0000007e5078>] pcm3168a_i2c_driver_exit+0x10/0xf98 [snd_soc_pcm3168a_i2c] [ 223.724980] [<ffff00000811bd28>] SyS_delete_module+0x198/0x1d4 [ 223.730811] Exception stack(0xffff00000a0a3ec0 to 0xffff00000a0a4000) [ 223.737250] 3ec0: 0000aaaafeb4b268 0000000000000800 14453f6470497100 0000fffffaa520d8 [ 223.745079] 3ee0: 0000fffffaa520d9 000000000000000a 1999999999999999 0000000000000000 [ 223.752909] 3f00: 000000000000006a 0000ffffa8f7d1d8 000000000000000a 0000000000000005 [ 223.760739] 3f20: 0000000000000000 0000000000000000 000000000000002d 0000000000000000 [ 223.768568] 3f40: 0000aaaae19b9f68 0000ffffa8f411f0 0000000000000000 0000aaaae19b9000 [ 223.776398] 3f60: 0000fffffaa533b8 0000fffffaa531f0 0000000000000000 0000000000000001 [ 223.784227] 3f80: 0000fffffaa53ec6 0000000000000000 0000aaaafeb4b200 0000aaaafeb4a010 [ 223.792057] 3fa0: 0000000000000000 0000fffffaa53130 0000aaaae199f36c 0000fffffaa53130 [ 223.799886] 3fc0: 0000ffffa8f411f8 0000000000000000 0000aaaafeb4b268 000000000000006a [ 223.807715] 3fe0: 0000000000000000 0000000000000000 0000000000000000 0000000000000000 [ 223.815546] [<ffff0000080832c0>] el0_svc_naked+0x34/0x38 [ 223.820855] ---[ end trace eaf8939a3698b1aa ]--- Fix this issue by only disable clock and regulators in remove callback when CONFIG_PM isn't defined Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/codecs/pcm3168a.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 115f21658222..08d3fe192e65 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -758,15 +758,22 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(pcm3168a_probe); -void pcm3168a_remove(struct device *dev) +static void pcm3168a_disable(struct device *dev) { struct pcm3168a_priv *pcm3168a = dev_get_drvdata(dev); - pm_runtime_disable(dev); regulator_bulk_disable(ARRAY_SIZE(pcm3168a->supplies), - pcm3168a->supplies); + pcm3168a->supplies); clk_disable_unprepare(pcm3168a->scki); } + +void pcm3168a_remove(struct device *dev) +{ + pm_runtime_disable(dev); +#ifndef CONFIG_PM + pcm3168a_disable(dev); +#endif +} EXPORT_SYMBOL_GPL(pcm3168a_remove); #ifdef CONFIG_PM @@ -821,10 +828,7 @@ static int pcm3168a_rt_suspend(struct device *dev) regcache_cache_only(pcm3168a->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(pcm3168a->supplies), - pcm3168a->supplies); - - clk_disable_unprepare(pcm3168a->scki); + pcm3168a_disable(dev); return 0; } -- 2.17.0

2 2

[alsa-devel] [PATCH] ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()
by Kuninori Morimoto 28 Nov '18

28 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd dvc.c is using flags to avoid duplicating register for MIXer case. OTOH, commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind sound card without rebinding all drivers. Because of above patch and dvc.c flags, it can't re-register kctrl if only sound card was rebinded, because dvc is keeping old flags. (Of course it will be no problem if rsnd driver also be rebinded, but it is not purpose of above patch). This patch checks current card registered kctrl when registering. In MIXer case, it can avoid duplicate register if card already has same kctrl. In rebind case, it can re-register kctrl because card registered kctl had been removed when unbinding. This patch is updated version of commit b918f1bc7f1ce ("ASoC: rsnd: DVC kctrl sets once") Reported-by: Nguyen Viet Dung <dung.nguyen.aj(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Nguyen Viet Dung <dung.nguyen.aj(a)renesas.com> Cc: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- > Mark This patch is for v4.20-rcX sound/soc/sh/rcar/core.c | 12 ++++++++++++ sound/soc/sh/rcar/dvc.c | 8 -------- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 56469ac..031cd5b 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1476,6 +1476,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, }; int ret; + /* + * 1) Avoid duplicate register (ex. MIXer case) + * 2) re-register if card was rebinded + */ + list_for_each_entry(kctrl, &card->controls, list) { + struct rsnd_kctrl_cfg *c = kctrl->private_data; + + if (strcmp(kctrl->id.name, name) == 0 && + c->mod == mod) + return 0; + } + if (size > RSND_MAX_CHANNELS) return -EINVAL; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index d65f24b..27c3761 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -40,11 +40,8 @@ struct rsnd_dvc { struct rsnd_kctrl_cfg_s ren; /* Ramp Enable */ struct rsnd_kctrl_cfg_s rup; /* Ramp Rate Up */ struct rsnd_kctrl_cfg_s rdown; /* Ramp Rate Down */ - u32 flags; }; -#define KCTRL_INITIALIZED (1 << 0) - #define rsnd_dvc_get(priv, id) ((struct rsnd_dvc *)(priv->dvc) + id) #define rsnd_dvc_nr(priv) ((priv)->dvc_nr) @@ -227,9 +224,6 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, int channels = rsnd_rdai_channels_get(rdai); int ret; - if (rsnd_flags_has(dvc, KCTRL_INITIALIZED)) - return 0; - /* Volume */ ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? @@ -285,8 +279,6 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, if (ret < 0) return ret; - rsnd_flags_set(dvc, KCTRL_INITIALIZED); - return 0; } -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: sdm845: Add board specific dapm widgets" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: qcom: sdm845: Add board specific dapm widgets has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f1806ebb08cc28e17dc3320790a01ccdff3d17d Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Sat, 24 Nov 2018 19:09:45 +0800 Subject: [PATCH] ASoC: qcom: sdm845: Add board specific dapm widgets Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab15..95d8d4422dae 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: qcom: sdm845: Create and setup jack in init callback" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: qcom: sdm845: Create and setup jack in init callback has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 80e733fda42f511b346106351ba7b73d5a8196d4 Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Sat, 24 Nov 2018 19:09:46 +0800 Subject: [PATCH] ASoC: qcom: sdm845: Create and setup jack in init callback Add a callback for init ops on dai_link to create and setup jack. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 57 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 55 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 95d8d4422dae..43c03f8e8cdc 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -8,6 +8,8 @@ #include <linux/of_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/jack.h> +#include <uapi/linux/input-event-codes.h> #include "common.h" #include "qdsp6/q6afe.h" @@ -17,6 +19,8 @@ #define MI2S_BCLK_RATE 1536000 struct sdm845_snd_data { + struct snd_soc_jack jack; + bool jack_setup; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count; @@ -100,6 +104,54 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); + int i, rval; + + if (!pdata->jack_setup) { + struct snd_jack *jack; + + rval = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pdata->jack, NULL, 0); + + if (rval < 0) { + dev_err(card->dev, "Unable to add Headphone Jack\n"); + return rval; + } + + jack = pdata->jack.jack; + + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + pdata->jack_setup = true; + } + + for (i = 0 ; i < dai_link->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; + + component = dai->component; + rval = snd_soc_component_set_jack( + component, &pdata->jack, NULL); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set jack: %d\n", rval); + return rval; + } + } + + return 0; +} + + static int sdm845_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; @@ -220,7 +272,7 @@ static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), }; -static void sdm845_add_be_ops(struct snd_soc_card *card) +static void sdm845_add_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; int i; @@ -230,6 +282,7 @@ static void sdm845_add_be_ops(struct snd_soc_card *card) link->ops = &sdm845_be_ops; link->be_hw_params_fixup = sdm845_be_hw_params_fixup; } + link->init = sdm845_dai_init; } } @@ -264,7 +317,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) data->card = card; snd_soc_card_set_drvdata(card, data); - sdm845_add_be_ops(card); + sdm845_add_ops(card); ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "Sound card registration failed\n"); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 0/2] ASoC: omap-mcpdm: Support for handling pdmclk
by Peter Ujfalusi 28 Nov '18

28 Nov '18

Hi, This series is on top of the pm_qos changes I have sent earlier today. I have sent this originally two years ago: https://patchwork.kernel.org/patch/9119551/ But needed to revert it as the MFD and clk changes were not merged in time: https://patchwork.kernel.org/patch/9287431/ The dts patches are already upstream, only the patch for McPDM is missing. I have separated the binding document update and the driver patch and also added the return value check for the clk_prepare_enable() as per Mark's comment. Regards, Peter --- Peter Ujfalusi (2): dt-bindings: sound: omap-mcpdm: Update documentation for pdmclk ASoC: omap-mcpdm: Add support for pdmclk clock handling .../devicetree/bindings/sound/omap-mcpdm.txt | 10 ++++++++ sound/soc/omap/omap-mcpdm.c | 23 +++++++++++++++++++ 2 files changed, 33 insertions(+) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

4 7

[alsa-devel] [PATCH v5 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 28 Nov '18

28 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. This series add a mode-switch delay which will prevent the clock from being turned off without complying with manufacturer timing specifications. Also introduces a module parameter for the wakeup_delay parameter. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message v3: Updated commit message and introduced module param for wakeup_delay v4: Addressed parameter name to align with existing parameter. Added dt entry in Documentation v5: Validate modeswitch_delay to ensure sensible value Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay Documentation/devicetree/bindings/sound/dmic.txt | 2 ++ sound/soc/codecs/dmic.c | 40 ++++++++++++++++++++++++ 2 files changed, 42 insertions(+) -- 1.9.1

2 3

[alsa-devel] Applied "ASoC: dmic: introduce module_param wakeup_delay" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: dmic: introduce module_param wakeup_delay has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f6f30a609c526dbf6d59490a8c85adaf6ac9b0fa Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Wed, 28 Nov 2018 12:22:46 +0530 Subject: [PATCH] ASoC: dmic: introduce module_param wakeup_delay Introducing a module param for wakeup_delay in order to align with modeswitch_delay parameter. With this change, both wakeup_delay and modeswitch_delay parameters can be passed as module parameters. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/dmic.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index f4eb0a438a3a..da921da50ef0 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -34,6 +34,9 @@ static int modeswitch_delay; module_param(modeswitch_delay, uint, 0644); +static int wakeup_delay; +module_param(wakeup_delay, uint, 0644); + struct dmic { struct gpio_desc *gpio_en; int wakeup_delay; @@ -115,6 +118,8 @@ static int dmic_component_probe(struct snd_soc_component *component) &dmic->wakeup_delay); device_property_read_u32(component->dev, "modeswitch-delay-ms", &dmic->modeswitch_delay); + if (wakeup_delay) + dmic->wakeup_delay = wakeup_delay; if (modeswitch_delay) dmic->modeswitch_delay = modeswitch_delay; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: Make DSP replies more human readable" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: Intel: Skylake: Make DSP replies more human readable has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From aa15679b2dc898049e9117fbe3ddda0b50fa52d2 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> Date: Mon, 29 Jan 2018 21:45:59 +0530 Subject: [PATCH] ASoC: Intel: Skylake: Make DSP replies more human readable Add more meaning to the IPC replies for easy debugging. Replace the switch case with a lookup table to lookup for the IPC replies and print in human readable form. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-sst-ipc.c | 44 ++++++++++++++++++++------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 5234fafb758a..8708755a8f9a 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -392,18 +392,43 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, return 0; } -static int skl_ipc_set_reply_error_code(u32 reply) +struct skl_ipc_err_map { + const char *msg; + enum skl_ipc_glb_reply reply; + int err; +}; + +static struct skl_ipc_err_map skl_err_map[] = { + {"DSP out of memory", IPC_GLB_REPLY_OUT_OF_MEMORY, -ENOMEM}, + {"DSP busy", IPC_GLB_REPLY_BUSY, -EBUSY}, +}; + +static int skl_ipc_set_reply_error_code(struct sst_generic_ipc *ipc, u32 reply) { - switch (reply) { - case IPC_GLB_REPLY_OUT_OF_MEMORY: - return -ENOMEM; + int i; - case IPC_GLB_REPLY_BUSY: - return -EBUSY; + for (i = 0; i < ARRAY_SIZE(skl_err_map); i++) { + if (skl_err_map[i].reply == reply) + break; + } - default: + if (i == ARRAY_SIZE(skl_err_map)) { + dev_err(ipc->dev, "ipc FW reply: %d FW Error Code: %u\n", + reply, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); return -EINVAL; } + + if (skl_err_map[i].err < 0) + dev_err(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", + skl_err_map[i].msg, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + else + dev_info(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", + skl_err_map[i].msg, + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + + return skl_err_map[i].err; } void skl_ipc_process_reply(struct sst_generic_ipc *ipc, @@ -441,10 +466,7 @@ void skl_ipc_process_reply(struct sst_generic_ipc *ipc, } } else { - msg->errno = skl_ipc_set_reply_error_code(reply); - dev_err(ipc->dev, "ipc FW reply: reply=%d\n", reply); - dev_err(ipc->dev, "FW Error Code: %u\n", - ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + msg->errno = skl_ipc_set_reply_error_code(ipc, reply); switch (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { case IPC_GLB_LOAD_MULTIPLE_MODS: case IPC_GLB_LOAD_LIBRARY: -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC" to the asoc tree
by Mark Brown 28 Nov '18

28 Nov '18

The patch ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 61f94ee4a7435c35d78b22e4ae6e0551908000ae Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> Date: Mon, 29 Jan 2018 21:46:00 +0530 Subject: [PATCH] ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC If mclk/sclk is already running, FW responds with IPC reply MCLK/SCLK already running. Add these to the IPC reply lookup table. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-sst-ipc.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 8708755a8f9a..9f3ce73593ae 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -249,6 +249,8 @@ enum skl_ipc_glb_reply { IPC_GLB_REPLY_INVALID_CONFIG_DATA_LEN = 121, IPC_GLB_REPLY_GATEWAY_NOT_INITIALIZED = 140, IPC_GLB_REPLY_GATEWAY_NOT_EXIST = 141, + IPC_GLB_REPLY_SCLK_ALREADY_RUNNING = 150, + IPC_GLB_REPLY_MCLK_ALREADY_RUNNING = 151, IPC_GLB_REPLY_PPL_NOT_INITIALIZED = 160, IPC_GLB_REPLY_PPL_NOT_EXIST = 161, @@ -401,6 +403,10 @@ struct skl_ipc_err_map { static struct skl_ipc_err_map skl_err_map[] = { {"DSP out of memory", IPC_GLB_REPLY_OUT_OF_MEMORY, -ENOMEM}, {"DSP busy", IPC_GLB_REPLY_BUSY, -EBUSY}, + {"SCLK already running", IPC_GLB_REPLY_SCLK_ALREADY_RUNNING, + IPC_GLB_REPLY_SCLK_ALREADY_RUNNING}, + {"MCLK already running", IPC_GLB_REPLY_MCLK_ALREADY_RUNNING, + IPC_GLB_REPLY_MCLK_ALREADY_RUNNING}, }; static int skl_ipc_set_reply_error_code(struct sst_generic_ipc *ipc, u32 reply) -- 2.19.1

1 0

[alsa-devel] [PATCH 0/4] ASoC; davinci-mcasp: Pin handling updates
by Peter Ujfalusi 28 Nov '18

28 Nov '18

Hi, The series will improve the McASP pin handling: - To avoid non configured clocks leaking out from McASP - make sure that the AXR pin (TX) state is correct in all scenarios - Allow user configurable DISMOD for the tx pin instead of hardwiring it in the code to low The DISMOD configuration is needed when the codec requires the TX line to be high during inactive slots (mu-law codecs for example). Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Clear TXSTAT register before activating serializers ASoC: davinci-mcasp: Update PDIR (pin direction) register handling bindings: sound: davinci-mcasp: Document dismod optional property ASoC: davinci-mcasp: Implement configurable dismod handling .../bindings/sound/davinci-mcasp-audio.txt | 5 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 109 +++++++++++++++--- sound/soc/davinci/davinci-mcasp.h | 30 ++--- 4 files changed, 112 insertions(+), 33 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] Pioneer Interface 2 support?
by Daniel James 28 Nov '18

28 Nov '18

Hi folks, A user of the PiDeck project is interested in helping to add ALSA support for this USB audio interface: https://www.pioneerdj.com/en-gb/product/software/interface-2/audio-interfac… Some initial information is here: https://github.com/pideck/pideck/issues/85 Is there anyone willing to hack on this please? Does the info script available at http://www.alsa-project.org/main/index.php/Help_To_Debug provide additional details that would be useful to have? Thanks! Daniel

2 2

[alsa-devel] [PATCH 1/4] ASoC: qcom: sdm845: Add board specific dapm widgets
by Cheng-Yi Chiang 28 Nov '18

28 Nov '18

Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.20.0.rc0.387.gc7a69e6b6c-goog

2 8

[alsa-devel] HDA headset button support
by Kailang 28 Nov '18

28 Nov '18

Hi Takashi, New Chrome platform need to support HS button function. I sent it to Intel chrome team already. Could you take a look for attach patch? I think it maybe could upstream. It just waiting for test by Intel Chrome team. Thanks. BR, Kailang

2 1

[alsa-devel] [alsa-lib][PATCH v2] conf: USB-Audio: Add Dell WD19 Dock in the IEC958 blacklist
by Hui Wang 28 Nov '18

28 Nov '18

This Dock doesn't have IEC958 physical output, so add it to the blacklist to prevent it being opened. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- In the v2, the 2nd patch of adding wd19 ucm is dropped. here only adding wd19 into the blacklist, I don't know if wd15 has this issue or not, if it has, will adding wd15 to the blacklist too. src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 05779c47..3a12a729 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -60,6 +60,7 @@ USB-Audio.pcm.iec958_device { "SWTOR Gaming Headset by Razer" 999 "USB Device 0x46d_0x821" 999 "USB Device 0x46d_0x992" 999 + "WD19 Dock" 999 } # Second iec958 device number, if any. -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Add vendor and product name for Dell WD19 Dock
by Hui Wang 28 Nov '18

28 Nov '18

Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide useful string for the vendor and product names too. In order to share the UCM with WD15, here we keep the profile_name same as the WD15. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/usb/quirks-table.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 849953e5775c..37fc0447c071 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* Dell WD19 Dock */ +{ + USB_DEVICE(0x0bda, 0x402e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD19 Dock", + .profile_name = "Dell-WD15-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.17.1

2 1

[alsa-devel] [PATCH] conf: Move UCM profile snippets into ucm-includes subdirectory
by Takashi Iwai 28 Nov '18

28 Nov '18

We have placed UCM profile snippets to be included by the main config files also in the same directory, src/conf/ucm, it confuses alsaucm program that scans over all subdirectories. It thinks such a file is also the main config file, and spews errors like: % alsaucm ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr alsaucm: unable to obtain card list: No such file or directory For avoiding this, move UCM profile snippets into the own root directory, src/conf/ucm-includes, and include them by changing the include with "<searchdir:ucm-includes>". Suggested-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 9 +++++---- src/conf/Makefile.am | 2 +- src/conf/ucm-includes/Makefile.am | 5 +++++ src/conf/{ucm => ucm-includes}/bytcr/Makefile.am | 2 +- .../{ucm => ucm-includes}/bytcr/PlatformDisableSeq.conf | 0 .../{ucm => ucm-includes}/bytcr/PlatformEnableSeq.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/EnableSeq.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/HeadPhones.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/HeadsetMic.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/InternalMic.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/Makefile.am | 2 +- src/conf/{ucm => ucm-includes}/nau8824/MonoSpeaker.conf | 0 src/conf/{ucm => ucm-includes}/nau8824/Speaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/DigitalMics.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/EnableSeq.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/HeadPhones.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/HeadsetMic.conf | 0 .../{ucm => ucm-includes}/rt5640/IN1-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5640/IN3-InternalMic.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/Makefile.am | 2 +- src/conf/{ucm => ucm-includes}/rt5640/MonoSpeaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5640/Speaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/EnableSeq.conf | 0 .../{ucm => ucm-includes}/rt5651/HeadPhones-swapped.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/HeadPhones.conf | 0 .../{ucm => ucm-includes}/rt5651/IN1-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5651/IN12-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5651/IN2-InternalMic.conf | 0 .../{ucm => ucm-includes}/rt5651/IN3-HeadsetMic.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/Makefile.am | 2 +- src/conf/{ucm => ucm-includes}/rt5651/MonoSpeaker.conf | 0 src/conf/{ucm => ucm-includes}/rt5651/Speaker.conf | 0 src/conf/ucm/Makefile.am | 4 ---- .../ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5640/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf | 2 +- .../bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf | 2 +- src/conf/ucm/bytcr-rt5651/HiFi.conf | 2 +- src/conf/ucm/chtnau8824/HiFi.conf | 2 +- .../ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf | 2 +- 50 files changed, 32 insertions(+), 30 deletions(-) create mode 100644 src/conf/ucm-includes/Makefile.am rename src/conf/{ucm => ucm-includes}/bytcr/Makefile.am (72%) rename src/conf/{ucm => ucm-includes}/bytcr/PlatformDisableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/bytcr/PlatformEnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/EnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/HeadPhones.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/HeadsetMic.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/Makefile.am (78%) rename src/conf/{ucm => ucm-includes}/nau8824/MonoSpeaker.conf (100%) rename src/conf/{ucm => ucm-includes}/nau8824/Speaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/DigitalMics.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/EnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/HeadPhones.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/HeadsetMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/IN1-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/IN3-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/Makefile.am (81%) rename src/conf/{ucm => ucm-includes}/rt5640/MonoSpeaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5640/Speaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/EnableSeq.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/HeadPhones-swapped.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/HeadPhones.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN1-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN12-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN2-InternalMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/IN3-HeadsetMic.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/Makefile.am (84%) rename src/conf/{ucm => ucm-includes}/rt5651/MonoSpeaker.conf (100%) rename src/conf/{ucm => ucm-includes}/rt5651/Speaker.conf (100%) diff --git a/configure.ac b/configure.ac index 4c9d860fd0e5..0b7a1bdc42fa 100644 --- a/configure.ac +++ b/configure.ac @@ -720,7 +720,6 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/broxton-rt298/Makefile \ - src/conf/ucm/bytcr/Makefile \ src/conf/ucm/bytcr-rt5640/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/Makefile \ src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/Makefile \ @@ -747,18 +746,20 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \ - src/conf/ucm/nau8824/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ src/conf/ucm/PAZ00/Makefile \ src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/Makefile \ - src/conf/ucm/rt5640/Makefile \ - src/conf/ucm/rt5651/Makefile \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/TECLAST-X80Pro-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ + src/conf/ucm-includes/Makefile \ + src/conf/ucm-includes/bytcr/Makefile \ + src/conf/ucm-includes/nau8824/Makefile \ + src/conf/ucm-includes/rt5640/Makefile \ + src/conf/ucm-includes/rt5651/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/Makefile.am b/src/conf/Makefile.am index 2b46f95c2cb7..4a4022e9bc1b 100644 --- a/src/conf/Makefile.am +++ b/src/conf/Makefile.am @@ -1,7 +1,7 @@ SUBDIRS=cards pcm if BUILD_UCM -SUBDIRS += ucm +SUBDIRS += ucm ucm-includes endif if BUILD_TOPOLOGY SUBDIRS += topology diff --git a/src/conf/ucm-includes/Makefile.am b/src/conf/ucm-includes/Makefile.am new file mode 100644 index 000000000000..f94bba7b837f --- /dev/null +++ b/src/conf/ucm-includes/Makefile.am @@ -0,0 +1,5 @@ +SUBDIRS=\ +bytcr \ +rt5640 \ +rt5651 \ +nau8824 diff --git a/src/conf/ucm/bytcr/Makefile.am b/src/conf/ucm-includes/bytcr/Makefile.am similarity index 72% rename from src/conf/ucm/bytcr/Makefile.am rename to src/conf/ucm-includes/bytcr/Makefile.am index f7a54cbdedcd..f9565a53eaf2 100644 --- a/src/conf/ucm/bytcr/Makefile.am +++ b/src/conf/ucm-includes/bytcr/Makefile.am @@ -1,4 +1,4 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/bytcr +ucmdir = $(alsaconfigdir)/ucm-includes/bytcr ucm_DATA = PlatformEnableSeq.conf PlatformDisableSeq.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr/PlatformDisableSeq.conf b/src/conf/ucm-includes/bytcr/PlatformDisableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformDisableSeq.conf rename to src/conf/ucm-includes/bytcr/PlatformDisableSeq.conf diff --git a/src/conf/ucm/bytcr/PlatformEnableSeq.conf b/src/conf/ucm-includes/bytcr/PlatformEnableSeq.conf similarity index 100% rename from src/conf/ucm/bytcr/PlatformEnableSeq.conf rename to src/conf/ucm-includes/bytcr/PlatformEnableSeq.conf diff --git a/src/conf/ucm/nau8824/EnableSeq.conf b/src/conf/ucm-includes/nau8824/EnableSeq.conf similarity index 100% rename from src/conf/ucm/nau8824/EnableSeq.conf rename to src/conf/ucm-includes/nau8824/EnableSeq.conf diff --git a/src/conf/ucm/nau8824/HeadPhones.conf b/src/conf/ucm-includes/nau8824/HeadPhones.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadPhones.conf rename to src/conf/ucm-includes/nau8824/HeadPhones.conf diff --git a/src/conf/ucm/nau8824/HeadsetMic.conf b/src/conf/ucm-includes/nau8824/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/nau8824/HeadsetMic.conf rename to src/conf/ucm-includes/nau8824/HeadsetMic.conf diff --git a/src/conf/ucm/nau8824/InternalMic.conf b/src/conf/ucm-includes/nau8824/InternalMic.conf similarity index 100% rename from src/conf/ucm/nau8824/InternalMic.conf rename to src/conf/ucm-includes/nau8824/InternalMic.conf diff --git a/src/conf/ucm/nau8824/Makefile.am b/src/conf/ucm-includes/nau8824/Makefile.am similarity index 78% rename from src/conf/ucm/nau8824/Makefile.am rename to src/conf/ucm-includes/nau8824/Makefile.am index b39b8bb4c934..aa65f4bd9b53 100644 --- a/src/conf/ucm/nau8824/Makefile.am +++ b/src/conf/ucm-includes/nau8824/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/nau8824 +ucmdir = $(alsaconfigdir)/ucm-includes/nau8824 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadsetMic.conf InternalMic.conf \ MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/nau8824/MonoSpeaker.conf b/src/conf/ucm-includes/nau8824/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/nau8824/MonoSpeaker.conf rename to src/conf/ucm-includes/nau8824/MonoSpeaker.conf diff --git a/src/conf/ucm/nau8824/Speaker.conf b/src/conf/ucm-includes/nau8824/Speaker.conf similarity index 100% rename from src/conf/ucm/nau8824/Speaker.conf rename to src/conf/ucm-includes/nau8824/Speaker.conf diff --git a/src/conf/ucm/rt5640/DigitalMics.conf b/src/conf/ucm-includes/rt5640/DigitalMics.conf similarity index 100% rename from src/conf/ucm/rt5640/DigitalMics.conf rename to src/conf/ucm-includes/rt5640/DigitalMics.conf diff --git a/src/conf/ucm/rt5640/EnableSeq.conf b/src/conf/ucm-includes/rt5640/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5640/EnableSeq.conf rename to src/conf/ucm-includes/rt5640/EnableSeq.conf diff --git a/src/conf/ucm/rt5640/HeadPhones.conf b/src/conf/ucm-includes/rt5640/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadPhones.conf rename to src/conf/ucm-includes/rt5640/HeadPhones.conf diff --git a/src/conf/ucm/rt5640/HeadsetMic.conf b/src/conf/ucm-includes/rt5640/HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5640/HeadsetMic.conf rename to src/conf/ucm-includes/rt5640/HeadsetMic.conf diff --git a/src/conf/ucm/rt5640/IN1-InternalMic.conf b/src/conf/ucm-includes/rt5640/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN1-InternalMic.conf rename to src/conf/ucm-includes/rt5640/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5640/IN3-InternalMic.conf b/src/conf/ucm-includes/rt5640/IN3-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5640/IN3-InternalMic.conf rename to src/conf/ucm-includes/rt5640/IN3-InternalMic.conf diff --git a/src/conf/ucm/rt5640/Makefile.am b/src/conf/ucm-includes/rt5640/Makefile.am similarity index 81% rename from src/conf/ucm/rt5640/Makefile.am rename to src/conf/ucm-includes/rt5640/Makefile.am index a6fbdc572529..2cbea4da4543 100644 --- a/src/conf/ucm/rt5640/Makefile.am +++ b/src/conf/ucm-includes/rt5640/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5640 +ucmdir = $(alsaconfigdir)/ucm-includes/rt5640 ucm_DATA = DigitalMics.conf EnableSeq.conf HeadPhones.conf HeadsetMic.conf \ IN1-InternalMic.conf IN3-InternalMic.conf MonoSpeaker.conf Speaker.conf EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/rt5640/MonoSpeaker.conf b/src/conf/ucm-includes/rt5640/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5640/MonoSpeaker.conf rename to src/conf/ucm-includes/rt5640/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5640/Speaker.conf b/src/conf/ucm-includes/rt5640/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5640/Speaker.conf rename to src/conf/ucm-includes/rt5640/Speaker.conf diff --git a/src/conf/ucm/rt5651/EnableSeq.conf b/src/conf/ucm-includes/rt5651/EnableSeq.conf similarity index 100% rename from src/conf/ucm/rt5651/EnableSeq.conf rename to src/conf/ucm-includes/rt5651/EnableSeq.conf diff --git a/src/conf/ucm/rt5651/HeadPhones-swapped.conf b/src/conf/ucm-includes/rt5651/HeadPhones-swapped.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones-swapped.conf rename to src/conf/ucm-includes/rt5651/HeadPhones-swapped.conf diff --git a/src/conf/ucm/rt5651/HeadPhones.conf b/src/conf/ucm-includes/rt5651/HeadPhones.conf similarity index 100% rename from src/conf/ucm/rt5651/HeadPhones.conf rename to src/conf/ucm-includes/rt5651/HeadPhones.conf diff --git a/src/conf/ucm/rt5651/IN1-InternalMic.conf b/src/conf/ucm-includes/rt5651/IN1-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN1-InternalMic.conf rename to src/conf/ucm-includes/rt5651/IN1-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN12-InternalMic.conf b/src/conf/ucm-includes/rt5651/IN12-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN12-InternalMic.conf rename to src/conf/ucm-includes/rt5651/IN12-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN2-InternalMic.conf b/src/conf/ucm-includes/rt5651/IN2-InternalMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN2-InternalMic.conf rename to src/conf/ucm-includes/rt5651/IN2-InternalMic.conf diff --git a/src/conf/ucm/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm-includes/rt5651/IN3-HeadsetMic.conf similarity index 100% rename from src/conf/ucm/rt5651/IN3-HeadsetMic.conf rename to src/conf/ucm-includes/rt5651/IN3-HeadsetMic.conf diff --git a/src/conf/ucm/rt5651/Makefile.am b/src/conf/ucm-includes/rt5651/Makefile.am similarity index 84% rename from src/conf/ucm/rt5651/Makefile.am rename to src/conf/ucm-includes/rt5651/Makefile.am index acc8ea491043..f513b8c4d6f6 100644 --- a/src/conf/ucm/rt5651/Makefile.am +++ b/src/conf/ucm-includes/rt5651/Makefile.am @@ -1,5 +1,5 @@ alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/rt5651 +ucmdir = $(alsaconfigdir)/ucm-includes/rt5651 ucm_DATA = EnableSeq.conf HeadPhones.conf HeadPhones-swapped.conf \ IN1-InternalMic.conf IN2-InternalMic.conf IN12-InternalMic.conf \ IN3-HeadsetMic.conf Speaker.conf MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/MonoSpeaker.conf b/src/conf/ucm-includes/rt5651/MonoSpeaker.conf similarity index 100% rename from src/conf/ucm/rt5651/MonoSpeaker.conf rename to src/conf/ucm-includes/rt5651/MonoSpeaker.conf diff --git a/src/conf/ucm/rt5651/Speaker.conf b/src/conf/ucm-includes/rt5651/Speaker.conf similarity index 100% rename from src/conf/ucm/rt5651/Speaker.conf rename to src/conf/ucm-includes/rt5651/Speaker.conf diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 7cdc6176584a..2c98eac0b745 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -2,7 +2,6 @@ SUBDIRS=\ ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN \ broadwell-rt286 \ broxton-rt298 \ -bytcr \ bytcr-rt5640 \ bytcr-rt5640-mono-spk-dmic1-mic \ bytcr-rt5640-mono-spk-in1-mic \ @@ -29,13 +28,10 @@ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ HDAudio-Lenovo-DualCodecs \ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ -nau8824 \ PandaBoard \ PandaBoardES \ PAZ00 \ PIPO-W2S-Defaultstring-CherryTrailCR \ -rt5640 \ -rt5651 \ SDP4430 \ skylake-rt286 \ TECLAST-X80Pro-Defaultstring-CherryTrailCR \ diff --git a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..f9c64ea16bd5 100644 --- a/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/PIPO-W2S-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,4 @@ -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf index 938b359290d4..15b521be02e5 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf index f36b245e5092..f9aa2cb2d2b1 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf index 1a08ededc40c..0bec7f056de1 100644 --- a/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-mono-spk-in3-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf index 4587b2e1294e..7c8388a0e153 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-dmic1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf index decd396047aa..f074576b053b 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf index d243bf823511..b3c2d03a301c 100644 --- a/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640-stereo-spk-in3-mic/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5640/HiFi.conf b/src/conf/ucm/bytcr-rt5640/HiFi.conf index c833fe74845b..97823e3684d2 100644 --- a/src/conf/ucm/bytcr-rt5640/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5640/HiFi.conf @@ -1,6 +1,6 @@ # Use case Configuration for bytcr-rt5640 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf index dd5b35fc02cb..391e977ddbd0 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf index 777ed7836ee5..f0b80194e39c 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic-hp-swapped/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf index 9f56c077b65a..049b2ebec9ad 100644 --- a/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-mono-spk-in2-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf index e2a579ab2261..63570fb45fbb 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf index ae52e7196110..85516cd23856 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf index a73ff5f86b8f..1767906ffd99 100644 --- a/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf index 8733403cf28d..f254eac979cb 100644 --- a/src/conf/ucm/bytcr-rt5651/HiFi.conf +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -1,6 +1,6 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { EnableSequence [ diff --git a/src/conf/ucm/chtnau8824/HiFi.conf b/src/conf/ucm/chtnau8824/HiFi.conf index dc52800e5eaa..4bff322b33ee 100644 --- a/src/conf/ucm/chtnau8824/HiFi.conf +++ b/src/conf/ucm/chtnau8824/HiFi.conf @@ -1,4 +1,4 @@ -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { diff --git a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf index 799163dbe481..f9c64ea16bd5 100644 --- a/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf +++ b/src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/HiFi.conf @@ -1,4 +1,4 @@ -<searchdir:ucm> +<searchdir:ucm-includes> SectionVerb { -- 2.19.1

3 3

[alsa-devel] [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by Hans de Goede 28 Nov '18

28 Nov '18

Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is identical to all the cset statements this commit removes. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/chtrt5645/HiFi.conf | 116 +------------------------------ 1 file changed, 1 insertion(+), 115 deletions(-) diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index e81866cf..d993f6ae 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,121 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - # media mixer settings - # compress - cset "name='media0_in Gain 0 Switch' on" - cset "name='media0_in Gain 0 Volume' 0" - - # normal - cset "name='media1_in Gain 0 Switch' on" - cset "name='media1_in Gain 0 Volume' 0" - # swm loopback - cset "name='media2_in Gain 0 Switch' off" - cset "name='media2_in Gain 0 Volume' 0%" - # deep buffer - cset "name='media3_in Gain 0 Switch' on" - cset "name='media3_in Gain 0 Volume' 0" - - cset "name='media0_out mix 0 media0_in Switch' on" - cset "name='media0_out mix 0 media1_in Switch' on" - cset "name='media0_out mix 0 media2_in Switch' off" - cset "name='media0_out mix 0 media3_in Switch' on" - - cset "name='media1_out mix 0 media0_in Switch' off" - cset "name='media1_out mix 0 media1_in Switch' off" - cset "name='media1_out mix 0 media2_in Switch' off" - cset "name='media1_out mix 0 media3_in Switch' off" - - cset "name='pcm0_in Gain 0 Switch' on" - cset "name='pcm0_in Gain 0 Volume' 0" - - cset "name='pcm1_in Gain 0 Switch' off" - cset "name='pcm1_in Gain 0 Volume' 0%" - - # codec0_out settings (used if ssp2 is connected to aif1) - cset "name='codec_out0 mix 0 codec_in0 Switch' off" - cset "name='codec_out0 mix 0 codec_in1 Switch' off" - cset "name='codec_out0 mix 0 media_loop1_in Switch' off" - cset "name='codec_out0 mix 0 media_loop2_in Switch' off" - cset "name='codec_out0 mix 0 pcm0_in Switch' on" - cset "name='codec_out0 mix 0 pcm1_in Switch' off" - cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out0 Gain 0 Switch' on" - cset "name='codec_out0 Gain 0 Volume' 0" - - # modem_out settings (used if ssp0 is connected to aif2) - cset "name='modem_out mix 0 codec_in0 Switch' off" - cset "name='modem_out mix 0 codec_in1 Switch' off" - cset "name='modem_out mix 0 media_loop1_in Switch' off" - cset "name='modem_out mix 0 media_loop2_in Switch' off" - cset "name='modem_out mix 0 pcm0_in Switch' on" - cset "name='modem_out mix 0 pcm1_in Switch' off" - cset "name='modem_out mix 0 sprot_loop_in Switch' off" - cset "name='modem_out Gain 0 Switch' on" - cset "name='modem_out Gain 0 Volume' 0" - - # input settings - # pcm1_out settings - - # input used when SSP2 is connected - cset "name='codec_in0 Gain 0 Switch' on" - cset "name='codec_in0 Gain 0 Volume' 0" - - # input used when SSP0 is connected - cset "name='modem_in Gain 0 Switch' on" - cset "name='modem_in Gain 0 Volume' 0" - - cset "name='pcm1_out mix 0 codec_in0 Switch' on" - cset "name='pcm1_out mix 0 modem_in Switch' on" - cset "name='pcm1_out mix 0 codec_in1 Switch' off" - cset "name='pcm1_out mix 0 media_loop1_in Switch' off" - cset "name='pcm1_out mix 0 media_loop2_in Switch' off" - cset "name='pcm1_out mix 0 pcm0_in Switch' off" - cset "name='pcm1_out mix 0 pcm1_in Switch' off" - cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" - - cset "name='pcm1_out Gain 0 Switch' on" - cset "name='pcm1_out Gain 0 Volume' 0" - - # disable codec_out1 - cset "name='codec_out1 mix 0 codec_in0 Switch' off" - cset "name='codec_out1 mix 0 codec_in1 Switch' off" - cset "name='codec_out1 mix 0 media_loop1_in Switch' off" - cset "name='codec_out1 mix 0 media_loop2_in Switch' off" - cset "name='codec_out1 mix 0 pcm0_in Switch' off" - cset "name='codec_out1 mix 0 pcm1_in Switch' off" - cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out1 Gain 0 Switch' off" - cset "name='codec_out1 Gain 0 Volume' 0%" - - # disable codec_in1 - cset "name='codec_in1 Gain 0 Switch' off" - cset "name='codec_in1 Gain 0 Volume' 0%" - - # disable all loops - cset "name='media_loop1_out mix 0 codec_in0 Switch' off" - cset "name='media_loop1_out mix 0 codec_in1 Switch' off" - cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop1_out mix 0 pcm0_in Switch' off" - cset "name='media_loop1_out mix 0 pcm1_in Switch' off" - cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" - - cset "name='media_loop2_out mix 0 codec_in0 Switch' off" - cset "name='media_loop2_out mix 0 codec_in1 Switch' off" - cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop2_out mix 0 pcm0_in Switch' off" - cset "name='media_loop2_out mix 0 pcm1_in Switch' off" - cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" - - cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" - cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" - cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" - cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" - cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" + <bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" -- 2.18.0

6 30

[alsa-devel] [PATCH 0/5] max9867 driver update
by Ladislav Michl 28 Nov '18

28 Nov '18

Hi Mark, this is yet another round of driver update. Tested on AT91SAM9G20 based board with codec in master mode. All implemeted controls do work now. Ladislav Michl (5): ASoC: max9867: Fix whitespace ASoC: max9867: Fix power management ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix signal paths ASoC: max9867: Add copyright and module author sound/soc/codecs/max9867.c | 479 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 41 ++-- 2 files changed, 247 insertions(+), 273 deletions(-) -- 2.19.1

2 9

[alsa-devel] [PATCH v7 0/3] Huawei laptops
by Ayman Bagabas 28 Nov '18

28 Nov '18

This patch set is based on new audio LED triggers in branch topic/leds-trigger from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Changes from v6: * Use audio LED triggers patch set * Use KEY_CONFIG (XF86Tools) instead of KEY_PROG1. In Windows, the key is used to launch Huawei PC manager. XF86Tools is used by default on most desktop environments i.e. Gnome. Ayman Bagabas (3): ALSA: hda: fix front speakers on Huawei MBXP. x86: add support for Huawei WMI hotkeys. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 15 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei-wmi.c | 247 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 22 +++ 4 files changed, 285 insertions(+) create mode 100644 drivers/platform/x86/huawei-wmi.c -- 2.19.1

4 9

[alsa-devel] [PATCH] ALSA: usb-audio: Add vendor and product name for Dell WD19 Dock
by Hui Wang 28 Nov '18

28 Nov '18

Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide useful string for the vendor and product names too, and WD19 has a headset Jack and a Speaker out Jack. In order to support UCM for this Dock in the alsa-lib, here we add profile_name too. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/usb/quirks-table.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 849953e5775c..4c10015cc0ca 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* Dell WD19 Dock */ +{ + USB_DEVICE(0x0bda, 0x402e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Dell", + .product_name = "WD19 Dock", + .profile_name = "Dell-WD19-Dock", + .ifnum = QUIRK_NO_INTERFACE + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.17.1

2 2

[alsa-devel] [sound:for-linus 18/18] drivers/net//ethernet/via/via-rhine.c:1118:2: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 27 Nov '18

27 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 22449df3017d056abde1a1366abe6bb6a3e87751 commit: 22449df3017d056abde1a1366abe6bb6a3e87751 [18/18] sh: Always enable CONFIG_GENERIC_PCI_IOMAP config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 22449df3017d056abde1a1366abe6bb6a3e87751 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): drivers/net//ethernet/via/via-rhine.c: In function 'rhine_init_one_pci': >> drivers/net//ethernet/via/via-rhine.c:1118:2: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(pdev, ioaddr); ^~~~~~~~~~~ pcim_iounmap drivers/net//ethernet/via/via-rhine.c: In function 'rhine_remove_one_platform': drivers/net//ethernet/via/via-rhine.c:2467:24: warning: unused variable 'rp' [-Wunused-variable] struct rhine_private *rp = netdev_priv(dev); ^~ cc1: some warnings being treated as errors vim +1118 drivers/net//ethernet/via/via-rhine.c 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1060 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1061 rc = pci_enable_device(pdev); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1062 if (rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1063 goto err_out; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1064 ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1065 if (pdev->revision < VTunknown0) { 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1066 quirks |= rqRhineI; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1067 } else if (pdev->revision >= VT6102) { 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1068 quirks |= rqWOL | rqForceReset; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1069 if (pdev->revision < VT6105) { ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1070 quirks |= rqStatusWBRace; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1071 } else { ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1072 quirks |= rqIntPHY; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1073 if (pdev->revision >= VT6105_B0) ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1074 quirks |= rq6patterns; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1075 if (pdev->revision >= VT6105M) ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1076 quirks |= rqMgmt; ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1077 } ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1078 } ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1079 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1080 /* sanity check */ 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1081 if ((pci_resource_len(pdev, 0) < io_size) || 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1082 (pci_resource_len(pdev, 1) < io_size)) { 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1083 rc = -EIO; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1084 dev_err(hwdev, "Insufficient PCI resources, aborting\n"); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1085 goto err_out_pci_disable; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1086 } 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1087 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1088 pioaddr = pci_resource_start(pdev, 0); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1089 memaddr = pci_resource_start(pdev, 1); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1090 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1091 pci_set_master(pdev); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1092 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1093 rc = pci_request_regions(pdev, DRV_NAME); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1094 if (rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1095 goto err_out_pci_disable; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1096 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1097 ioaddr = pci_iomap(pdev, (quirks & rqNeedEnMMIO ? 1 : 0), io_size); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1098 if (!ioaddr) { 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1099 rc = -EIO; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1100 dev_err(hwdev, 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1101 "ioremap failed for device %s, region 0x%X @ 0x%lX\n", 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1102 dev_name(hwdev), io_size, memaddr); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1103 goto err_out_free_res; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1104 } 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1105 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1106 enable_mmio(pioaddr, quirks); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1107 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1108 rc = verify_mmio(hwdev, pioaddr, ioaddr, quirks); 5b579e212 drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-05-03 1109 if (rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1110 goto err_out_unmap; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1111 ca8b6e04b drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-30 1112 rc = rhine_init_one_common(&pdev->dev, quirks, 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1113 pioaddr, ioaddr, pdev->irq); 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1114 if (!rc) 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1115 return 0; 2d283862d drivers/net/ethernet/via/via-rhine.c Alexey Charkov 2014-04-22 1116 ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1117 err_out_unmap: ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 @1118 pci_iounmap(pdev, ioaddr); ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1119 err_out_free_res: ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1120 pci_release_regions(pdev); ae996154f drivers/net/ethernet/via/via-rhine.c Roger Luethi 2014-03-18 1121 err_out_pci_disable: ae996154f drivers/net/ethernet/via/via-rhine.c Roger Luethi 2014-03-18 1122 pci_disable_device(pdev); ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1123 err_out: ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1124 return rc; ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1125 } ^1da177e4 drivers/net/via-rhine.c Linus Torvalds 2005-04-16 1126 :::::: The code at line 1118 was first introduced by commit :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: max98373: Added max98373_reset for stable amp reset
by Ryan Lee 27 Nov '18

27 Nov '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Created max98373_reset function to minimize code duplication. Changed regmap_write to regmap_update_bits. Other bits except LSB need to be masked. Added reset verification step to make sure software reset is completed well. Software reset is done in 10ms in normal case. Revision ID is available when the amp is in the idle state which means software reset is completed. Software reset will be performed maximum 3 times to avoid amp reset failure. Generally it is done in the first trial. sleep time after software reset is increased + 30ms for every retrial. Maximum possible msleep time is 100 ms (initial 10 ms + 30 ms * 3 times). sound/soc/codecs/max98373.c | 41 +++++++++++++++++++++++++++++++++++------ 1 file changed, 35 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a09d013..55af7f02 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -724,14 +724,45 @@ static struct snd_soc_dai_driver max98373_dai[] = { } }; +static void max98373_reset(struct max98373_priv *max98373, struct device *dev) +{ + int ret, reg, count, delay; + + count = 0; + while (true) { + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + delay = 10000 + (count * 30000); + usleep_range(delay, delay + 1000); + + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + break; + } + + if (++count > 3) { + dev_err(dev, "Reset failed. (ret:%d)\n", ret); + break; + } + usleep_range(10000, 11000); + } +} + static int max98373_probe(struct snd_soc_component *component) { struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); /* Software Reset */ - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, component->dev); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,9 +849,7 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - regmap_write(max98373->regmap, - MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); - usleep_range(10000, 11000); + max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

3 6

[alsa-devel] [PATCH] slimbus: ngd: fix spelling mistake "exeeds" -> "exceeds"
by Colin King 27 Nov '18

27 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message, fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index ccf33217236c..71f094c9ec68 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -787,7 +787,7 @@ static int qcom_slim_ngd_xfer_msg(struct slim_controller *sctrl, if (txn->msg->num_bytes > SLIM_MSGQ_BUF_LEN || txn->rl > SLIM_MSGQ_BUF_LEN) { - dev_err(ctrl->dev, "msg exeeds HW limit\n"); + dev_err(ctrl->dev, "msg exceeds HW limit\n"); return -EINVAL; } -- 2.19.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.20
by Mark Brown 27 Nov '18

27 Nov '18

The following changes since commit 6be0f96d799f487f05eb412d362d5a1747d665c2: ASoC: stm32: sai: fix master clock naming (2018-10-23 10:51:45 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.20-rc4 for you to fetch changes up to ffdcc3638c58d55a6fa68b6e5dfd4fb4109652eb: ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLE (2018-11-23 14:05:30 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.20 Lots of fixes here, the majority of which are driver specific but there's a couple of core things and one notable driver specific one: - A core fix for a DAPM regression introduced during the component refactoring, we'd lost the code that forced a reevaluation of the DAPM graph after probe (which we suppress during init to save lots of recalcuation) and have now restored it. - A core fix for error handling using the newly added for_each_rtd_codec_dai_rollback() macro. - A fix for the names of widgets in the newly introduced pcm3060 driver, merged as a fix so we don't have a release with legacy names. ---------------------------------------------------------------- Andreas Dannenberg (1): ASoC: pcm186x: Fix device reset-registers trigger value Clément Péron (1): MAINTAINERS: add ASoC maintainers for sound dt-bindings Hans de Goede (1): ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0 Katsuhiro Suzuki (1): ASoC: rockchip: add missing slave_config setting for I2S Keyon Jie (1): ASoC: acpi: fix: continue searching when machine is ignored Kirill Marinushkin (1): ASoC: pcm3060: Rename output widgets Kuninori Morimoto (1): ASoC: rsnd: fixup clock start checker Peter Ujfalusi (4): ASoC: omap-abe-twl6040: Fix missing audio card caused by deferred probing ASoC: omap-mcbsp: Fix latency value calculation for pm_qos ASoC: omap-mcpdm: Add pm_qos handling to avoid under/overruns with CPU_IDLE ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLE Pierre-Louis Bossart (3): ASoC: fix oops w/ for_each_rtd_codec_dai_rollback() macro ASoC: Intel: Power down links before turning off display audio power ASoC: Intel: Skylake: fix Kconfigs, make HDaudio codec optional Richard Fitzgerald (1): ASoC: wm_adsp: Fix dma-unsafe read of scratch registers Rohit kumar (2): ASoC: qdsp6: q6afe: Fix wrong MI2S SD line mask ASoC: qcom: Set dai_link id to each dai_link Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: Only add routing once. ASoC: qdsp6: q6afe-dai: Fix the dai widgets Tzung-Bi Shih (1): ASoC: dapm: Recalculate audio map forcely when card instantiated Vasily Khoruzhick (3): ASoC: sunxi: rename SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP ASoC: sun8i-codec: fix crash on module removal ASoC: sun8i-codec: add missing route for ADC kbuild test robot (1): ASoC: stm32: sai: fix noderef.cocci warnings MAINTAINERS | 1 + include/sound/soc.h | 2 +- sound/soc/codecs/hdac_hdmi.c | 11 +- sound/soc/codecs/pcm186x.h | 2 +- sound/soc/codecs/pcm3060.c | 12 +- sound/soc/codecs/wm_adsp.c | 37 ++--- sound/soc/intel/Kconfig | 26 +++- sound/soc/intel/boards/Kconfig | 24 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++- sound/soc/intel/skylake/skl.c | 32 +++-- sound/soc/omap/omap-abe-twl6040.c | 67 ++++----- sound/soc/omap/omap-dmic.c | 9 ++ sound/soc/omap/omap-mcbsp.c | 6 +- sound/soc/omap/omap-mcpdm.c | 43 +++++- sound/soc/qcom/common.c | 9 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 208 +++++++++++++-------------- sound/soc/qcom/qdsp6/q6afe.c | 16 +-- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ----- sound/soc/qcom/qdsp6/q6routing.c | 19 +++ sound/soc/rockchip/rockchip_pcm.c | 1 + sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/soc-acpi.c | 10 +- sound/soc/soc-core.c | 1 + sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/Kconfig | 2 +- sound/soc/sunxi/sun8i-codec.c | 12 +- 26 files changed, 360 insertions(+), 259 deletions(-)

2 1

[alsa-devel] ALC300 enable
by Kailang 27 Nov '18

27 Nov '18

Hi Takashi, Attach patch will enable new codec ALC300. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Add disable auto-mute quirk for HP Spectre x360 laptop
by Girija Kumar Kasinadhuni 27 Nov '18

27 Nov '18

This device makes a loud buzzing sound when a headphone is inserted while playing audio at full volume through the speaker. Signed-off-by: Girija Kumar Kasinadhuni <gkumar(a)neverware.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674..eeff2ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5495,6 +5495,7 @@ enum { ALC255_FIXUP_DELL_HEADSET_MIC, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, + ALC295_FIXUP_HP_DISABLE_AUTOMUTE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6362,6 +6363,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC295_FIXUP_HP_DISABLE_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6486,6 +6491,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre X360 ", ALC295_FIXUP_HP_DISABLE_AUTOMUTE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.7.4

2 5

Re: [alsa-devel] CS8409 Macbook Pro 2016 2017
by Takashi Iwai 27 Nov '18

27 Nov '18

Please don't drop Cc to ML. Also avoid top-posting. On Sat, 04 Aug 2018 19:02:53 +0200, David Ulricht wrote: > > How can I do that ? > I tried verifying if /proc/asound/card0/codec#0 changes when i put > handphone jack in and out with "diff ver1 ver2" and it is exactly the same. Just do for each bit as you tried for GPIO output. In this case, the direction bit should be different (set 0), and you need to read. > > One strange thing I note is hda-jack-retask.fw contains: > [pincfg] > 0x24 0x90100080 > 0x25 0x90100082 > I have 0x90100080 for 0x24 but in /proc/asound/card0/codec#0 : > Node 0x24 [Pin Complex] wcaps 0x400101: Stereo > Pincap 0x00000010: OUT > Pin Default 0x90100110: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Connection: 1 > 0x02 > > Shouldn't "Pin Default 0x90100110" be 0x90100080 after applying the patch ? No, the codec config value itself won't change, the driver uses the given value only internally. Takashi > I can see that Applying patch firmware 'hda-jack-retask.fw' is before > hdaudioC0D0: autoconfig for Generic: line_outs=2 (0x24/0x25/0x0/0x0/0x0) > type:speaker > in dmesg and this worries me, maybe the pins are overriden by generic > cirrus logic ? > > > On Sat, Aug 4, 2018 at 7:26 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 04 Aug 2018 17:44:59 +0200, > > David Ulricht wrote: > > > > > > I used the following script running as root to verify the 8 GPIOs while > > > playing long youtube video, > > > I have running pulseaudio meanwhile and not restarting it, i think its > > okay > > > like that people on the ubuntu forums say that after using > > > hda-verb and setting a gpio active immediately sound appears, so i guess > > > there is no need to restart pulseaudio. > > > I have done the same test with headphones in. > > > No sound in both scenarios, no input no output detected (i'm watching > > > pavucontrol for input detection). > > > > This codec is fairly unique and doesn't provide the standard jack > > detection on each pin. Maybe it's via either GPIO or any other > > method. You can try to read GPIO pins (set as input) to see whether > > any of them corresponds to the jack detection, for example. > > > > > > Takashi > > > > > > > > #!/usr/bin/env python3 > > > import os,time > > > for x in range(0,256): > > > mhex=("0x%0.2X" % x) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA "+mhex) > > > time.sleep(3) > > > > > > Meanwhile script is running I'm watching /proc/asound/card0/codec#0 and > > > IO[0-7] changes its configuration, > > > in comparison Joel Krahemann in his scripts is changing the hex right > > after > > > /dev/snd/hwC0D0 (0x01) e.g. i dont think he is modifiying the GPIOs > > > correctly. > > > > > > I found the following post: > > > https://forum.manjaro.org/t/sound-not-working-on-a-2017- > > imac-27-5k-retina/43638 > > > According to https://bugzilla.kernel.org/show_bug.cgi?id=195671 iMac27 > > has > > > a similar soundcard. > > > How can i setup with command line or any other utility: > > > """select “off” in the top menu that propose to setup the Digital > > > Surround""" > > > this is some setting in xfce4-mixer package which is not available for > > > ubuntu since 2 LTS versions behind. I tried current Manjaro and couldn't > > > find this setting in audio mixer of xfce4 either. > > > > > > On Sat, Aug 4, 2018 at 9:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > On Sat, 04 Aug 2018 04:07:43 +0200, > > > > David Ulricht wrote: > > > > > > > > > > Hello, > > > > > > > > > > Macbook pro models containing CS8409 for sure are: > > > > > MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. > > > > > > > > > > I'm attaching working sound configuration from Windows 10 registry > > (note > > > > > that the cs420x might need to be ignored or might be important i > > don't > > > > > really know). I believe what is really used is in the > > PinConfigOverride > > > > > section. > > > > > > > > Comparing between BIOS default (init_pin_cfg in alsa-info.sh output) > > > > and your override (user_pin_cfg), there aren't so many changes. > > > > All changes are pretty minor, and I guess it won't influence on the > > > > driver behavior. > > > > > > > > > Interesting thing to note is that sound in MacOS is much louder than > > > > > Bootcamp Windows, would be nice to hear how loud is on linux. > > > > > > > > This is possibly with some vendor-specific GPIO or COEF verbs. > > > > Or it's some direct I2C control. The INI file mentions it. > > > > And that's above HD-audio driver's responsibility. > > > > > > > > > I did convert PinConfigOverride HEX strings to [pincfg] format in the > > > > > attached hda-jack-retask.fw thanks to Takashi's guidance. > > > > > > > > > > I'm attaching also alsamixer ASCII. Only PCM, no Master ? > > > > > > > > It's because the codec chip has no output amplifier at all. > > > > So there can be neither output volume nor mute control on this chip. > > > > > > > > > What I have tried is execute: > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ > > > > > putting @@ from 0x00 to 0x50 > > > > > > > > There are eight GPIOs, so you'd need to test each bit, i.e. 0x01, > > > > 0x02, 0x04, 0x08, ... 0x80. > > > > > > > > And how is the current situation? You can't play *and* record > > > > anything from any inputs / outputs? > > > > > > > > > > > > Takashi > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] [alsa-lib][PATCH 1/2] conf/ucm: Add a UCM profile for Dell WD19 Dock USB-audio
by Hui Wang 27 Nov '18

27 Nov '18

USB-audio device on Dell WD19 docking station provides two individual output PCM streams, one for headphone Jack and another for speaker out Jack. A UCM profile gives the proper roles for these. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- .../ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf | 5 ++++ src/conf/ucm/Dell-WD19-Dock/HiFi.conf | 26 +++++++++++++++++++ src/conf/ucm/Dell-WD19-Dock/Makefile.am | 4 +++ 3 files changed, 35 insertions(+) create mode 100644 src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf create mode 100644 src/conf/ucm/Dell-WD19-Dock/HiFi.conf create mode 100644 src/conf/ucm/Dell-WD19-Dock/Makefile.am diff --git a/src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf b/src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf new file mode 100644 index 00000000..465ff550 --- /dev/null +++ b/src/conf/ucm/Dell-WD19-Dock/Dell-WD19-Dock.conf @@ -0,0 +1,5 @@ +Comment "USB-audio on Dell WD19 docking station" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} diff --git a/src/conf/ucm/Dell-WD19-Dock/HiFi.conf b/src/conf/ucm/Dell-WD19-Dock/HiFi.conf new file mode 100644 index 00000000..e1427a79 --- /dev/null +++ b/src/conf/ucm/Dell-WD19-Dock/HiFi.conf @@ -0,0 +1,26 @@ +SectionDevice."Headphone" { + Comment "Headphone" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:Dock,0" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + Value { + PlaybackChannels "2" + PlaybackPCM "hw:Dock,1" + } +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CaptureChannels "2" + CapturePCM "hw:Dock,0" + } +} diff --git a/src/conf/ucm/Dell-WD19-Dock/Makefile.am b/src/conf/ucm/Dell-WD19-Dock/Makefile.am new file mode 100644 index 00000000..6549ae1b --- /dev/null +++ b/src/conf/ucm/Dell-WD19-Dock/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/Dell-WD19-Dock +ucm_DATA = Dell-WD19-Dock.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) -- 2.17.1

1 1

[alsa-devel] [PATCH v7 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 27 Nov '18

27 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables the use of micmute LED for these devices: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1326f32f4574..9766fd249bdf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5776,7 +5776,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6608,6 +6610,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.19.1

1 0

[alsa-devel] [PATCH v7 1/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 27 Nov '18

27 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 993d34c141c2..1326f32f4574 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5490,6 +5490,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5761,6 +5762,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6591,6 +6608,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.19.1

1 0

[alsa-devel] [PATCH v9 0/8] Add support for audiocodec in Allwinner A64
by Chen-Yu Tsai 26 Nov '18

26 Nov '18

Hi everyone, This is v9 of the A64 audio codec series, originally started by Vasily. These are the last changes to make the A64 audio codec work correctly, and be accurately represented in the device tree and in software. Notably, the new driver changes cover regulator supplies for the simple audio amplifier, which is used in the Pinebook for the internal speakers, and the audio codec's internal headphone amp. There's also an extra patch that enables SDM fractional clock support for the audio PLL on the A64, which gives us accurate audio playback. Patch 1 & 2 add the cpvdd regulator supply to the sun50i-codec-analog dt bindings & driver. Patch 3 & 4 add a regulator supply to the simple-audio-amplifier dt bindings & driver. Patch 5 adds SDM fractional clock support for the audio PLL on the A64. Patch 6 ~ 8 add the device tree bits and enables audio on several boards. I have one extra patch for the Bananapi M64 that's not included here, as I'm waiting for an accessory to test the microphone bias. I'll send that one out once it's tested. As always, we'll take the clock and dts patches after Mark has merged the ASoC ones. And a big thanks to Vasily for having worked on this for so long. Regards ChenYu v9: - Dropped simple amplifier headphone amp device tree changes - Dropped "make enable-gpio optional for simple amplifier" - Lowered oversampling rate from 512 to 128 - Add regulator supply for audio codec CPVDD (headphone amp charge pump) - Hook up CPVDD regulator in board device trees - Add patch to enable SDM fractional clock support for audio PLL on A64 v8: - drop merged patches - add regulator property for simple amplifier, make enable-gpio optional - add regulator to speaker amp, introduce headphone amp v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v5: - collect all the tags v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 Chen-Yu Tsai (3): ASoC: dt-bindings: sun50i-codec-analog: Add headphone amp regulator supply ASoC: sunxi: sun50i-codec-analog: Add support for cpvdd regulator supply clk: sunxi-ng: sun50i: a64: Use sigma-delta modulation for audio PLL Vasily Khoruzhick (5): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../bindings/sound/simple-amplifier.txt | 5 ++ .../bindings/sound/sun50i-codec-analog.txt | 2 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 27 ++++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 49 +++++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 26 +++++++++ .../boot/dts/allwinner/sun50i-a64-sopine.dtsi | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ drivers/clk/sunxi-ng/ccu-sun50i-a64.c | 37 ++++++++----- sound/soc/codecs/simple-amplifier.c | 4 ++ sound/soc/sunxi/sun50i-codec-analog.c | 2 + 10 files changed, 197 insertions(+), 13 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH] ASoC: amd: Fix a NULL vs IS_ERR() check in probe
by Dan Carpenter 26 Nov '18

26 Nov '18

The platform_device_register_full() function doesn't return NULL, it returns error pointers. Fixes: 7894a7e7ea3d ("ASoC: amd: create ACP3x PCM platform device") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/raven/pci-acp3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index c28457fd9b81..facec2472b34 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -97,10 +97,10 @@ static int snd_acp3x_probe(struct pci_dev *pci, pdevinfo.size_data = sizeof(irqflags); adata->pdev = platform_device_register_full(&pdevinfo); - if (!adata->pdev) { + if (IS_ERR(adata->pdev)) { dev_err(&pci->dev, "cannot register %s device\n", pdevinfo.name); - ret = -ENODEV; + ret = PTR_ERR(adata->pdev); goto unmap_mmio; } break; -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup for SSIU subnode
by Kuninori Morimoto 26 Nov '18

26 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit da48a6eb82ea2 ("ASoC: rsnd: add SSIU BUSIF support for Document") updated Documentation for SSIU, but 1) we want to keep old/deprecated DMA description, 2) it is missing SSIU subnode properties. This patch tidyup these Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 47dc819..4d19d2b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -327,8 +327,13 @@ SSI subnode properties: - no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case - dma : Should contain Audio DMAC entry - dma-names : SSI case "rx" (=playback), "tx" (=capture) + Deprecated: see SSIU subnode properties SSIU case "rxu" (=playback), "txu" (=capture) +SSIU subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "rx" (=playback), "tx" (=capture) + SRC subnode properties: - dma : Should contain Audio DMAC entry - dma-names : "rx" (=playback), "tx" (=capture) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: soc.h: makes snd_soc_of_parse_audio_prefix() inline
by Kuninori Morimoto 26 Nov '18

26 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 3b7103562c03c ("ASoC: soc-core: add snd_soc_of_parse_node_prefix()") maked snd_soc_of_parse_audio_prefix() as #define. But it'd be better to make this a static inline rather than a #define. It helps with error messages and type safety. This patch makes it inline. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 37155f9..8ec1de8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1481,9 +1481,15 @@ void snd_soc_of_parse_node_prefix(struct device_node *np, struct snd_soc_codec_conf *codec_conf, struct device_node *of_node, const char *propname); -#define snd_soc_of_parse_audio_prefix(card, conf, node, name) \ - snd_soc_of_parse_node_prefix((card)->dev->of_node, \ - (conf), (node), (name)) +static inline +void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, + struct snd_soc_codec_conf *codec_conf, + struct device_node *of_node, + const char *propname) +{ + snd_soc_of_parse_node_prefix(card->dev->of_node, + codec_conf, of_node, propname); +} int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); -- 2.7.4

2 1

[alsa-devel] [RFC 0/9] ASoC/ARM: Merge the davinci and omap audio directories
by Peter Ujfalusi 26 Nov '18

26 Nov '18

Hi, This RFC series is on top of the omap-mcbsp cleanup series: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-November/141604.h… For Texas Instruments audio support we have two directories at the moment: davinci omap In reality TI have the following SoC lines supported by Linux: daVinci OMAP Sitara Keystone We already needed to have cross directory 'links' for DRA7x/AM5 since they use McASP (davinci directory) and sDMA (omap directory). Keystone II line of SoCs are using drivers from davinci directory (McASP and EDMA). Which is kind of fine as they are in the same place. With the new AM654 (http://www.ti.com/lit/pdf/spruid7) we will have completely new DMA and there is no place to put that as it does not belong under davinci or omap, but it makes no sense to create yet another TI related directory just for the small udma-pcm. Another issue with the current support is how the Kconfigs are written. The DAI drivers are depending on the ASoC platform drivers and it is a bit of a mess at the moment. For simple-sound-card one need to know what DMA is used and then select the DAI driver (if the Kconfig is right). I have also cleaned up the Kconfig prompts and comments a bit to be more precise. As can be seen in the diffstat I tried to do the merger with minimal code change, resisting the urge to rename the davinci-i2c driver to davinci-asp. With the merger I have created a new Kconfig structure: - Submenu for TI related settings for clarity - Only the DAI drivers are selectable and the platform drivers are selected based on the available (enabled) DMA engines - The remaining ASoC machine drivers are selecting the DAIs they need - The davinci-evm is now a single entry, combining all the various daVinci EVMS it supports and selects the needed DAIs based on the enabled boards The series also includes patches to updated the arch/arm/ files (defconfigs, source files using audio Kconfig symbols). I believe with v1 all of this can be merged via ASoC tree as the chances of conflicts are low for the touched arch/arm files. But Sekhar and Tony can correct me if I'm wrong. I think I have catched all (I hope) issues regarding to Kconfig thanks to the various autobuild machines out there kindly testing my linux-next-wip branch. I would appreciate any testing comments before I send the v1! Regards, Peter --- Peter Ujfalusi (9): ASoC: ti: Merge davinci and omap directories MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option ARM: OMAP2: Update for new MCBSP Kconfig option ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options ARM: omap2plus_defconfig: Update the audio options ARM: omap1_defconfig: Do not select ASoC by default ARM: davinci_all_defconfig: Update the audio options ASoC: ti: Kconfig: Remove the deprecated options MAINTAINERS | 11 +- arch/arm/configs/davinci_all_defconfig | 5 +- arch/arm/configs/omap1_defconfig | 2 - arch/arm/configs/omap2plus_defconfig | 14 +- arch/arm/mach-davinci/board-dm365-evm.c | 4 +- arch/arm/mach-omap1/Makefile | 2 +- arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/pdata-quirks.c | 4 +- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/davinci/Kconfig | 106 --------- sound/soc/davinci/Makefile | 16 -- sound/soc/omap/Kconfig | 120 ---------- sound/soc/omap/Makefile | 30 --- sound/soc/ti/Kconfig | 209 ++++++++++++++++++ sound/soc/ti/Makefile | 44 ++++ sound/soc/{omap => ti}/ams-delta.c | 0 sound/soc/{davinci => ti}/davinci-evm.c | 4 +- sound/soc/{davinci => ti}/davinci-i2s.c | 0 sound/soc/{davinci => ti}/davinci-i2s.h | 0 sound/soc/{davinci => ti}/davinci-mcasp.c | 14 +- sound/soc/{davinci => ti}/davinci-mcasp.h | 0 sound/soc/{davinci => ti}/davinci-vcif.c | 0 sound/soc/{davinci => ti}/edma-pcm.c | 0 sound/soc/{davinci => ti}/edma-pcm.h | 4 +- sound/soc/{omap => ti}/n810.c | 0 sound/soc/{omap => ti}/omap-abe-twl6040.c | 0 sound/soc/{omap => ti}/omap-dmic.c | 0 sound/soc/{omap => ti}/omap-dmic.h | 0 .../omap-hdmi-audio.c => ti/omap-hdmi.c} | 0 sound/soc/{omap => ti}/omap-mcbsp-priv.h | 0 sound/soc/{omap => ti}/omap-mcbsp-st.c | 0 sound/soc/{omap => ti}/omap-mcbsp.c | 0 sound/soc/{omap => ti}/omap-mcbsp.h | 0 sound/soc/{omap => ti}/omap-mcpdm.c | 0 sound/soc/{omap => ti}/omap-mcpdm.h | 0 sound/soc/{omap => ti}/omap-twl4030.c | 0 sound/soc/{omap => ti}/omap3pandora.c | 0 sound/soc/{omap => ti}/osk5912.c | 0 sound/soc/{omap => ti}/rx51.c | 0 sound/soc/{omap => ti}/sdma-pcm.c | 0 sound/soc/{omap => ti}/sdma-pcm.h | 4 +- 42 files changed, 294 insertions(+), 307 deletions(-) delete mode 100644 sound/soc/davinci/Kconfig delete mode 100644 sound/soc/davinci/Makefile delete mode 100644 sound/soc/omap/Kconfig delete mode 100644 sound/soc/omap/Makefile create mode 100644 sound/soc/ti/Kconfig create mode 100644 sound/soc/ti/Makefile rename sound/soc/{omap => ti}/ams-delta.c (100%) rename sound/soc/{davinci => ti}/davinci-evm.c (99%) rename sound/soc/{davinci => ti}/davinci-i2s.c (100%) rename sound/soc/{davinci => ti}/davinci-i2s.h (100%) rename sound/soc/{davinci => ti}/davinci-mcasp.c (99%) rename sound/soc/{davinci => ti}/davinci-mcasp.h (100%) rename sound/soc/{davinci => ti}/davinci-vcif.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.c (100%) rename sound/soc/{davinci => ti}/edma-pcm.h (91%) rename sound/soc/{omap => ti}/n810.c (100%) rename sound/soc/{omap => ti}/omap-abe-twl6040.c (100%) rename sound/soc/{omap => ti}/omap-dmic.c (100%) rename sound/soc/{omap => ti}/omap-dmic.h (100%) rename sound/soc/{omap/omap-hdmi-audio.c => ti/omap-hdmi.c} (100%) rename sound/soc/{omap => ti}/omap-mcbsp-priv.h (100%) rename sound/soc/{omap => ti}/omap-mcbsp-st.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.c (100%) rename sound/soc/{omap => ti}/omap-mcbsp.h (100%) rename sound/soc/{omap => ti}/omap-mcpdm.c (100%) rename sound/soc/{omap => ti}/omap-mcpdm.h (100%) rename sound/soc/{omap => ti}/omap-twl4030.c (100%) rename sound/soc/{omap => ti}/omap3pandora.c (100%) rename sound/soc/{omap => ti}/osk5912.c (100%) rename sound/soc/{omap => ti}/rx51.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.c (100%) rename sound/soc/{omap => ti}/sdma-pcm.h (85%) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

5 14

[alsa-devel] Fwd: [BUG] The infamous MOTU Ultralite * bitcrunch issue
by Stanisław Drozd 26 Nov '18

26 Nov '18

[Originally posted in alsa-user, but that list doesn't appear to be very widely used] Hello everyone, I recently updated the firmware on my UltraLite mk4 (IF YOURS IS WORKING OK, DON'T UPGRADE or you'll repeat my mistake). Current firmware on my piece is 1.3.4+232, 07/27/18 17:46:22, BIOS is Aug-04-2017 b568a4c. After the update an issue appeared - a moment after initializing the device and playing normally, it starts sounding as if it was put through a bitcruncher effect. After a longer while the output loses gain only to disappear completely after some more seconds pass (the gain change is not smooth, it seems to go away in steps). What I have tried: * Switching internal clock to LTC on the interface - would hold the sync together for a little longer; I did NOT send the actual timecode because I had trouble finding something to generate it for me * Switching sample rates - No noticeable change in behavior * Testing with speaker-test - No change in behavior, looks like this has to happen on ALSA level or lower * Changing clock source in Cadence (a JACK helper) settings and reinitializing - No effect Possible clues I saw: * dmesg started showing this when I plug the device in: [ 533.419099] usb 1-2.3: new high-speed USB device number 8 using xhci_hcd [ 533.527638] usb 1-2.3: New USB device found, idVendor=07fd, idProduct=0005, bcdDevice= 2.00 [ 533.527646] usb 1-2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 533.527650] usb 1-2.3: Product: UltraLite-mk4 [ 533.527654] usb 1-2.3: Manufacturer: MOTU [ 533.527658] usb 1-2.3: SerialNumber: 0001f2fffe008e19 [ 534.377873] usb 1-2.3: clock source 1 is not valid, cannot use <and about 50 more lines of the clock source message> Questions: * What other places can I go to to report this? * What other settings can I tweak? Thanks, Stanislaw PS: I have a basic understanding of driver work, I think I could help with creating a patch with some guidance

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops
by Hui Wang 26 Nov '18

26 Nov '18

We have several Lenovo laptops with the codec alc285, when playing sound via headphone, we can hear click/pop noise in the headphone, if we let the headphone share the DAC of NID 0x2 with the speaker, the noise disappears. The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon. I have tried to set preferred_dacs and override_conn, but neither of them worked. Thanks for Kailang, he told me to invalidate the NID 0x3 through override_wcaps. BugLink: https://bugs.launchpad.net/bugs/1805079 Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1118fd1bbf1a..e66da22272fd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5358,6 +5358,16 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, spec->gen.preferred_dacs = preferred_pairs; } +/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */ +static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_wcaps(codec, 0x03, 0); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5495,6 +5505,7 @@ enum { ALC255_FIXUP_DELL_HEADSET_MIC, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, + ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6362,6 +6373,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_invalidate_dacs, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7035,6 +7050,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, -- 2.17.1

2 1

Re: [alsa-devel] [LKP] [ALSA] 3a6d1980fe: kernel_BUG_at_mm/page_alloc.c
by Takashi Iwai 26 Nov '18

26 Nov '18

On Mon, 26 Nov 2018 01:28:05 +0100, kernel test robot wrote: > > FYI, we noticed the following commit (built with gcc-7): > > commit: 3a6d1980fe96dbbfe3ae58db0048867f5319cdbf ("ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers") > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git for-next .... > [ 174.830284] kernel BUG at mm/page_alloc.c:2850! OK, this seem to be an issue about using __GFP_COMP and alloc_pages_exact(). I didn't hit this while testing locally maybe just because I used the buffer size in power-of-two casually. I dropped the patch until we sort this out. thanks, Takashi

1 0

[alsa-devel] [PATCH 00/10] ASoC: merge audio-graph-scu into audio-graph - step2
by Kuninori Morimoto 26 Nov '18

26 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. These are step2 patches Kuninori Morimoto (10): ASoC: soc-core: add snd_soc_of_parse_node_prefix() ASoC: simple-scu-card: tidyup "prefix" parsing on Doc ASoC: simple-scu-card: tidyup "prefix" parsing ASoC: audio-graph-scu-card: tidyup "prefix" parsing on Doc ASoC: audio-graph-scu-card: tidyup "prefix" parsing ASoC: audio-graph-scu-card: tidyup "convert-rate/channels" parsing on Doc ASoC: audio-graph-scu-card: tidyup "convert-rate/channels" parsing ASoC: audio-graph-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: audio-graph-card: tidyup typo SND_AUDIO_GRAPH_CARD ASoC: audio-graph-card: tidyup define position .../bindings/sound/audio-graph-scu-card.txt | 4 +- .../devicetree/bindings/sound/simple-scu-card.txt | 2 +- include/sound/soc.h | 6 +- sound/soc/generic/Kconfig | 2 +- sound/soc/generic/audio-graph-card.c | 10 +-- sound/soc/generic/audio-graph-scu-card.c | 71 ++++++++++------------ sound/soc/generic/simple-scu-card.c | 7 +++ sound/soc/soc-core.c | 11 ++-- 8 files changed, 59 insertions(+), 54 deletions(-) -- 2.7.4

2 22

[alsa-devel] [PATCH] ALSA: pcm: Fix starvation on down_write_nonblock()
by Chanho Min 25 Nov '18

25 Nov '18

Commit 67ec1072b053 (ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream) fixes deadlock for non-atomic PCM stream. But, This patch causes antother stuck. If writer is RT thread and reader is a normal thread, the reader thread will be difficult to get scheduled. It may not give chance to release read locks and writer gets stuck for a long time or forever if they are pinned to single cpu. To fix this, The writer gives reader a chance to be scheduled by using the minimum msleep() instaed of spinning. This is for concept, We may need to change the function name and comments or suggest another approach. Signed-off-by: Chanho Min <chanho.min(a)lge.com> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 66c90f4..88d4aab 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -96,7 +96,7 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); static inline void down_write_nonblock(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) - cond_resched(); + msleep(1); } #define PCM_LOCK_DEFAULT 0 -- 2.1.4

2 2

[alsa-devel] Fwd: [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by youling 257 25 Nov '18

25 Nov '18

---------- Forwarded message --------- From: youling 257 <youling257(a)gmail.com> Date: 2018年11月24日周六下午6:42 Subject: Re: [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>, Takashi Iwai <tiwai(a)suse.de>, Hans de Goede <hdegoede(a)redhat.com>, <alsa-devel(a)vger.kernel.org> alsaucm -c chtrt5645 set _verb HiFi set _enadev Speaker ALSA lib conf.c:823:(get_char_skip_comments) Cannot access file bytcr/PlatformEnableSeq.conf ALSA lib conf.c:1877:(snd_config_load1) _toplevel_:17:18:Unexpected char ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/ASUSTeKCOMPUTERINC.-X205TA-1.0-X205TA/ASUSTeKCOMPUTERINC.-X205TA-1.0-X205TA.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card ASUSTeKCOMPUTERINC.-X205TA-1.0-X205TA ALSA lib conf.c:823:(get_char_skip_comments) Cannot access file bytcr/PlatformEnableSeq.conf ALSA lib conf.c:1877:(snd_config_load1) _toplevel_:17:18:Unexpected char ALSA lib utils.c:75:(uc_mgr_config_load) could not load configuration file /usr/share/alsa/ucm/chtrt5645/HiFi.conf ALSA lib parser.c:1101:(parse_verb_file) error: failed to open verb file /usr/share/alsa/ucm/chtrt5645/HiFi.conf : -2 ALSA lib main.c:946:(snd_use_case_mgr_open) error: failed to import chtrt5645 use case configuration -22 alsaucm: error failed to open sound card chtrt5645: Invalid argument

1 0

[alsa-devel] [PATCH] ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers
by Takashi Iwai 24 Nov '18

24 Nov '18

snd_malloc_pages() and snd_free_pages() are merely thin wrappers of the standard page allocator / free functions. Even the arguments are compatible with some standard helpers, so there is little merit of keeping these wrappers. This patch replaces the all existing callers of snd_malloc_pages() and snd_free_pages() with the direct calls of the standard helper functions. In this version, we use a recently introduced one, alloc_pages_exact(), which suits better than the old snd_malloc_pages() implementation. Then we can avoid the waste of pages by alignment to power-of-two. Also, the __GFP_COMP flag that has been always added in the old implementation is dropped for most of callers now. The only one that may still need it is the PCM buffer allocation which can be quite large. The rest are single page allocations, so no sense to keep that flag. With these conversions, snd_malloc_pages() and snd_free_pages() are dropped altogether from the whole tree. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 4 --- sound/core/memalloc.c | 54 +++------------------------------ sound/core/pcm.c | 14 ++++----- sound/usb/usx2y/usX2Yhwdep.c | 3 +- sound/usb/usx2y/usbusx2y.c | 3 +- sound/usb/usx2y/usx2yhwdeppcm.c | 6 ++-- 6 files changed, 19 insertions(+), 65 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index af3fa577fa06..c56cb2871368 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -152,9 +152,5 @@ int snd_dma_alloc_pages_fallback(int type, struct device *dev, size_t size, struct snd_dma_buffer *dmab); void snd_dma_free_pages(struct snd_dma_buffer *dmab); -/* basic memory allocation functions */ -void *snd_malloc_pages(size_t size, gfp_t gfp_flags); -void snd_free_pages(void *ptr, size_t size); - #endif /* __SOUND_MEMALLOC_H */ diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 59a4adc286ed..d12bf811e751 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,53 +30,6 @@ #endif #include <sound/memalloc.h> -/* - * - * Generic memory allocators - * - */ - -/** - * snd_malloc_pages - allocate pages with the given size - * @size: the size to allocate in bytes - * @gfp_flags: the allocation conditions, GFP_XXX - * - * Allocates the physically contiguous pages with the given size. - * - * Return: The pointer of the buffer, or %NULL if no enough memory. - */ -void *snd_malloc_pages(size_t size, gfp_t gfp_flags) -{ - int pg; - - if (WARN_ON(!size)) - return NULL; - if (WARN_ON(!gfp_flags)) - return NULL; - gfp_flags |= __GFP_COMP; /* compound page lets parts be mapped */ - pg = get_order(size); - return (void *) __get_free_pages(gfp_flags, pg); -} -EXPORT_SYMBOL(snd_malloc_pages); - -/** - * snd_free_pages - release the pages - * @ptr: the buffer pointer to release - * @size: the allocated buffer size - * - * Releases the buffer allocated via snd_malloc_pages(). - */ -void snd_free_pages(void *ptr, size_t size) -{ - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - free_pages((unsigned long) ptr, pg); -} -EXPORT_SYMBOL(snd_free_pages); - /* * * Bus-specific memory allocators @@ -188,8 +141,9 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->bytes = 0; switch (type) { case SNDRV_DMA_TYPE_CONTINUOUS: - dmab->area = snd_malloc_pages(size, - (__force gfp_t)(unsigned long)device); + dmab->area = alloc_pages_exact(size, + (__force gfp_t)(unsigned long)device | + __GFP_COMP); dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA @@ -273,7 +227,7 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) { switch (dmab->dev.type) { case SNDRV_DMA_TYPE_CONTINUOUS: - snd_free_pages(dmab->area, dmab->bytes); + free_pages_exact(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA #ifdef CONFIG_GENERIC_ALLOCATOR diff --git a/sound/core/pcm.c b/sound/core/pcm.c index fdb9b92fc8d6..1c0ecb22b1f4 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1004,22 +1004,22 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return -ENOMEM; size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)); - runtime->status = snd_malloc_pages(size, GFP_KERNEL); + runtime->status = alloc_pages_exact(size, GFP_KERNEL); if (runtime->status == NULL) { kfree(runtime); return -ENOMEM; } - memset((void*)runtime->status, 0, size); + memset(runtime->status, 0, size); size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)); - runtime->control = snd_malloc_pages(size, GFP_KERNEL); + runtime->control = alloc_pages_exact(size, GFP_KERNEL); if (runtime->control == NULL) { - snd_free_pages((void*)runtime->status, + free_pages_exact(runtime->status, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))); kfree(runtime); return -ENOMEM; } - memset((void*)runtime->control, 0, size); + memset(runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); init_waitqueue_head(&runtime->tsleep); @@ -1045,9 +1045,9 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) runtime = substream->runtime; if (runtime->private_free != NULL) runtime->private_free(runtime); - snd_free_pages((void*)runtime->status, + free_pages_exact(runtime->status, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))); - snd_free_pages((void*)runtime->control, + free_pages_exact(runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); /* Avoid concurrent access to runtime via PCM timer interface */ diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index c1dd9a7b48df..bfe1108416cf 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -75,7 +75,8 @@ static int snd_us428ctls_mmap(struct snd_hwdep * hw, struct file *filp, struct v if (!us428->us428ctls_sharedmem) { init_waitqueue_head(&us428->us428ctls_wait_queue_head); - if(!(us428->us428ctls_sharedmem = snd_malloc_pages(sizeof(struct us428ctls_sharedmem), GFP_KERNEL))) + us428->us428ctls_sharedmem = alloc_pages_exact(sizeof(struct us428ctls_sharedmem), GFP_KERNEL); + if (!us428->us428ctls_sharedmem) return -ENOMEM; memset(us428->us428ctls_sharedmem, -1, sizeof(struct us428ctls_sharedmem)); us428->us428ctls_sharedmem->CtlSnapShotLast = -2; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index da4a5a541512..9f7bbed2c0f0 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -437,7 +437,8 @@ static void snd_usX2Y_card_private_free(struct snd_card *card) kfree(usX2Y(card)->In04Buf); usb_free_urb(usX2Y(card)->In04urb); if (usX2Y(card)->us428ctls_sharedmem) - snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem)); + free_pages_exact(usX2Y(card)->us428ctls_sharedmem, + sizeof(*usX2Y(card)->us428ctls_sharedmem)); if (usX2Y(card)->card_index >= 0 && usX2Y(card)->card_index < SNDRV_CARDS) snd_usX2Y_card_used[usX2Y(card)->card_index] = 0; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 4fd9276b8e50..4453c2b911d0 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -488,7 +488,9 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream); if (NULL == usX2Y->hwdep_pcm_shm) { - if (NULL == (usX2Y->hwdep_pcm_shm = snd_malloc_pages(sizeof(struct snd_usX2Y_hwdep_pcm_shm), GFP_KERNEL))) + usX2Y->hwdep_pcm_shm = alloc_pages_exact(sizeof(struct snd_usX2Y_hwdep_pcm_shm), + GFP_KERNEL); + if (!usX2Y->hwdep_pcm_shm) return -ENOMEM; memset(usX2Y->hwdep_pcm_shm, 0, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } @@ -700,7 +702,7 @@ static void snd_usX2Y_hwdep_pcm_private_free(struct snd_hwdep *hwdep) { struct usX2Ydev *usX2Y = hwdep->private_data; if (NULL != usX2Y->hwdep_pcm_shm) - snd_free_pages(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); + free_pages_exact(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } -- 2.19.1

3 3

[alsa-devel] [PATCH 1/2] ALSA: wss: Fix invalid snd_free_pages() at error path
by Takashi Iwai 24 Nov '18

24 Nov '18

Some spurious calls of snd_free_pages() have been overlooked and remain in the error paths of wss driver code. Since runtime->dma_area is managed by the PCM core helper, we shouldn't release manually. Drop the superfluous calls. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/wss/wss_lib.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 32453f81b95a..3a5008837576 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1531,7 +1531,6 @@ static int snd_wss_playback_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma1); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->playback_substream = substream; @@ -1572,7 +1571,6 @@ static int snd_wss_capture_open(struct snd_pcm_substream *substream) if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma2); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } chip->capture_substream = substream; -- 2.19.1

2 3

[alsa-devel] [PATCH v3 0/3] Huawei laptops
by Ayman Bagabas 24 Nov '18

24 Nov '18

Changes from v2: * Support for Huawei MBX * Style and formating issues [PATCH v3 1/3] The first patch adds support for missing hotkeys on some models. [PATCH v3 2/3] This one enables the front speakers on the Huawei Matebook X Pro (MBXP). This solves bug 200501 https://bugzilla.kernel.org/show_bug.cgi?id=200501 It simply uses the pins configurations generated by hdajackretast using the settings posted on the bug page https://imgur.com/a/N1xsCVZ [PATCH v3 3/3] This enables the micmute LED on Huawei laptops using ACPI calls. Ayman Bagabas (3): x86: add support for Huawei WMI hotkeys. ALSA: hda: fix front speakers on Huawei MBXP. ALSA: hda: add support for Huawei WMI micmute LED drivers/platform/x86/Kconfig | 11 + drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei_wmi.c | 236 +++++++++++++++++++ include/linux/platform_data/x86/huawei_wmi.h | 9 + sound/pci/hda/huawei_wmi_helper.c | 47 ++++ sound/pci/hda/patch_realtek.c | 29 +++ 6 files changed, 333 insertions(+) create mode 100644 drivers/platform/x86/huawei_wmi.c create mode 100644 include/linux/platform_data/x86/huawei_wmi.h create mode 100644 sound/pci/hda/huawei_wmi_helper.c -- 2.17.2

5 30

[alsa-devel] Undefined behaviour in ac97_codec.c - shift exponent 68 is too large for 32-bit type 'int'
by Meelis Roos 23 Nov '18

23 Nov '18

I updated one of my old laptops (ECS Desknote 532 with Transmeta CPU) to newest kernel (4.20.0-rc3-00145-gedeca3a769ad) and turned on UBSAN checks. Got the following UBSAN warning multiple times per boot. The soundcard: 00:04.0 Multimedia audio controller [0401]: ULi Electronics Inc. M5455 PCI AC-Link Controller Audio Device [10b9:5455] (rev 10) Subsystem: Elitegroup Computer Systems M5455 PCI AC-Link Controller Audio Device [1019:0f56] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (16000ns min), Cache Line Size: 128 bytes Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at e400 [size=256] Region 1: Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: <access denied> Kernel driver in use: snd_intel8x0 Kernel modules: snd_intel8x0 /proc/asound/cards: 0 [M5455 ]: ICH - ALi M5455 ALi M5455 with ALC655 at irq 5 gcc version 8.2.0 (Debian 8.2.0-9) [ 15.688683] snd_intel8x0 0000:00:04.0: intel8x0_measure_ac97_clock: measured 58318 usecs (2808 samples) [ 15.689033] snd_intel8x0 0000:00:04.0: clocking to 48000 ... [ 19.667746] ================================================================================ [ 19.668078] UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7 [ 19.668268] shift exponent 68 is too large for 32-bit type 'int' [ 19.668465] CPU: 0 PID: 199 Comm: alsactl Not tainted 4.20.0-rc3-00145-gedeca3a769ad #2 [ 19.668602] Hardware name: Elitegroup Co. 532/532, BIOS 080010 02/22/2005 [ 19.668602] Call Trace: [ 19.668602] dump_stack+0x16/0x19 [ 19.668602] ubsan_epilogue+0xb/0x29 [ 19.668602] __ubsan_handle_shift_out_of_bounds.cold.15+0x26/0x78 [ 19.668602] snd_ac97_put_spsa.cold.50+0xf/0x24 [snd_ac97_codec] [ 19.668602] ? _copy_from_user+0x33/0xd0 [ 19.668602] snd_ctl_ioctl+0x69e/0x820 [ 19.668602] ? __seccomp_filter+0x60/0x320 [ 19.668602] ? snd_ctl_elem_add_user+0x8a0/0x8a0 [ 19.668602] do_vfs_ioctl+0x90/0x6c0 [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x32/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ? __secure_computing+0x2b/0x80 [ 19.668602] ? syscall_trace_enter+0x141/0x1b0 [ 19.668602] ? __switch_to_asm+0x26/0x4c [ 19.668602] ksys_ioctl+0x39/0x70 [ 19.668602] sys_ioctl+0x11/0x13 [ 19.668602] do_fast_syscall_32+0x90/0x1c0 [ 19.668602] entry_SYSENTER_32+0x6b/0xbd [ 19.668602] EIP: 0xb7fd69ad [ 19.668602] Code: 54 cd ff ff 85 d2 8b 98 58 cd ff ff 89 c8 74 02 89 0a 5b 5d c3 8b 04 24 c3 8b 14 24 c3 8b 3c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 8d 76 00 58 b8 77 00 00 00 cd 80 90 8d 76 [ 19.668602] EAX: ffffffda EBX: 00000003 ECX: c2c45513 EDX: bffff670 [ 19.668602] ESI: 00000000 EDI: 00000001 EBP: bffff9c8 ESP: bffff508 [ 19.668602] DS: 007b ES: 007b FS: 0000 GS: 0033 SS: 007b EFLAGS: 00000292 [ 19.668602] ================================================================================ -- Meelis Roos <mroos(a)linux.ee>

2 3

[alsa-devel] [PATCH] ALSA: hda/realtek - fix headset mic detection for MSI MS-B171
by Anisse Astier 23 Nov '18

23 Nov '18

MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the MS-B120, in order for the headset mic to be properly detected. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: stable(a)vger.kernel.org Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..1e789e43a5f4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6531,6 +6531,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: hda: Add ASRock N68C-S UCC the power_save blacklist
by Hans de Goede 23 Nov '18

23 Nov '18

Power-saving is causing plops on audio start/stop on the built-in audio of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to the power_save blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d8eb2b5f51ae..0bbdf1a01e76 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2169,6 +2169,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0397, "Asrock N68C-S UCC", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), -- 2.19.1

2 1

[alsa-devel] [PATCH 0/5] ALSA: firewire-tascam: support notification of
by Takashi Sakamoto 23 Nov '18

23 Nov '18

Hi, Units of TASCAM FireWire series transfer its state in tx isochronous packets. The way to handle change of the state is a longstanding issue since I firstly reveal the mechanism in this list[1]. In this time, Scott Bahling dedicates for further investigation to seek better implementation[2][3][4]. Finally, we identified meaning of the most of bits in the image of state, and decided which changes are worth for userspace applications to handle as events. This commit adds support notification of the events. Userspace applications can handle the notification via ALSA hwdep interface. Furthermore, they can retrieve image of the latest states via the interface. Changes from the initial proposal[5]: - obsolete usage of mmap(2) due to several disadvantages. Instead, use event queue for change notification, and add ioctl command to retrieve image of states. Changes from the latest work[6]: - To reduce time in an interrupt context in which tx isochronous packets are handled, data in the event and the image is passed to userspace in big-endian order. Additionally, image of the states are passed to userspace in big-endian order as well. Userspace applications are responsible to convert the data in host-endianness order. - improve efficiency to copy batch of events to userspace in hwdep interface - rename structure and macro name. This patchset is also available in my remote branch for backport purpose to Linux v4.17 or later[7]. I've already prepare userspace implementation; libhinwa. A 'topic/firewire-tascam-pr' remote branch includes patchset for the additional feature[8]. This library produces GObject class to handle the event notification and emits GObject signal, and has a method 'HinawaSndTscm.get_state()', with enough consideration to endianness. Developers can process change of state with these two APIs. > Scott As the above, I changed method name from 'HinawaSndTscm.get_status()'. I'd like you to change your local implementation for 'hinawa-utils'. Sorry to add further work to you. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094817.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140051.… [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140593.ht… [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-November/141395.h… [5] https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspace [6] https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspac… [7] https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspac… [8] https://github.com/takaswie/libhinawa/tree/topic/tascam-userspace-pr Regards Takashi Sakamoto (5): ALSA: firewire-tascam: pick up data of state from tx isochronous pakcets ALSA: firewire-tascam: add new hwdep ioctl command to get state image ALSA: firewire-tascam: queue events for change of control surface ALSA: firewire-tascam: apply small refactoring to handle several type of event for hwdep interface ALSA: firewire-tascam: notify events of change of state for userspace applications include/uapi/sound/firewire.h | 20 +++++ sound/firewire/tascam/amdtp-tascam.c | 51 +++++++++++- sound/firewire/tascam/tascam-hwdep.c | 115 +++++++++++++++++++++++---- sound/firewire/tascam/tascam.h | 9 +++ 4 files changed, 180 insertions(+), 15 deletions(-) -- 2.19.1

2 7

[alsa-devel] Applied "ASoC: dt-bindings: add regulator property to simple amplifier" to the asoc tree
by Mark Brown 23 Nov '18

23 Nov '18

The patch ASoC: dt-bindings: add regulator property to simple amplifier has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8e84de031a11c4240f3654fd4c380542e358f5f6 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Thu, 22 Nov 2018 18:23:20 +0800 Subject: [PATCH] ASoC: dt-bindings: add regulator property to simple amplifier Amplifier may have associated regulator, so add a property for it. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/simple-amplifier.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt index 8647edae7af0..7182ac4f1e65 100644 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -4,9 +4,14 @@ Required properties: - compatible : "dioo,dio2125" or "simple-audio-amplifier" - enable-gpios : the gpio connected to the enable pin of the simple amplifier +Optional properties: +- VCC-supply : power supply for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + Example: amp: analog-amplifier { compatible = "simple-audio-amplifier"; + VCC-supply = <&regulator>; enable-gpios = <&gpio GPIOH_3 0>; }; -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: simple-amplifier: add VCC regulator widget" to the asoc tree
by Mark Brown 23 Nov '18

23 Nov '18

The patch ASoC: simple-amplifier: add VCC regulator widget has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6debd01a74b8c15ab3f2a6f7ff4a7dcbada101cb Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Thu, 22 Nov 2018 18:23:21 +0800 Subject: [PATCH] ASoC: simple-amplifier: add VCC regulator widget Amplifier may have assosicated regulator, so add a widget for it and appropriate route. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/simple-amplifier.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index 85524acf3e9c..c07e8a80b4b7 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -19,6 +19,7 @@ #include <linux/gpio/consumer.h> #include <linux/module.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #define DRV_NAME "simple-amplifier" @@ -58,11 +59,14 @@ static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { (SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)), SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), + SND_SOC_DAPM_REGULATOR_SUPPLY("VCC", 20, 0), }; static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, + { "OUTL", NULL, "VCC" }, + { "OUTR", NULL, "VCC" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; -- 2.19.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: omap: Fix and add pm_qos configuration
by Peter Ujfalusi 23 Nov '18

23 Nov '18

Hi, The defconfig for OMAP2+ now have the CPU_IDLE enabled which can cause audio artifacts because we try to enter too low power state from where the wakeup takes longer than the FIFO can tolerate on the dai side. While adding pm_qos to McPDM and DMIC I have noticed that the McBSP calculation was not correct as we need usec for the latency value. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcbsp: Fix latency value calculation for pm_qos ASoC: omap-mcpdm: Add pm_qos handling to avoid under/overruns with CPU_IDLE ASoC: omap-dmic: Add pm_qos handling to avoid overruns with CPU_IDLE sound/soc/omap/omap-dmic.c | 9 ++++++++ sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-mcpdm.c | 43 ++++++++++++++++++++++++++++++++++++- 3 files changed, 54 insertions(+), 4 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

4 14

[alsa-devel] [PATCH] ASoC: sunxi: rename SND_SUN8I_ADDA_PR_REGMAP
by Arnd Bergmann 22 Nov '18

22 Nov '18

The newly added SND_SUN50I_CODEC_ANALOG driver selects a nonexisting symbol SND_SUNXI_ADDA_PR_REGMAP, which was evidently intended to replace SND_SUN8I_ADDA_PR_REGMAP, but since they are now mismatched, we can run into link errors for some configurations: sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe': sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init' The new name appears to be more sensible, and as the symbol is hidden, there are no downsides in the rename, so use that consistently now. Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sunxi/Kconfig | 4 ++-- sound/soc/sunxi/Makefile | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 66aad0d3f9c7..0f992bdf1a10 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -23,7 +23,7 @@ config SND_SUN8I_CODEC config SND_SUN8I_CODEC_ANALOG tristate "Allwinner sun8i Codec Analog Controls Support" depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select SND_SUN8I_ADDA_PR_REGMAP + select SND_SUNXI_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. @@ -54,7 +54,7 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. -config SND_SUN8I_ADDA_PR_REGMAP +config SND_SUNXI_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..c4dd2803a011 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -5,4 +5,4 @@ obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o -obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUNXI_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o -- 2.18.0

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[alsa-devel] [PATCH 0/2] ALSA: control: Race fix and cleanup for user ctls
by Takashi Iwai 22 Nov '18

22 Nov '18

Hi, there seems a possible race between addition and removal of a user ctl element as spotted by syzkaller. This patchset addresses it and also provides a cleanup thereafter. The first one should go to 4.20 while the second one should go for 4.21. Takashi === Takashi Iwai (2): ALSA: control: Fix race between adding and removing a user element ALSA: control: Consolidate helpers for adding and replacing ctl elements sound/core/control.c | 171 ++++++++++++++++++++----------------------- 1 file changed, 81 insertions(+), 90 deletions(-) -- 2.19.1

1 2

[alsa-devel] [PATCH 0/2] ASoC: SDM845: Update MI2S and TDM configuration
by Rohit kumar 21 Nov '18

21 Nov '18

Update bit clock rate, slot width for TDM and MI2S interfaces. Also add support for secondary MI2S TX interface in SDM845 machine driver. Rohit kumar (2): ASoC: sdm845: Update slot_width for Quaternary TDM port ASoC: sdm845: Add support for Secondary MI2S interface sound/soc/qcom/sdm845.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 6

[alsa-devel] [PATCH] ASoC: rsnd: Add device tree support for r8a77995
by Kuninori Morimoto 21 Nov '18

21 Nov '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Simply document new compat strings. There appears to be no need for a driver updates. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 453e8ec..47dc819 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -277,6 +277,7 @@ Required properties: - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - "renesas,rcar_sound-r8a77990" (R-Car E3) + - "renesas,rcar_sound-r8a77995" (R-Car D3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup rsnd_parse_connect_ssiu_compatible()
by Kuninori Morimoto 21 Nov '18

21 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_parse_connect_ssiu_compatible() is doing - using rsnd_ssiu_id(), but we use it via rsnd_mod_id() - we can break loop if rsnd_dai_connect() was called This patch fixup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssiu.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 650b14e..04b1a48 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -382,9 +382,11 @@ static void rsnd_parse_connect_ssiu_compatible(struct rsnd_priv *priv, for_each_rsnd_ssiu(ssiu, priv, i) { mod = rsnd_mod_get(ssiu); - if ((rsnd_mod_id(ssi_mod) == rsnd_ssiu_id(mod)) && - (rsnd_mod_id_sub(mod) == 0)) + if ((rsnd_mod_id(ssi_mod) == rsnd_mod_id(mod)) && + (rsnd_mod_id_sub(mod) == 0)) { rsnd_dai_connect(mod, io, mod->type); + return; + } } } -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: makes rsnd_ssi_is_dma_mode() static
by Kuninori Morimoto 21 Nov '18

21 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ssi.c only is using rsnd_ssi_is_dma_mode(). Let's move it as static function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 5 +++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 64c3a3b..3c57129 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -726,7 +726,6 @@ extern const char * const volume_ramp_rate[]; int rsnd_ssi_probe(struct rsnd_priv *priv); void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); -int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 361dc2c..6439a74 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -117,6 +117,8 @@ struct rsnd_ssi { (rsnd_ssi_run_mods(io) & (1 << rsnd_mod_id(mod))) #define rsnd_ssi_can_output_clk(mod) (!__rsnd_ssi_is_pin_sharing(mod)) +static int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); + int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) { struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io); @@ -1003,12 +1005,11 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .get_status = rsnd_ssi_get_status, }; -int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) +static int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) { return mod->ops == &rsnd_ssi_dma_ops; } - /* * ssi mod function */ -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: merge audio-graph-scu into audio-graph - step1
by Kuninori Morimoto 21 Nov '18

21 Nov '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. ex) audio-graph can handle normal sound card CPU1 -- Codec1 CPU2 -- Codec2 audio-graph-scu can handle DPCM sound card CPU1 --- Codec1 CPU2 -/ But, we can't use these in same time CPU1 --- Codec1 CPU2 --- Codec2 CPU3 -/ So, this is the time to merge audio-graph and audio-graph-scu. The patch-set are a little bit huge (and few patches are depend on posted OF patch. I'm now waiting...), thus, I separated these into few steps. This is step1 as prepare merging. I think it will be 4 or more steps. "simple card" and "audio graph card" are similar sound card, and now I'm focusing to "audio graph card" merging only. Basically simple-card and simple-scu-card are similar too, and we can merge it. But I have no plan for it now. Please try it someone ;P Kuninori Morimoto (4): ASoC: simple-card-utils: remove asoc_simple_card_clk_register() ASoC: simple-card-utils: accept NULL parameter on asoc_simple_card_xxx() ASoC: simple-card-utils: tidyup asoc_simple_card_parse_convert() ASoC: simple_card_utils: remove "option" from asoc_simple_card_of_parse_routing() include/sound/simple_card_utils.h | 6 +++--- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-scu-card.c | 4 ++-- sound/soc/generic/simple-card-utils.c | 34 +++++++++++++++----------------- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-scu-card.c | 4 ++-- 6 files changed, 25 insertions(+), 27 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH v4 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 21 Nov '18

21 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. This series add a mode-switch delay which will prevent the clock from being turned off without complying with manufacturer timing specifications. Also introduces a module parameter for the wakeup_delay parameter. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message v3: Updated commit message and introduced module param for wakeup_delay v4: Addressed parameter name to align with existing parameter. Added dt entry in Documentation Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay Documentation/devicetree/bindings/sound/dmic.txt | 2 ++ sound/soc/codecs/dmic.c | 36 ++++++++++++++++++++++++ 2 files changed, 38 insertions(+) -- 1.9.1

3 5

[alsa-devel] [PATCH alsa-lib] pcm_dshare: Fix segfault when not binding channel 0
by Kirill Marinushkin 21 Nov '18

21 Nov '18

Configuration to reproduce: ~~~~ pcm.share_right { type dshare ipc_key 73 ipc_perm 0666 slave { pcm "hw:0,0" } bindings { # the seagfault happens when we don't bind channel 0 1 1 } } ~~~~ Execute to reproduce: ~~~~ $ aplay -D plug:share_right test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Segmentation fault ~~~~ For channels whithout binding, values are set to UINT_MAX in function `snd_pcm_direct_parse_bindings()`: ~~~~ for (chn = 0; chn < count; chn++) bindings[chn] = UINT_MAX; /* don't route */ ~~~~ But, these values are not checked when playing, which causes the segfault. This commit fixes the issue. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- src/pcm/pcm_dshare.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 54dc9d48..2bb735fe 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -67,7 +67,9 @@ static void do_silence(snd_pcm_t *pcm) format = dshare->shmptr->s.format; for (chn = 0; chn < channels; chn++) { dchn = dshare->bindings ? dshare->bindings[chn] : chn; - snd_pcm_area_silence(&dst_areas[dchn], 0, dshare->shmptr->s.buffer_size, format); + if (dchn != UINT_MAX) + snd_pcm_area_silence(&dst_areas[dchn], 0, + dshare->shmptr->s.buffer_size, format); } } @@ -91,7 +93,9 @@ static void share_areas(snd_pcm_direct_t *dshare, } else { for (chn = 0; chn < channels; chn++) { dchn = dshare->bindings ? dshare->bindings[chn] : chn; - snd_pcm_area_copy(&dst_areas[dchn], dst_ofs, &src_areas[chn], src_ofs, size, format); + if (dchn != UINT_MAX) + snd_pcm_area_copy(&dst_areas[dchn], dst_ofs, + &src_areas[chn], src_ofs, size, format); } } @@ -834,8 +838,11 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->spcm = spcm; } - for (chn = 0; chn < dshare->channels; chn++) - dshare->u.dshare.chn_mask |= (1ULL<<dshare->bindings[chn]); + for (chn = 0; chn < dshare->channels; chn++) { + unsigned int dchn = dshare->bindings ? dshare->bindings[chn] : chn; + if (dchn != UINT_MAX) + dshare->u.dshare.chn_mask |= (1ULL << dchn); + } if (dshare->shmptr->u.dshare.chn_mask & dshare->u.dshare.chn_mask) { SNDERR("destination channel specified in bindings is already used"); dshare->u.dshare.chn_mask = 0; -- 2.13.6

2 1

[alsa-devel] [RFC PATCH] ASoC: Intel: Skylake: fix Kconfigs, make HDaudio codec optional
by Pierre-Louis Bossart 20 Nov '18

20 Nov '18

The Skylake driver currently has a set of problems supporting load/unload modules. We need to make the HDaudio codec support optional to help narrow down the issues. Support for HDaudio codecs also leads to a Kconfig issue. We want the hdac_hda codec to be compilable independently of Skylake (e.g. with ALL_CODECS) but when Skylake is selected as built-in the hdac_hda codec needs to use the same option due a a code dependency Solve both problems by adding a user-selectable boolean Kconfig, select HDAC_HDA as needed and make the HDaudio codec support in the Skylake driver optional. Tests on a Chell Chromebook device without HDaudio show no regression for speaker and HDMI playback. This is submitted as an RFC to allow for comments and more validation. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 26 +++++++++++++++++++++++--- sound/soc/intel/boards/Kconfig | 24 ++++++++++++++---------- sound/soc/intel/skylake/skl.c | 20 ++++++++++++++++++-- 3 files changed, 55 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0caa1f4eb94d..18e717703685 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -101,22 +101,42 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI codec, then enable this option by saying Y or m. This is a recommended option -config SND_SOC_INTEL_SKYLAKE_SSP_CLK - tristate - config SND_SOC_INTEL_SKYLAKE tristate "SKL/BXT/KBL/GLK/CNL... Platforms" depends on PCI && ACPI + select SND_SOC_INTEL_SKYLAKE_COMMON + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with the DSP enabled in the BIOS + then enable this option by saying Y or m. + +if SND_SOC_INTEL_SKYLAKE + +config SND_SOC_INTEL_SKYLAKE_SSP_CLK + tristate + +config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + bool "HDAudio codec support" + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with an HDaudio codec + then enable this option by saying Y + +config SND_SOC_INTEL_SKYLAKE_COMMON + tristate select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS then enable this option by saying Y or m. +endif ## SND_SOC_INTEL_SKYLAKE + config SND_SOC_ACPI_INTEL_MATCH tristate select SND_SOC_ACPI if ACPI diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 73ca1350aa31..b177db2a0dbb 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,16 +293,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH Say Y if you have such a device. If unsure select "N". -config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH - tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_SOC_HDAC_HDMI - select SND_SOC_HDAC_HDA - help - This adds support for ASoC machine driver for Intel platforms - SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on MFD_INTEL_LPSS && I2C && ACPI @@ -319,4 +309,18 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH endif ## SND_SOC_INTEL_SKYLAKE +if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + +config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH + tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_SOC_HDAC_HDMI + # SND_SOC_HDAC_HDA is already selected + help + This adds support for ASoC machine driver for Intel platforms + SKL/KBL/BXT/APL with iDisp, HDA audio codecs. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ac0b4ff21acc..901ef5902762 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -37,7 +37,9 @@ #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) #include "../../../soc/codecs/hdac_hda.h" +#endif /* * initialize the PCI registers @@ -654,6 +656,8 @@ static void skl_clock_device_unregister(struct skl *skl) platform_device_unregister(skl->clk_dev); } +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + #define IDISP_INTEL_VENDOR_ID 0x80860000 /* @@ -672,6 +676,8 @@ static void load_codec_module(struct hda_codec *codec) #endif } +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ + /* * Probe the given codec address */ @@ -681,9 +687,11 @@ static int probe_codec(struct hdac_bus *bus, int addr) (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; struct skl *skl = bus_to_skl(bus); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) struct hdac_hda_priv *hda_codec; - struct hdac_device *hdev; int err; +#endif + struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -693,6 +701,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec), GFP_KERNEL); if (!hda_codec) @@ -711,6 +720,13 @@ static int probe_codec(struct hdac_bus *bus, int addr) load_codec_module(&hda_codec->codec); } return 0; +#else + hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + return snd_hdac_ext_bus_device_init(bus, addr, hdev); +#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } /* Codec initialization */ @@ -866,7 +882,7 @@ static int skl_create(struct pci_dev *pci, hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); -#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA) +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); -- 2.17.1

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[alsa-devel] [PATCH 0/3] Intel SST fixes for Chromebooks/HDaudio platorms
by Pierre-Louis Bossart 20 Nov '18

20 Nov '18

While testing Mark's for-next branch on Chromebooks, 3 problems were exposed a. the search loop for ACPI IDs is broken on SKL/KBL which are the most advanced users of quirks. This is embarrassing since it's been broken for a very very long time and those platforms have been stuck in time with 3.18 and 4.4 kernels. b. the change I personally made to use a common interface for machine driver didn't work either for all the Chromebooks using information on dmic number. My bad. c. the change to use the common interface also missed the fact that the Skylake driver passed the information using drv_data instead of platform_data as done by the Atom and SOF drivers. The third patch aligns the Skylake driver with the two others. We should now have no dependencies between machine and platform drivers, which will help reuse the same code. These patches were tested on HDaudio platforms (ApolloLake and Whisky Lake) as well as Chell (SKL) and Soraka (KBL) Chromebooks - for the latter case we are still investigating DSP IPC errors that are unrelated to these changes Keyon Jie (1): ASoC: acpi: fix: continue searching when machine is ignored Pierre-Louis Bossart (2): ASoC: Intel: fix interface for Chromebook machine drivers ASoC: Intel: use platform_data for machine drivers include/sound/soc-acpi.h | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 10 +++++----- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 10 +++++----- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 10 +++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 10 +++++----- sound/soc/intel/skylake/skl.c | 15 +++++++++++---- sound/soc/intel/skylake/skl.h | 3 --- sound/soc/soc-acpi.c | 10 ++++++++-- 9 files changed, 41 insertions(+), 30 deletions(-) -- 2.17.1

2 6

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for CBM_CFS and CBS_CFM clocking modes
by Peter Ujfalusi 20 Nov '18

20 Nov '18

The codec can support any variation of bclk/fs master/slave configuration. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a271e6e6b8f..6aa0edf8c5ef 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1260,6 +1260,16 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, aic3x->master = 0; iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER); break; + case SND_SOC_DAIFMT_CBM_CFS: + aic3x->master = 1; + iface_areg |= BIT_CLK_MASTER; + iface_areg &= ~WORD_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aic3x->master = 1; + iface_areg |= WORD_CLK_MASTER; + iface_areg &= ~BIT_CLK_MASTER; + break; default: return -EINVAL; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: Intel: hdac_hdmi: add Icelake support
by Bard liao 20 Nov '18

20 Nov '18

From: Bard liao <bard.liao(a)intel.com> Add Icelake device id. Also, Icelake's pin2port mapping table is complicated. So we use a mapping table to do the pin2port mapping. Signed-off-by: Bard liao <bard.liao(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 63 ++++++++++++++++++++++++++++++------ 1 file changed, 54 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..fac397326515 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -121,8 +121,16 @@ struct hdac_hdmi_dai_port_map { struct hdac_hdmi_cvt *cvt; }; +/* + * pin to port mapping table where the value indicate the pin number and + * the index indicate the port number with 1 base. + */ +static const int icl_pin2port_map[] = {0x4, 0x6, 0x8, 0xa, 0xb}; + struct hdac_hdmi_drv_data { unsigned int vendor_nid; + const int *port_map; /* pin to port mapping table */ + int port_num; }; struct hdac_hdmi_priv { @@ -1329,11 +1337,12 @@ static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) return 0; } -#define INTEL_VENDOR_NID 0x08 -#define INTEL_GLK_VENDOR_NID 0x0b +#define INTEL_VENDOR_NID_0x2 0x02 +#define INTEL_VENDOR_NID_0x8 0x08 +#define INTEL_VENDOR_NID_0xb 0x0b #define INTEL_GET_VENDOR_VERB 0xf81 #define INTEL_SET_VENDOR_VERB 0x781 -#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ +#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void hdac_hdmi_skl_enable_all_pins(struct hdac_device *hdev) @@ -1538,7 +1547,26 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, static int hdac_hdmi_pin2port(void *aptr, int pin) { - return pin - 4; /* map NID 0x05 -> port #1 */ + struct hdac_device *hdev = aptr; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + const int *map = hdmi->drv_data->port_map; + int i; + + if (!hdmi->drv_data->port_num) + return pin - 4; /* map NID 0x05 -> port #1 */ + + /* + * looking for the pin number in the mapping table and return + * the index which indicate the port number + */ + for (i = 0; i < hdmi->drv_data->port_num; i++) { + if (pin == map[i]) + return i + 1; + } + + /* return -1 if pin number exceeds our expectation */ + dev_err(&hdev->dev, "Can't find the port for pin %d\n", pin); + return -1; } static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) @@ -1549,9 +1577,18 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) struct hdac_hdmi_port *hport = NULL; struct snd_soc_component *component = hdmi->component; int i; - - /* Don't know how this mapping is derived */ - hda_nid_t pin_nid = port + 0x04; + hda_nid_t pin_nid; + + if (!hdmi->drv_data->port_num) { + /* for legacy platforms */ + pin_nid = port + 0x04; + } else if (port < hdmi->drv_data->port_num) { + /* get pin number from the pin2port mapping table */ + pin_nid = hdmi->drv_data->port_map[port - 1]; + } else { + dev_err(&hdev->dev, "Can't find the pin for port %d\n", port); + return; + } dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__, pin_nid, pipe); @@ -1973,12 +2010,18 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx) return port->eld.info.spk_alloc; } +static struct hdac_hdmi_drv_data intel_icl_drv_data = { + .vendor_nid = INTEL_VENDOR_NID_0x2, + .port_map = icl_pin2port_map, + .port_num = ARRAY_SIZE(icl_pin2port_map), +}; + static struct hdac_hdmi_drv_data intel_glk_drv_data = { - .vendor_nid = INTEL_GLK_VENDOR_NID, + .vendor_nid = INTEL_VENDOR_NID_0xb, }; static struct hdac_hdmi_drv_data intel_drv_data = { - .vendor_nid = INTEL_VENDOR_NID, + .vendor_nid = INTEL_VENDOR_NID_0x8, }; static int hdac_hdmi_dev_probe(struct hdac_device *hdev) @@ -2259,6 +2302,8 @@ static const struct hda_device_id hdmi_list[] = { &intel_glk_drv_data), HDA_CODEC_EXT_ENTRY(0x8086280d, 0x100000, "Geminilake HDMI", &intel_glk_drv_data), + HDA_CODEC_EXT_ENTRY(0x8086280f, 0x100000, "Icelake HDMI", + &intel_icl_drv_data), {} }; -- 2.17.1

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[alsa-devel] [PATCH 1/2] ASoC: max9892x: Add documentation for reset-gpio support
by Cheng-Yi Chiang 20 Nov '18

20 Nov '18

max98927 codec driver will support reset-gpio binding so it can toggle reset line in its probe function. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Documentation/devicetree/bindings/sound/max9892x.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt index f6171591ddc61..8bf30483eedcd 100644 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -29,6 +29,8 @@ Required properties: Range : 0 (off), 1 (on), Default : 0 - reg : the I2C address of the device for I2C + - reset-gpio: reset gpio the chip is connected to. This is supported + on max98927. Example: @@ -38,4 +40,5 @@ codec: max98927@3a { imon-slot-no = <1>; interleave-mode = <0>; reg = <0x3a>; + reset-gpio = <&tlmm 75 GPIO_ACTIVE_LOW>; }; -- 2.16.4

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