February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: Add DT bindings documentation for max9759 amplifier" to the asoc tree
by Mark Brown 20 Feb '18

20 Feb '18

The patch ASoC: Add DT bindings documentation for max9759 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 01182d6529ae499bb73129f4766b45c2ae8e700b Mon Sep 17 00:00:00 2001 From: Neil Armstrong <narmstrong(a)baylibre.com> Date: Mon, 19 Feb 2018 14:45:57 +0100 Subject: [PATCH] ASoC: Add DT bindings documentation for max9759 amplifier This patch adds DT bindings documentation for Maxim MAX9759 speaker amplifier. Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/maxim,max9759.txt | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max9759.txt diff --git a/Documentation/devicetree/bindings/sound/maxim,max9759.txt b/Documentation/devicetree/bindings/sound/maxim,max9759.txt new file mode 100644 index 000000000000..737a996374d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max9759.txt @@ -0,0 +1,18 @@ +Maxim MAX9759 Speaker Amplifier +=============================== + +Required properties: +- compatible : "maxim,max9759" +- shutdown-gpios : the gpio connected to the shutdown pin +- mute-gpios : the gpio connected to the mute pin +- gain-gpios : the 2 gpios connected to the g1 and g2 pins + +Example: + +max9759: analog-amplifier { + compatible = "maxim,max9759"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>; + gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>, + <&gpio3 25 GPIO_ACTIVE_LOW>; +}; -- 2.16.1

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[alsa-devel] Applied "ASoC: max9759: Add Amplifier Driver" to the asoc tree
by Mark Brown 20 Feb '18

20 Feb '18

The patch ASoC: max9759: Add Amplifier Driver 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 fa8d915172b8c10ec0734c4021e99e9705023b07 Mon Sep 17 00:00:00 2001 From: Neil Armstrong <narmstrong(a)baylibre.com> Date: Mon, 19 Feb 2018 14:45:58 +0100 Subject: [PATCH] ASoC: max9759: Add Amplifier Driver The max9759 is a gpio controlled amplifier. Tested on a Variscite Dart MX6 SoM based custom board. Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9759.c | 207 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 214 insertions(+) create mode 100644 sound/soc/codecs/max9759.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2b331f7266ab..f9ea4603a903 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98373 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C + select SND_SOC_MAX9759 select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -1187,6 +1188,10 @@ config SND_SOC_ZX_AUD96P22 config SND_SOC_LM4857 tristate +config SND_SOC_MAX9759 + tristate "Maxim MAX9759 speaker Amplifier" + select GPIOLIB + config SND_SOC_MAX9768 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index da1571336f1e..af4238d90c99 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -80,6 +80,7 @@ snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o +snd-soc-max9759-objs := max9759.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o @@ -325,6 +326,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o +obj-$(CONFIG_SND_SOC_MAX9759) += snd-soc-max9759.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c new file mode 100644 index 000000000000..ecfb4a80424b --- /dev/null +++ b/sound/soc/codecs/max9759.c @@ -0,0 +1,207 @@ +// SPDX-Licence-Identifier: GPL-2.0 +/* + * MAX9759 Amplifier Driver + * + * Copyright (c) 2017 BayLibre, SAS. + * Author: Neil Armstrong <narmstrong(a)baylibre.com> + */ + +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#define DRV_NAME "max9759" + +struct max9759 { + struct gpio_desc *gpiod_shutdown; + struct gpio_desc *gpiod_mute; + struct gpio_descs *gpiod_gain; + bool is_mute; + unsigned int gain; +}; + +static int pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->gpiod_shutdown, 0); + else + gpiod_set_value_cansleep(priv->gpiod_shutdown, 1); + + return 0; +} + +/* From 6dB to 24dB in steps of 6dB */ +static const DECLARE_TLV_DB_SCALE(speaker_gain_tlv, 600, 600, 0); + +static int speaker_gain_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + ucontrol->value.integer.value[0] = priv->gain; + + return 0; +} + +static const bool speaker_gain_table[4][2] = { + /* G1, G2 */ + {true, true}, /* +6dB */ + {false, true}, /* +12dB */ + {true, false}, /* +18dB */ + {false, false}, /* +24dB */ +}; + +static int speaker_gain_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + if (ucontrol->value.integer.value[0] > 3) + return -EINVAL; + + priv->gain = ucontrol->value.integer.value[0]; + + /* G1 */ + gpiod_set_value_cansleep(priv->gpiod_gain->desc[0], + speaker_gain_table[priv->gain][0]); + /* G2 */ + gpiod_set_value_cansleep(priv->gpiod_gain->desc[1], + speaker_gain_table[priv->gain][1]); + + return 1; +} + +static int speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + ucontrol->value.integer.value[0] = !priv->is_mute; + + return 0; +} + +static int speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct max9759 *priv = snd_soc_component_get_drvdata(c); + + priv->is_mute = !ucontrol->value.integer.value[0]; + + gpiod_set_value_cansleep(priv->gpiod_mute, priv->is_mute); + + return 1; +} + +static const struct snd_kcontrol_new max9759_dapm_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Gain Volume", 0, 0, 3, 0, + speaker_gain_control_get, speaker_gain_control_put, + speaker_gain_tlv), + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, + speaker_mute_get, speaker_mute_put), +}; + +static const struct snd_soc_dapm_widget max9759_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + SND_SOC_DAPM_PGA_E("PGA", SND_SOC_NOPM, 0, 0, NULL, 0, pga_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route max9759_dapm_routes[] = { + { "PGA", NULL, "INL" }, + { "PGA", NULL, "INR" }, + { "OUTL", NULL, "PGA" }, + { "OUTR", NULL, "PGA" }, +}; + +static const struct snd_soc_component_driver max9759_component_driver = { + .controls = max9759_dapm_controls, + .num_controls = ARRAY_SIZE(max9759_dapm_controls), + .dapm_widgets = max9759_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9759_dapm_widgets), + .dapm_routes = max9759_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9759_dapm_routes), +}; + +static int max9759_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct max9759 *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + + priv->gpiod_shutdown = devm_gpiod_get(dev, "shutdown", GPIOD_OUT_HIGH); + if (IS_ERR(priv->gpiod_shutdown)) { + err = PTR_ERR(priv->gpiod_shutdown); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'shutdown' gpio: %d", err); + return err; + } + + priv->gpiod_mute = devm_gpiod_get(dev, "mute", GPIOD_OUT_HIGH); + if (IS_ERR(priv->gpiod_mute)) { + err = PTR_ERR(priv->gpiod_mute); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'mute' gpio: %d", err); + return err; + } + priv->is_mute = true; + + priv->gpiod_gain = devm_gpiod_get_array(dev, "gain", GPIOD_OUT_HIGH); + if (IS_ERR(priv->gpiod_gain)) { + err = PTR_ERR(priv->gpiod_gain); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'gain' gpios: %d", err); + return err; + } + priv->gain = 0; + + if (priv->gpiod_gain->ndescs != 2) { + dev_err(dev, "Invalid 'gain' gpios count: %d", + priv->gpiod_gain->ndescs); + return -EINVAL; + } + + return devm_snd_soc_register_component(dev, &max9759_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id max9759_ids[] = { + { .compatible = "maxim,max9759", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9759_ids); +#endif + +static struct platform_driver max9759_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(max9759_ids), + }, + .probe = max9759_probe, +}; + +module_platform_driver(max9759_driver); + +MODULE_DESCRIPTION("ASoC MAX9759 amplifier driver"); +MODULE_AUTHOR("Neil Armstrong <narmstrong(a)baylibre.com>"); +MODULE_LICENSE("GPL"); -- 2.16.1

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[alsa-devel] [PATCH 0/2] ASoC: Add support for max9759 Amplifier Driver
by Neil Armstrong 20 Feb '18

20 Feb '18

The MAX9759 is a gpio controlled speaker amplifier, this patchset add the DT Bindings and the corresponding ASoC component driver. Neil Armstrong (2): ASoC: Add DT bindings documentation for max9759 amplifier ASoC: max9759: Add Amplifier Driver .../devicetree/bindings/sound/maxim,max9759.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9759.c | 207 +++++++++++++++++++++ 4 files changed, 232 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max9759.txt create mode 100644 sound/soc/codecs/max9759.c -- 2.7.4

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[alsa-devel] [4.4-stable 10/22] ASoC: Intel: Kconfig: fix build when ACPI is not enabled
by Arnd Bergmann 20 Feb '18

20 Feb '18

From: Vinod Koul <vinod.koul(a)intel.com> commit 3493d4a86457c7de9f1e602b4267c9b0f9ec1c9f upstream. Randy reported following error when ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI +which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, CONFIG_SND_SST_IPC_ACPI was already dependent upon ACPI, but that was not solving it. So move the depends up to machine drivers and remove from CONFIG_SND_SST_IPC_ACPI. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> [arnd: rebased to 4.4-stable kernel] Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d430ef5a4f38..79c29330c56a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -24,7 +24,6 @@ config SND_SST_IPC_PCI config SND_SST_IPC_ACPI tristate select SND_SST_IPC - depends on ACPI config SND_SOC_INTEL_SST tristate @@ -91,7 +90,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -103,7 +102,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -115,7 +114,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- 2.9.0

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[alsa-devel] [PATCH] ASoC: uniphier: evea: fix typo 'eva' -> 'evea'
by Katsuhiro Suzuki 20 Feb '18

20 Feb '18

Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index cad7e60de21d..ac2d30b55fb6 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -292,7 +292,7 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } -static const struct snd_kcontrol_new eva_controls[] = { +static const struct snd_kcontrol_new evea_controls[] = { SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, @@ -384,8 +384,8 @@ static struct snd_soc_component_driver soc_codec_evea = { .num_dapm_widgets = ARRAY_SIZE(evea_widgets), .dapm_routes = evea_routes, .num_dapm_routes = ARRAY_SIZE(evea_routes), - .controls = eva_controls, - .num_controls = ARRAY_SIZE(eva_controls), + .controls = evea_controls, + .num_controls = ARRAY_SIZE(evea_controls), .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- 2.16.1

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[alsa-devel] Need help implementing an external PCM/control plugin pair
by Dimitris Papavasiliou 20 Feb '18

20 Feb '18

Hi, I'm in the process of putting together a loudness compensation plugin for ALSA. The filtering part is implemented as an external PCM plugin and it is, more or less, ready, meaning that you can set its parameters via configuration options and it'll happily filter away in real time. I'm trying now to create a matching mixer plugin, to allow on-the-fly adjustment of the parameters, and I'm not sure what the best way of accomplishing that would be. My main concerns are: 1. I've figured out how to create a basic external control plugin, that would allow me to expose a set of controls, which the user can fiddle with, using alsamixer et al., but what would be the best way of communicating changes in the values to the PCM plugin? I could imagine using methods external to ALSA, such as a file with stored values, which the ctl plugin updates and which the PCM plugin monitors for changes, or perhaps I could try to open the mixer from inside the PCM plugin, set callbacks for each of its elements and enter a snd_mixer_wait/snd_mixer_handle_events loop, in a separate thread of course. Is there reason to prefer one way over the other? Perhaps there is a better way, I'm not aware of? 2. I'd like to support event subscription/notification, but I'm not sure how to make that work. My understanding (from searching the net and looking at ALSA's source), is that I need to provide some sort of file descriptor, which the subscribing code can poll, so as to block until there's an update. Since I have no obvious candidates for such a file, I've tried creating a pipe, handing one end to the control plugin upon initialization and then writing single bytes to the other end when there's a change in one of the controls. This works but, as far as I can see, the pipe is never actually read from, from the subscribing code, so that I would have to manually read from it somehwere, say in the read_event callback, in order to flush it. In the end, I'll probably use the descriptor from an inotify watch on the file where the plugin stores the control settings, but is this a valid approach overall? Is there some documentation specifying the exact way in which poll descriptors are handled by ALSA (in the case of an extrnal CTL plugin), that I've missed? If not, are my assumptions above (that the descriptor is only polled and that its contents are never actually read) correct? 3. Finally, and although this is not essential, I'd like to provide metadata for some controls, to support dB ranges, as that's the most natural way to set them. I see hints that this is supported for external control plugins (SND_CTL_EXT_ACCESS_TLV_READ/WRITE, the snd_ctl_ext_tlv_rw_t callback), but I'm not sure how to implement it. Is it documented somewhere? If not, could someone describe the basic approach? Thanks, Dimitris

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[alsa-devel] [PATCH v4 0/2] Add support for AK5558 ADC
by Daniel Baluta 19 Feb '18

19 Feb '18

We support normal mode, TDM mode and pm. Changes since v3: [addressed Mark's comments] * use C++ comments for SPDX identifier * removed some unused widgets and controls (channel enable MUXes, TDM, DSD related stuff) * removed set_dai_mute, set_bias_level, set_dai_sysclk as they are not used. * refactored probe/remove to have symmetric operations * use snd_soc_update_bits instead of hardcode read/write operation. Changes since v2: [addressed comments from Andy, Fabio and Rob] * sort include files * use probe_new * reword the binding document * use adc@10 instead of ak5558@10 * remove file name at the beginning of codec source code. * make i2c_probe and i2c_remove parameters naming consistent. Changes since v1: [addressed comments from Andy and Fabio] * fix GPIO polarity from active high to active low for correct documentation * fix license header by using SPDX identifier * remove debug prints at the beginning of functions. * only support auto clock switching (manual switching was dead code anyway) (in the future we could add a DT property to choose between manual and auto) * Use gpiod API * use GENMASK * introduce power_off/power_on Daniel Baluta (2): ASoC: codecs: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC Documentation/devicetree/bindings/sound/ak5558.txt | 22 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 417 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 52 +++ 5 files changed, 499 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h -- 2.7.4

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[alsa-devel] [PATCH] ASoC: ADAU7002: Adding ACPI id
by Akshu Agrawal 19 Feb '18

19 Feb '18

Adding ACPI entry id for probing the driver. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/codecs/adau7002.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c index 49475e3..fdff868 100644 --- a/sound/soc/codecs/adau7002.c +++ b/sound/soc/codecs/adau7002.c @@ -7,6 +7,7 @@ * Licensed under the GPL-2. */ +#include <linux/acpi.h> #include <linux/init.h> #include <linux/module.h> #include <linux/of.h> @@ -69,10 +70,19 @@ static int adau7002_remove(struct platform_device *pdev) MODULE_DEVICE_TABLE(of, adau7002_dt_ids); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id adau7002_acpi_match[] = { + { "ADAU7002", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, adau7002_acpi_match); +#endif + static struct platform_driver adau7002_driver = { .driver = { .name = "adau7002", .of_match_table = of_match_ptr(adau7002_dt_ids), + .acpi_match_table = ACPI_PTR(adau7002_acpi_match), }, .probe = adau7002_probe, .remove = adau7002_remove, -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: tlv320aic31xx: Optimize the master mode I2S clock handling
by Peter Ujfalusi 19 Feb '18

19 Feb '18

Hi, Instead of forcing DAC/ADC to be turned on, we can use P0 R29, Bit2 to let the I2S clocks running, but we do not want it to run all the time and we do not want I2S clocks when it is not needed (analog loopback mode). Use DAPM_SUPPLY for the bit and connect it to the in/out AIF. The DAPM routes can be modify to not include the DAC/ADC. Regards, Peter --- Peter Ujfalusi (2): ASoC: tlv320aic31xx: Rename AIF_IN from 'DAC IN' to 'AIF IN' ASoC: tlv320aic31xx: Do not force power on the DAC/ADC in clock master mode sound/soc/codecs/tlv320aic31xx.c | 46 +++++++++++++++++++++++++--------------- sound/soc/codecs/tlv320aic31xx.h | 1 + 2 files changed, 30 insertions(+), 17 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] [PATCH 0/4] ASoC: missing replace codec to component
by Kuninori Morimoto 19 Feb '18

19 Feb '18

Hi Mark These are new added driver on mark/for-next branch, but based on old "codec" framework. We want to convert it to component. These patches are do it Kuninori Morimoto (4): ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/codecs/ak4458.c | 105 +++++++++++++------------- sound/soc/codecs/ak5558.c | 65 ++++++++-------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 12 +-- 4 files changed, 94 insertions(+), 92 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

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