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

[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] [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] [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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Re: [alsa-devel] INFO: task hung in evdev_release
by syzbot 01 Apr '19

01 Apr '19

syzbot has bisected this bug to: commit e32d99af6830c9a8f37b4f2637ef0cdc60fa79fb Author: Jerome Brunet <jbrunet(a)baylibre.com> Date: Tue Jul 17 15:42:50 2018 +0000 ASoC: meson: add axg fifos DT binding documentation bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=13fba69b200000 start commit: 9dcd936c Merge tag 'for-4.19/dm-fixes-4' of git://git.kern.. git tree: upstream final crash: https://syzkaller.appspot.com/x/report.txt?x=1007a69b200000 console output: https://syzkaller.appspot.com/x/log.txt?x=17fba69b200000 kernel config: https://syzkaller.appspot.com/x/.config?x=88e9a8a39dc0be2d dashboard link: https://syzkaller.appspot.com/bug?extid=a979743610b4755d4d57 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=10fcd826400000 Reported-by: syzbot+a979743610b4755d4d57(a)syzkaller.appspotmail.com Fixes: e32d99af6830 ("ASoC: meson: add axg fifos DT binding documentation") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

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[alsa-devel] [PATCH v2 0/7] ASoC: tlv320aic26: Add device tree support and bug fixes
by Cormier, Jonathan 31 Mar '19

31 Mar '19

Hi, The following patches were created to get the tlv320aic26 working on our AM335x SoM. I'm resubmitting this patch series. Previous version here: https://lore.kernel.org/patchwork/cover/601439/ This patch series relies on the following patch for proper operation. commit 93d0ad8f374c ("ASoC: tlv320aic26: Convert to params_width()") Changes since RFC: - Removed the unnecessary "#if defined(CONFIG_OF)" - Dropped the DSP_B patch Changes since V1: - Reordered Kconfig patch to first so each patch can be built - Rebased changes to Linux 5.1-rc1 Cormier, Jonathan (7): ASoC: tlv320aic26: Allow tlv320aic26 to be selected in menuconfig dt-bindings: ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Fix module autoload ASoC: tlv320aic26: Fix regmap by setting reg_defaults and reg_stride ASoC: tlv320aic26: hw_params was unintentionally clearing AIC26 master mode ASoC: tlv320aic26: Change Capture Mute to Capture Switch to match alsa documentation. .../devicetree/bindings/sound/tlv320aic26.txt | 65 +++++++++++++++++++ sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/tlv320aic26.c | 55 ++++++++++++++-- 3 files changed, 116 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic26.txt -- 2.21.0

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[alsa-devel] [PATCH] ASoC: hdmi: add device PM dependency
by libin.yang＠intel.com 31 Mar '19

31 Mar '19

From: Libin Yang <libin.yang(a)intel.com> HDMI audio codec register operation depends on the display audio power domain. The hdmi audio codec dapm event callback must be called after display audio power is turned on. Add hdac_hdmi_device_link_add() in HDMI audio codec driver. The customer audio driver can call the function to set the audio power dependency. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 ++++++++++++ sound/soc/codecs/hdac_hdmi.h | 6 ++++++ 2 files changed, 18 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..c991407 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1792,6 +1792,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + + return device_link_add(consumer, &hdev->dev, flags); +} +EXPORT_SYMBOL_GPL(hdac_hdmi_device_link_add); + static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { diff --git a/sound/soc/codecs/hdac_hdmi.h b/sound/soc/codecs/hdac_hdmi.h index 4fa2fc9..9239c6b 100644 --- a/sound/soc/codecs/hdac_hdmi.h +++ b/sound/soc/codecs/hdac_hdmi.h @@ -7,4 +7,10 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int pcm, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm); + +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags); + #endif /* __HDAC_HDMI_H__ */ -- 2.7.4

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