April 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [asoc:for-5.2 90/97] sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'?
by kbuild test robot 01 Apr '19

01 Apr '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 35146467bd43704c52d1d5760af9839a10ad5b12 commit: 11c0269017b212fd47c593307d2dc3eb9713b2d0 [90/97] ASoC: Mediatek: MT8183: Add machine driver with TS3A227 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 11c0269017b212fd47c593307d2dc3eb9713b2d0 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function 'mt8183_mt6358_ts3a227_max98357_dev_probe': >> sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'? dai_link->platform = NULL; ^~~~~~~~ platforms vim +325 sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c 298 299 static int 300 mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) 301 { 302 struct snd_soc_card *card = &mt8183_mt6358_ts3a227_max98357_card; 303 struct device_node *platform_node; 304 struct snd_soc_dai_link *dai_link; 305 struct pinctrl *default_pins; 306 int ret, i; 307 308 card->dev = &pdev->dev; 309 310 platform_node = of_parse_phandle(pdev->dev.of_node, 311 "mediatek,platform", 0); 312 if (!platform_node) { 313 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); 314 return -EINVAL; 315 } 316 317 for_each_card_prelinks(card, i, dai_link) { 318 /* In the alsa soc-core, the "platform" will be 319 * allocated by devm_kzalloc if null. 320 * There is a special case that registerring 321 * sound card is failed at the first time, but 322 * the "platform" will not null when probe is trying 323 * again. It's not expected normally. 324 */ > 325 dai_link->platform = NULL; 326 327 if (dai_link->platform_name) 328 continue; 329 dai_link->platform_of_node = platform_node; 330 } 331 332 mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node = 333 of_parse_phandle(pdev->dev.of_node, 334 "mediatek,headset-codec", 0); 335 if (!mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) { 336 dev_err(&pdev->dev, 337 "Property 'mediatek,headset-codec' missing/invalid\n"); 338 return -EINVAL; 339 } 340 341 ret = devm_snd_soc_register_card(&pdev->dev, card); 342 if (ret) 343 dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", 344 __func__, ret); 345 346 default_pins = 347 devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); 348 if (IS_ERR(default_pins)) { 349 dev_err(&pdev->dev, "%s set pins failed\n", 350 __func__); 351 return PTR_ERR(default_pins); 352 } 353 354 return ret; 355 } 356 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: uapi: #include <time.h> in asound.h
by Daniel Mentz 01 Apr '19

01 Apr '19

The uapi header asound.h defines types based on struct timespec. We need to #include <time.h> to get access to the definition of this struct. Previously, we encountered the following error message when building applications with a clang/bionic toolchain: kernel-headers/sound/asound.h:350:19: error: field has incomplete type 'struct timespec' struct timespec trigger_tstamp; ^ The absence of the time.h #include statement does not cause build errors with glibc, because its version of stdlib.h indirectly includes time.h. Signed-off-by: Daniel Mentz <danielmentz(a)google.com> --- include/uapi/sound/asound.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 404d4b9ffe76..df1153cea0b7 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -32,6 +32,7 @@ #ifndef __KERNEL__ #include <stdlib.h> +#include <time.h> #endif /* -- 2.21.0.392.gf8f6787159e-goog

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[alsa-devel] Applied "ASoC: tlv320aic32x4: Fix Common Pins" to the asoc tree
by Mark Brown 01 Apr '19

01 Apr '19

The patch ASoC: tlv320aic32x4: Fix Common Pins 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 c63adb28f6d913310430f14c69f0a2ea55eed0cc Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Sat, 30 Mar 2019 09:02:02 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Fix Common Pins The common pins were mistakenly not added to the DAPM graph. Adding these pins will allow valid graphs to be created. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 96f1526cb258..5520044929f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -490,6 +490,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN2_R"), SND_SOC_DAPM_INPUT("IN3_L"), SND_SOC_DAPM_INPUT("IN3_R"), + SND_SOC_DAPM_INPUT("CM_L"), + SND_SOC_DAPM_INPUT("CM_R"), }; static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { -- 2.20.1

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[alsa-devel] [PATCH] ASoC: cs42l51: add multi endpoint support
by Olivier Moysan 01 Apr '19

01 Apr '19

Support multiple endpoints on cs42L51 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: STM32MP157C-DK2 board uses CS42L51 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. >From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: cs42l51_port: port { cs42l51_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; cs42l51_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/codecs/cs42l51.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index fd2bd74024c1..80da3cd73e04 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -464,6 +464,13 @@ static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) return snd_soc_component_write(component, CS42L51_DAC_OUT_CTL, reg); } +static int cs42l51_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_dai_ops cs42l51_dai_ops = { .hw_params = cs42l51_hw_params, .set_sysclk = cs42l51_set_dai_sysclk, @@ -526,6 +533,7 @@ static const struct snd_soc_component_driver soc_component_device_cs42l51 = { .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), .dapm_routes = cs42l51_routes, .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), + .of_xlate_dai_id = cs42l51_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt5677-spi: Add ACPI ID
by Oder Chiou 01 Apr '19

01 Apr '19

Add the ACPI ID for the product "chromebook pixel 2015" to match the coreboot settings. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677-spi.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84501c2020c7..167a02773a0b 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -25,6 +25,7 @@ #include <linux/sysfs.h> #include <linux/clk.h> #include <linux/firmware.h> +#include <linux/acpi.h> #include "rt5677-spi.h" @@ -226,9 +227,16 @@ static int rt5677_spi_probe(struct spi_device *spi) return 0; } +static const struct acpi_device_id rt5677_spi_acpi_id[] = { + { "RT5677AA", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id); + static struct spi_driver rt5677_spi_driver = { .driver = { .name = "rt5677", + .acpi_match_table = ACPI_PTR(rt5677_spi_acpi_id), }, .probe = rt5677_spi_probe, }; -- 2.21.0

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[alsa-devel] [PATCH 0/4] ASoC: Mediatek: MT8183: Add support for machine driver with DA7219 or TS3A227
by Shunli Wang 01 Apr '19

01 Apr '19

ASoC: Mediatek: MT8183: Add support for machine driver with DA7219 or TS3A227 Shunli Wang (4): ASoC: Mediatek: MT8183: Add machine driver with TS3A227 ASoC: Mediatek: Add document for mt8183-mt6358-ts3a227-max98357 ASoC: Mediatek: MT8183: Add machine driver with DA7219 ASoC: Mediatek: Add document for mt8183-da7219-max98357 .../bindings/sound/mt8183-da7219-max98357.txt | 15 + .../sound/mt8183-mt6358-ts3a227-max98357.txt | 15 + sound/soc/mediatek/Kconfig | 26 ++ sound/soc/mediatek/mt8183/Makefile | 2 + sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 472 +++++++++++++++++++++ .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 382 +++++++++++++++++ 6 files changed, 912 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt create mode 100644 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c -- 1.9.1

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[alsa-devel] [PATCH][next] ASoC: fsl: fix spelling mistake: "missign" -> "missing"
by Colin King 01 Apr '19

01 Apr '19

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> --- sound/soc/fsl/imx-audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 72e37ca6cfbb..7983bd339c01 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -161,7 +161,7 @@ static int imx_audmix_probe(struct platform_device *pdev) } if (!audmix_np) { - dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + dev_err(&pdev->dev, "Missing DT node for parent device.\n"); return -EINVAL; } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: Fix typo in comment
by Charles Keepax 01 Apr '19

01 Apr '19

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5e615fbddf9a7..0aa62b26f61cc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3263,7 +3263,7 @@ static void wm_halo_stop_core(struct wm_adsp *dsp) regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, HALO_CORE_EN, 0); - /* reset halo core with CORE_SOFT_REEST */ + /* reset halo core with CORE_SOFT_RESET */ regmap_update_bits(dsp->regmap, dsp->base + HALO_CORE_SOFT_RESET, HALO_CORE_SOFT_RESET_MASK, 1); } -- 2.11.0

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[alsa-devel] Applied "ASoC: simple-card-utils: remove set but not used variable 'dai_name'" to the asoc tree
by Mark Brown 01 Apr '19

01 Apr '19

The patch ASoC: simple-card-utils: remove set but not used variable 'dai_name' 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 35146467bd43704c52d1d5760af9839a10ad5b12 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Mon, 25 Mar 2019 03:17:40 +0000 Subject: [PATCH] ASoC: simple-card-utils: remove set but not used variable 'dai_name' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_parse_clk': sound/soc/generic/simple-card-utils.c:164:18: warning: parameter 'dai_name' set but not used [-Wunused-but-set-parameter] It's not used since commit 0580dde59438 ("ASoC: simple-card-utils: add asoc_simple_debug_info()"), so can be removed. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Mukesh Ojha <mojha(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4ed68348f939..db1458a19985 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -173,10 +173,8 @@ int asoc_simple_parse_clk(struct device *dev, * see * soc-core.c :: snd_soc_init_multicodec() */ - if (dlc) { + if (dlc) dai_of_node = dlc->of_node; - dai_name = dlc->dai_name; - } /* * Parse dai->sysclk come from "clocks = <&xxx>" -- 2.20.1

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[alsa-devel] [PATCH v3 1/2] ASoC: tlv320aic31xx: Add headphone/headset detection
by Andrew F. Davis 01 Apr '19

01 Apr '19

This device can detect the insertion/removal of headphones and headsets. Enable reporting this status by enabling this interrupt and forwarding this to upper-layers if a jack has been defined. This jack definition and the resulting operation from a jack detection event must currently be defined by sound card platform code until CODEC outputs to jack mappings can be defined generically. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Rebase Changes from v2: - Rebase sound/soc/codecs/tlv320aic31xx.c | 48 +++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic31xx.h | 11 ++++++++ 2 files changed, 58 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 6744f736501f..3e11b3d41f3d 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -25,6 +25,7 @@ #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -89,6 +90,7 @@ static bool aic31xx_volatile(struct device *dev, unsigned int reg) case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ case AIC31XX_INTRDACFLAG2: case AIC31XX_INTRADCFLAG2: + case AIC31XX_HSDETECT: return true; } return false; @@ -157,6 +159,7 @@ struct aic31xx_priv { int micbias_vg; struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct snd_soc_jack *jack; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1233,6 +1236,20 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } +int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + + aic31xx->jack = jack; + + /* Enable/Disable jack detection */ + regmap_write(aic31xx->regmap, AIC31XX_HSDETECT, + jack ? AIC31XX_HSD_ENABLE : 0); + + return 0; +} + static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); @@ -1258,6 +1275,7 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, + .set_jack = aic31xx_set_jack, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), @@ -1362,8 +1380,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) dev_err(dev, "Short circuit on Left output is detected\n"); if (value & AIC31XX_HPRSCDETECT) dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & AIC31XX_HSPLUG) { + unsigned int val; + int status = 0; + + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); + if (ret) { + dev_err(dev, "Failed to read headset type: %d\n", ret); + goto exit; + } + + switch ((val & AIC31XX_HSD_TYPE_MASK) >> + AIC31XX_HSD_TYPE_SHIFT) { + case AIC31XX_HSD_HP: + status |= SND_JACK_HEADPHONE; + break; + case AIC31XX_HSD_HS: + status |= SND_JACK_HEADSET; + break; + default: + break; + } + + if (aic31xx->jack) + snd_soc_jack_report(aic31xx->jack, status, + AIC31XX_JACK_MASK); + } if (value & ~(AIC31XX_HPLSCDETECT | - AIC31XX_HPRSCDETECT)) + AIC31XX_HPRSCDETECT | + AIC31XX_HSPLUG)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1476,6 +1521,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_HSPLUGDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 2636f2c6bc79..9985ceed97ae 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -20,6 +20,9 @@ #define AIC31XX_MINIDSP_BIT BIT(2) #define DAC31XX_BIT BIT(3) +#define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_HEADSET) + enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, @@ -220,6 +223,14 @@ struct aic31xx_pdata { /* AIC31XX_DACMUTE */ #define AIC31XX_DACMUTE_MASK GENMASK(3, 2) +/* AIC31XX_HSDETECT */ +#define AIC31XX_HSD_ENABLE BIT(7) +#define AIC31XX_HSD_TYPE_MASK GENMASK(6, 5) +#define AIC31XX_HSD_TYPE_SHIFT 5 +#define AIC31XX_HSD_NONE 0x00 +#define AIC31XX_HSD_HP 0x01 +#define AIC31XX_HSD_HS 0x03 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- 2.21.0

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