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

[alsa-devel] [PATCH v2 0/5] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 11 Oct '19

11 Oct '19

Thanks for reviewing the RFC patchset. Here is new patchset addressing all the comments from RFC. This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is on DragonBoard DB845c based of SDM845 SoC. WCD934x and WSA881x patches will be posted soon. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since RFC: - updated bindings as suggested to take care of more bus parameters. - fixed error code of snd_soc_dai_get_sdw_stream() dummy function. - Cleaned up driver to handle read/writes in same way without special casing. - removed unused defines Srinivas Kandagatla (4): soundwire: stream: make stream name a const pointer ASoC: core: add support to snd_soc_dai_get_sdw_stream() dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller Vinod Koul (1): soundwire: Add compute_params callback .../bindings/soundwire/qcom,sdw.txt | 167 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 919 ++++++++++++++++++ drivers/soundwire/stream.c | 12 +- include/linux/soundwire/sdw.h | 6 +- include/sound/soc-dai.h | 10 + 7 files changed, 1124 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

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[alsa-devel] [PATCH 20/36] ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS
by Arnd Bergmann 11 Oct '19

11 Oct '19

The constant requires indirectly including a machine header file, but it's not actually used any more since commit 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API"), so remove it completely. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c-i2s-v2.h | 3 +-- sound/soc/samsung/s3c2412-i2s.c | 2 +- 3 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 7e196b599be1..6a254fdaa6fb 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -616,8 +616,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) + struct s3c_i2sv2_info *i2s) { struct device *dev = dai->dev; unsigned int iismod; diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index fe42b77999fd..8c6fc0d3d77e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -83,8 +83,7 @@ extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, * @base: The base address for the registers. */ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); + struct s3c_i2sv2_info *i2s); /** * s3c_i2sv2_cleanup - cleanup resources allocated in s3c_i2sv2_probe diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..5bf571762e89 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -49,7 +49,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, &s3c2412_i2s_pcm_stereo_in); - ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s); if (ret) return ret; -- 2.20.0

2 1

[alsa-devel] Counting links in simple-card
by Daniel Baluta 11 Oct '19

11 Oct '19

Hi Morimoto-san, In simple-card.c simple_get_dais_count function there is this comment: * ex1) * CPU0 --- Codec0 * CPU1 --- Codec1 * CPU2 -----/ * CPU3 --- Codec2 * * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec * => 7 DAIs = 4xCPU + 3xCodec * => 1 ccnf = 1xdummy-Codec So, I don't understand exactly how did you get the formula for links. AFAICT there should be 3 x CPU-Codec + 1 x CPU-dummy. thanks, Daniel.

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[alsa-devel] [PATCH v2] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 10 Oct '19

10 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added type and description and removed number of items from 'model' property. Best regards, Maciej Falkowski --- .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2-audio.yaml | 84 +++++++++++++++++++ 2 files changed, 84 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..62b5b9a1dec0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are + the WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + +required: + - compatible + - model + - audio-codec + - i2s-controller + - audio-amplifier + - samsung,audio-routing + - mic-bias-gpios + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

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[alsa-devel] [PATCH 19/36] ARM: s3c: rx1950: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-rx1950.c | 16 +++++++ sound/soc/samsung/rx1950_uda1380.c | 72 +++++++++-------------------- 2 files changed, 39 insertions(+), 49 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index 29f9b345a531..9f3b0f18cd36 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -724,6 +724,20 @@ static struct i2c_board_info rx1950_i2c_devices[] = { }, }; +static struct gpiod_lookup_table rx1950_audio_gpio_table = { + .dev_id = "rx1950-audio", + .table = { + GPIO_LOOKUP("GPIOG", 12, "hp-gpio", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOA", 1, "speaker-power", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device rx1950_audio = { + .name = "rx1950-audio", + .id = -1, +}; + static struct platform_device *rx1950_devices[] __initdata = { &s3c2410_device_dclk, &s3c_device_lcd, @@ -742,6 +756,7 @@ static struct platform_device *rx1950_devices[] __initdata = { &power_supply, &rx1950_battery, &rx1950_leds, + &rx1950_audio, }; static void __init rx1950_map_io(void) @@ -809,6 +824,7 @@ static void __init rx1950_init_machine(void) gpio_direction_output(S3C2410_GPJ(6), 0); pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); + gpiod_add_lookup_table(&rx1950_audio_gpio_table); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 4b247e91ae5b..cf96eaaf6b4e 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -12,16 +12,13 @@ // Vasily Khoruzhick <anarsoul(a)gmail.com> #include <linux/types.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/jack.h> -#include <mach/gpio-samsung.h> #include "regs-iis.h" -#include <asm/mach-types.h> - #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); @@ -58,7 +55,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { [0] = { - .gpio = S3C2410_GPG(12), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,8 +119,6 @@ static struct snd_soc_card rx1950_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static struct platform_device *s3c24xx_snd_device; - static int rx1950_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -134,13 +128,15 @@ static int rx1950_startup(struct snd_pcm_substream *substream) &hw_rates); } +struct gpio_desc *gpiod_speaker_power; + static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C2410_GPA(1), 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C2410_GPA(1), 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -214,57 +210,35 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int __init rx1950_init(void) +static int rx1950_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_rx1950()) - return -ENODEV; + struct device *dev = &pdev->dev; /* configure some gpios */ - ret = gpio_request(S3C2410_GPA(1), "speaker-power"); - if (ret) - goto err_gpio; - - ret = gpio_direction_output(S3C2410_GPA(1), 0); - if (ret) - goto err_gpio_conf; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_plat_alloc; - } - - platform_set_drvdata(s3c24xx_snd_device, &rx1950_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) { - platform_device_put(s3c24xx_snd_device); - goto err_plat_add; + gpiod_speaker_power = devm_gpiod_get(dev, "speaker-power", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "cannot get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - return 0; - -err_plat_add: -err_plat_alloc: -err_gpio_conf: - gpio_free(S3C2410_GPA(1)); + hp_jack_gpios[0].gpiod_dev = dev; + rx1950_asoc.dev = dev; -err_gpio: - return ret; + return devm_snd_soc_register_card(dev, &rx1950_asoc); } -static void __exit rx1950_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C2410_GPA(1)); -} +struct platform_driver rx1950_audio = { + .driver = { + .name = "rx1950-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = rx1950_probe, +}; -module_init(rx1950_init); -module_exit(rx1950_exit); +module_platform_driver(rx1950_audio); /* Module information */ MODULE_AUTHOR("Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC RX1950"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:rx1950-audio"); -- 2.20.0

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[alsa-devel] [PATCH 18/36] ARM: s3c: gta02-audio: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-gta02.c | 17 ++++++ sound/soc/samsung/neo1973_wm8753.c | 85 +++++++++--------------------- 2 files changed, 43 insertions(+), 59 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 594901f3b8e5..526fd0933289 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -15,6 +15,7 @@ #include <linux/delay.h> #include <linux/timer.h> #include <linux/init.h> +#include <linux/gpio/machine.h> #include <linux/gpio.h> #include <linux/gpio_keys.h> #include <linux/workqueue.h> @@ -474,6 +475,20 @@ static struct platform_device gta02_buttons_device = { }, }; +static struct gpiod_lookup_table gta02_audio_gpio_table = { + .dev_id = "neo1973-audio", + .table = { + GPIO_LOOKUP("GPIOJ", 2, "amp-shut", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOJ", 1, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device gta02_audio = { + .name = "neo1973-audio", + .id = -1, +}; + static void __init gta02_map_io(void) { s3c24xx_init_io(gta02_iodesc, ARRAY_SIZE(gta02_iodesc)); @@ -498,6 +513,7 @@ static struct platform_device *gta02_devices[] __initdata = { &gta02_buttons_device, &s3c_device_adc, &s3c_device_ts, + &gta02_audio, }; static void gta02_poweroff(void) @@ -524,6 +540,7 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 38f536bafa09..b41e3dc297e7 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -11,14 +11,11 @@ #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/soc.h> -#include <mach/gpio-samsung.h> -#include <asm/mach-types.h> #include "regs-iis.h" - #include "../codecs/wm8753.h" #include "s3c24xx-i2s.h" @@ -166,6 +163,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; +static struct gpio_desc *gpiod_hp_in, *gpiod_amp_shut; static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -173,7 +171,7 @@ static int lm4853_set_spk(struct snd_kcontrol *kcontrol, { gta02_speaker_enabled = ucontrol->value.integer.value[0]; - gpio_set_value(S3C2410_GPJ(2), !gta02_speaker_enabled); + gpiod_set_value(gpiod_hp_in, !gta02_speaker_enabled); return 0; } @@ -188,7 +186,7 @@ static int lm4853_get_spk(struct snd_kcontrol *kcontrol, static int lm4853_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(S3C2410_GPJ(1), SND_SOC_DAPM_EVENT_OFF(event)); + gpiod_set_value(gpiod_amp_shut, SND_SOC_DAPM_EVENT_OFF(event)); return 0; } @@ -308,13 +306,8 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -static const struct gpio neo1973_gta02_gpios[] = { - { S3C2410_GPJ(2), GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, - { S3C2410_GPJ(1), GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, -}; - static struct snd_soc_card neo1973 = { - .name = "neo1973", + .name = "neo1973gta02", .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), @@ -332,62 +325,36 @@ static struct snd_soc_card neo1973 = { .fully_routed = true, }; -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) +static int neo1973_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_neo1973_gta02()) - return -ENODEV; + struct device *dev = &pdev->dev; - if (machine_is_neo1973_gta02()) { - neo1973.name = "neo1973gta02"; - neo1973.num_aux_devs = 1; - - ret = gpio_request_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - if (ret) - return ret; + gpiod_hp_in = devm_gpiod_get(dev, "hp", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_hp_in)) { + dev_err(dev, "missing gpio %s\n", "hp"); + return PTR_ERR(gpiod_hp_in); } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) { - ret = -ENOMEM; - goto err_gpio_free; + gpiod_amp_shut = devm_gpiod_get(dev, "amp-shut", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_amp_shut)) { + dev_err(dev, "missing gpio %s\n", "amp-shut"); + return PTR_ERR(gpiod_amp_shut); } - platform_set_drvdata(neo1973_snd_device, &neo1973); - ret = platform_device_add(neo1973_snd_device); - - if (ret) - goto err_put_device; - - return 0; - -err_put_device: - platform_device_put(neo1973_snd_device); -err_gpio_free: - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } - return ret; + neo1973.dev = dev; + return devm_snd_soc_register_card(dev, &neo1973); } -module_init(neo1973_init); - -static void __exit neo1973_exit(void) -{ - platform_device_unregister(neo1973_snd_device); - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } -} -module_exit(neo1973_exit); +struct platform_driver neo1973_audio = { + .driver = { + .name = "neo1973-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = neo1973_probe, +}; +module_platform_driver(neo1973_audio); /* Module information */ MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:neo1973-audio"); -- 2.20.0

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[alsa-devel] [PATCH 17/36] ARM: s3c: h1940-audio: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-h1940.c | 18 ++++++++ sound/soc/samsung/h1940_uda1380.c | 71 ++++++++++-------------------- 2 files changed, 41 insertions(+), 48 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index 74d6b68e91c7..922d53bacbdf 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -475,6 +475,22 @@ static struct gpiod_lookup_table h1940_mmc_gpio_table = { }, }; +static struct gpiod_lookup_table h1940_audio_gpio_table = { + .dev_id = "h1940-audio", + .table = { + GPIO_LOOKUP("H1940_LATCH", + H1940_LATCH_AUDIO_POWER - H1940_LATCH_GPIO(0), + "speaker-power", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOG", 4, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device h1940_audio = { + .name = "h1940-audio", + .id = -1, +}; + static struct pwm_lookup h1940_pwm_lookup[] = { PWM_LOOKUP("samsung-pwm", 0, "pwm-backlight", NULL, 36296, PWM_POLARITY_NORMAL), @@ -652,6 +668,7 @@ static struct platform_device *h1940_devices[] __initdata = { &s3c_device_ts, &power_supply, &h1940_battery, + &h1940_audio, }; static void __init h1940_map_io(void) @@ -691,6 +708,7 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); + gpiod_add_lookup_table(&h1940_audio_gpio_table); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index a95c34e53a2b..5fc001b4effd 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -15,9 +15,6 @@ #include <sound/jack.h> #include "regs-iis.h" -#include <asm/mach-types.h> - -#include <mach/gpio-samsung.h> #include "s3c24xx-i2s.h" static const unsigned int rates[] = { @@ -31,6 +28,8 @@ static const struct snd_pcm_hw_constraint_list hw_rates = { .list = rates, }; +static struct gpio_desc *gpiod_speaker_power; + static struct snd_soc_jack hp_jack; static struct snd_soc_jack_pin hp_jack_pins[] = { @@ -47,7 +46,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { { - .gpio = S3C2410_GPG(4), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,9 +121,9 @@ static int h1940_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C_GPIO_END + 9, 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C_GPIO_END + 9, 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -151,8 +149,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"VINM", NULL, "Mic Jack"}, }; -static struct platform_device *s3c24xx_snd_device; - static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, @@ -194,55 +190,34 @@ static struct snd_soc_card h1940_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static int __init h1940_init(void) +static int h1940_probe(struct platform_device *pdev) { - int ret; + struct device *dev = &pdev->dev; - if (!machine_is_h1940()) - return -ENODEV; + h1940_asoc.dev = dev; + hp_jack_gpios[0].gpiod_dev = dev; + gpiod_speaker_power = devm_gpiod_get(&pdev->dev, "speaker-power", + GPIOD_OUT_LOW); - /* configure some gpios */ - ret = gpio_request(S3C_GPIO_END + 9, "speaker-power"); - if (ret) - goto err_out; - - ret = gpio_direction_output(S3C_GPIO_END + 9, 0); - if (ret) - goto err_gpio; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_gpio; + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "Could not get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) - goto err_plat; - - return 0; - -err_plat: - platform_device_put(s3c24xx_snd_device); -err_gpio: - gpio_free(S3C_GPIO_END + 9); - -err_out: - return ret; -} - -static void __exit h1940_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C_GPIO_END + 9); + return devm_snd_soc_register_card(dev, &h1940_asoc); } -module_init(h1940_init); -module_exit(h1940_exit); +static struct platform_driver h1940_audio_driver = { + .driver = { + .name = "h1940-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = h1940_probe, +}; +module_platform_driver(h1940_audio_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC H1940"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:h1940-audio"); -- 2.20.0

1 0

[alsa-devel] Record at 384 KHz
by GitHub issues - opened 10 Oct '19

10 Oct '19

alsa-project/alsa-lib issue #12 was opened from paddygoat: Tried everything I can think of to record at 384. How do I do it? Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/12 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH 0/9] ASoC: SOF updates for Intel/i.MX
by Pierre-Louis Bossart 10 Oct '19

10 Oct '19

Small set of updates for Intel and NXP platforms. On the Intel side, the code is now aligned with the HDaudio legacy with the sync_write. OEMs reported concerns with mute LEDs not supported with SOF/UCM, we can now tie a topology-defined control with a mute LED. Note that the large set of differences in the topology is largely due to code move, the new functionality is minimal in size. On the i.MX side, add topology support for ESAI parameters. This reflects on-going work on the firmware side. And last add debug module parameters for ACPI and PCI modules. For now the only flag used is for PM runtime, e.g. to measure the suspend-resume latencies of child devices without dealing with DSP reinitialization or keep the parent active while testing the SoundWire clock-stop-mode. Daniel Baluta (2): ASoC: SOF: imx: Describe ESAI parameters to be sent to DSP ASoC: SOF: imx: Read ESAI parameters and send them to DSP Jaska Uimonen (3): ASoC: SOF: enable dual control for pga AsoC: SOF: refactor control load code ASoC: SOF: acpi led support for switch controls Kai Vehmanen (2): ASoC: SOF: enable sync_write in hdac_bus Revert "ASoC: SOF: Force polling mode on CFL and CNL" Pierre-Louis Bossart (2): ASoC: SOF: acpi: add debug module param ASoC: SOF: pci: add debug module param include/sound/sof/dai-imx.h | 34 +++ include/sound/sof/dai.h | 2 + include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 7 +- sound/soc/sof/control.c | 32 +++ sound/soc/sof/intel/hda.c | 8 +- sound/soc/sof/pcm.c | 8 + sound/soc/sof/sof-acpi-dev.c | 12 +- sound/soc/sof/sof-pci-dev.c | 12 +- sound/soc/sof/sof-priv.h | 9 + sound/soc/sof/topology.c | 401 +++++++++++++++++++------------- 11 files changed, 354 insertions(+), 173 deletions(-) create mode 100644 include/sound/sof/dai-imx.h -- 2.20.1

2 20

[alsa-devel] Applied "ASoC: tas2562: Introduce the TAS2562 amplifier" to the asoc tree
by Mark Brown 10 Oct '19

10 Oct '19

The patch ASoC: tas2562: Introduce the TAS2562 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 c173dba44c2d2ec2adaa4e607e9729b74236bd57 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Tue, 8 Oct 2019 13:15:17 -0500 Subject: [PATCH] ASoC: tas2562: Introduce the TAS2562 amplifier Introduce the Texas Instruments TAS2562 amplifier driver with I/V sense for loud speaker applications. http://www.ti.com/lit/gpn/tas2562 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20191008181517.5332-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2562.c | 590 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 85 ++++++ 4 files changed, 682 insertions(+) create mode 100644 sound/soc/codecs/tas2562.c create mode 100644 sound/soc/codecs/tas2562.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ae213c6392ec..f4747ebc251e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2562 if I2C select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C @@ -1133,6 +1134,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2562 + tristate "Texas Instruments TAS2562 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2118407a1dd5..ddfd07071925 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-max98504-objs := max98504.o snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o +snd-soc-tas2562-objs := tas2562.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -481,6 +482,7 @@ obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o +obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c new file mode 100644 index 000000000000..729acd874c48 --- /dev/null +++ b/sound/soc/codecs/tas2562.c @@ -0,0 +1,590 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for the Texas Instruments TAS2562 CODEC +// Copyright (C) 2019 Texas Instruments Inc. + + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> +#include <linux/delay.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tas2562.h" + +#define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FORMAT_S32_LE) + +struct tas2562_data { + struct snd_soc_component *component; + struct gpio_desc *sdz_gpio; + struct regmap *regmap; + struct device *dev; + struct i2c_client *client; + int v_sense_slot; + int i_sense_slot; +}; + +static int tas2562_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas2562_data *tas2562 = + snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_ACTIVE); + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_MUTE); + break; + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_SHUTDOWN); + break; + + default: + dev_err(tas2562->dev, + "wrong power level setting %d\n", level); + return -EINVAL; + } + + return 0; +} + +static int tas2562_set_samplerate(struct tas2562_data *tas2562, int samplerate) +{ + int samp_rate; + int ramp_rate; + + switch (samplerate) { + case 7350: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ; + break; + case 8000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ; + break; + case 14700: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ; + break; + case 16000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ; + break; + case 22050: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ; + break; + case 24000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ; + break; + case 29400: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ; + break; + case 32000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ; + break; + case 44100: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ; + break; + case 48000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ; + break; + case 88200: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ; + break; + case 96000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ; + break; + case 176400: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ; + break; + case 192000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ; + break; + default: + dev_info(tas2562->dev, "%s, unsupported sample rate, %d\n", + __func__, samplerate); + return -EINVAL; + } + + snd_soc_component_update_bits(tas2562->component, TAS2562_TDM_CFG0, + TAS2562_TDM_CFG0_RAMPRATE_MASK, ramp_rate); + snd_soc_component_update_bits(tas2562->component, TAS2562_TDM_CFG0, + TAS2562_TDM_CFG0_SAMPRATE_MASK, samp_rate); + + return 0; +} + +static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret = 0; + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_16B); + break; + case 24: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_24B); + break; + case 32: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_32B); + break; + + case 0: + /* Do not change slot width */ + break; + default: + dev_err(tas2562->dev, "slot width not supported"); + ret = -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) +{ + int ret; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_16B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_24B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_32B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 4; + break; + + default: + dev_info(tas2562->dev, "Not supported params format\n"); + } + + ret = snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG5, + TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, + TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG6, + TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, + TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2562_set_bitwidth(tas2562, params_format(params)); + if (ret) { + dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + return ret; + } + + ret = tas2562_set_samplerate(tas2562, params_rate(params)); + if (ret) + dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + + return ret; +} + +static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2562_TDM_CFG1_RX_FALLING; + break; + default: + dev_err(tas2562->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, + TAS2562_TDM_CFG1_RX_EDGE_MASK, + asi_cfg_1); + if (ret < 0) { + dev_err(tas2562->dev, "Failed to set RX edge\n"); + return ret; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + case (SND_SOC_DAIFMT_DSP_A): + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = BIT(1); + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(tas2562->dev, "DAI Format is not found, fmt=0x%x\n", + fmt); + ret = -EINVAL; + break; + } + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, + TAS2562_TDM_CFG1_RX_OFFSET_MASK, + tdm_rx_start_slot); + + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + return snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + mute ? TAS2562_MUTE : 0); +} + +static int tas2562_codec_probe(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + + tas2562->component = component; + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 1); + + ret = snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_MUTE); + if (ret < 0) + return ret; + + return 0; +} + +#ifdef CONFIG_PM +static int tas2562_suspend(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(tas2562->regmap, true); + regcache_mark_dirty(tas2562->regmap); + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 0); + + return 0; +} + +static int tas2562_resume(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 1); + + regcache_cache_only(tas2562->regmap, false); + + return regcache_sync(tas2562->regmap); +} +#else +#define tas2562_suspend NULL +#define tas2562_resume NULL +#endif + +static const char * const tas2562_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL(tas2562_ASI1_src_enum, TAS2562_TDM_CFG2, 4, + tas2562_ASI1_src); + +static const struct snd_kcontrol_new tas2562_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2562_ASI1_src_enum); + +static int tas2562_dac_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 tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + break; + default: + break; + } + + return 0; +} + +static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2562_PWR_CTRL, TAS2562_ISENSE_POWER_EN, + 1, 1); + +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2562_PWR_CTRL, TAS2562_VSENSE_POWER_EN, + 1, 1); + +static const struct snd_kcontrol_new tas2562_snd_controls[] = { + SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, + tas2562_dac_tlv), +}; + +static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), + SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2562_PWR_CTRL, 2, 1, &vsense_switch), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON"), + SND_SOC_DAPM_OUTPUT("OUT"), +}; + +static const struct snd_soc_dapm_route tas2562_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + { "DAC", NULL, "DAC IN" }, + { "OUT", NULL, "DAC" }, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static const struct snd_soc_component_driver soc_component_dev_tas2562 = { + .probe = tas2562_codec_probe, + .suspend = tas2562_suspend, + .resume = tas2562_resume, + .set_bias_level = tas2562_set_bias_level, + .controls = tas2562_snd_controls, + .num_controls = ARRAY_SIZE(tas2562_snd_controls), + .dapm_widgets = tas2562_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2562_dapm_widgets), + .dapm_routes = tas2562_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2562_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { + .hw_params = tas2562_hw_params, + .set_fmt = tas2562_set_dai_fmt, + .set_tdm_slot = tas2562_set_dai_tdm_slot, + .digital_mute = tas2562_mute, +}; + +static struct snd_soc_dai_driver tas2562_dai[] = { + { + .name = "tas2562-amplifier", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, + .ops = &tas2562_speaker_dai_ops, + }, +}; + +static const struct regmap_range_cfg tas2562_ranges[] = { + { + .range_min = 0, + .range_max = 5 * 128, + .selector_reg = TAS2562_PAGE_CTRL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct reg_default tas2562_reg_defaults[] = { + { TAS2562_PAGE_CTRL, 0x00 }, + { TAS2562_SW_RESET, 0x00 }, + { TAS2562_PWR_CTRL, 0x0e }, + { TAS2562_PB_CFG1, 0x20 }, + { TAS2562_TDM_CFG0, 0x09 }, + { TAS2562_TDM_CFG1, 0x02 }, +}; + +static const struct regmap_config tas2562_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = 5 * 128, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = tas2562_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2562_reg_defaults), + .ranges = tas2562_ranges, + .num_ranges = ARRAY_SIZE(tas2562_ranges), +}; + +static int tas2562_parse_dt(struct tas2562_data *tas2562) +{ + struct device *dev = tas2562->dev; + int ret = 0; + + tas2562->sdz_gpio = devm_gpiod_get_optional(dev, "shut-down-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2562->sdz_gpio)) { + if (PTR_ERR(tas2562->sdz_gpio) == -EPROBE_DEFER) { + tas2562->sdz_gpio = NULL; + return -EPROBE_DEFER; + } + } + + ret = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2562->i_sense_slot); + if (ret) + dev_err(dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", ret); + + return ret; +} + +static int tas2562_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev = &client->dev; + struct tas2562_data *data; + int ret; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->client = client; + data->dev = &client->dev; + + tas2562_parse_dt(data); + + data->regmap = devm_regmap_init_i2c(client, &tas2562_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(dev, "failed to allocate register map: %d\n", ret); + return ret; + } + + dev_set_drvdata(&client->dev, data); + + return devm_snd_soc_register_component(dev, &soc_component_dev_tas2562, + tas2562_dai, + ARRAY_SIZE(tas2562_dai)); + +} + +static const struct i2c_device_id tas2562_id[] = { + { "tas2562", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2562_id); + +static const struct of_device_id tas2562_of_match[] = { + { .compatible = "ti,tas2562", }, + { }, +}; +MODULE_DEVICE_TABLE(of, tas2562_of_match); + +static struct i2c_driver tas2562_i2c_driver = { + .driver = { + .name = "tas2562", + .of_match_table = of_match_ptr(tas2562_of_match), + }, + .probe = tas2562_probe, + .id_table = tas2562_id, +}; + +module_i2c_driver(tas2562_i2c_driver); + +MODULE_AUTHOR("Dan Murphy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2562 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h new file mode 100644 index 000000000000..60f2bb1d198b --- /dev/null +++ b/sound/soc/codecs/tas2562.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * tas2562.h - ALSA SoC Texas Instruments TAS2562 Mono Audio Amplifier + * + * Copyright (C) 2019 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + */ + +#ifndef __TAS2562_H__ +#define __TAS2562_H__ + +#define TAS2562_PAGE_CTRL 0x00 + +#define TAS2562_REG(page, reg) ((page * 128) + reg) + +#define TAS2562_SW_RESET TAS2562_REG(0, 0x01) +#define TAS2562_PWR_CTRL TAS2562_REG(0, 0x02) +#define TAS2562_PB_CFG1 TAS2562_REG(0, 0x03) +#define TAS2562_MISC_CFG1 TAS2562_REG(0, 0x04) +#define TAS2562_MISC_CFG2 TAS2562_REG(0, 0x05) + +#define TAS2562_TDM_CFG0 TAS2562_REG(0, 0x06) +#define TAS2562_TDM_CFG1 TAS2562_REG(0, 0x07) +#define TAS2562_TDM_CFG2 TAS2562_REG(0, 0x08) +#define TAS2562_TDM_CFG3 TAS2562_REG(0, 0x09) +#define TAS2562_TDM_CFG4 TAS2562_REG(0, 0x0a) +#define TAS2562_TDM_CFG5 TAS2562_REG(0, 0x0b) +#define TAS2562_TDM_CFG6 TAS2562_REG(0, 0x0c) +#define TAS2562_TDM_CFG7 TAS2562_REG(0, 0x0d) +#define TAS2562_TDM_CFG8 TAS2562_REG(0, 0x0e) +#define TAS2562_TDM_CFG9 TAS2562_REG(0, 0x0f) +#define TAS2562_TDM_CFG10 TAS2562_REG(0, 0x10) +#define TAS2562_TDM_DET TAS2562_REG(0, 0x11) +#define TAS2562_REV_ID TAS2562_REG(0, 0x7d) + +/* Page 2 */ +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) + +#define TAS2562_RESET BIT(0) + +#define TAS2562_MODE_MASK 0x3 +#define TAS2562_ACTIVE 0x0 +#define TAS2562_MUTE 0x1 +#define TAS2562_SHUTDOWN 0x2 + +#define TAS2562_TDM_CFG1_RX_EDGE_MASK BIT(0) +#define TAS2562_TDM_CFG1_RX_FALLING 1 +#define TAS2562_TDM_CFG1_RX_OFFSET_MASK GENMASK(4, 0) + +#define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) +#define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) +#define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 + +#define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) + +#define TAS2562_TDM_CFG2_RXLEN_MASK GENMASK(0, 1) +#define TAS2562_TDM_CFG2_RXLEN_16B 0x0 +#define TAS2562_TDM_CFG2_RXLEN_24B BIT(0) +#define TAS2562_TDM_CFG2_RXLEN_32B BIT(1) + +#define TAS2562_TDM_CFG2_RXWLEN_MASK GENMASK(3, 2) +#define TAS2562_TDM_CFG2_RXWLEN_16B 0x0 +#define TAS2562_TDM_CFG2_RXWLEN_20B BIT(2) +#define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) +#define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) + +#define TAS2562_VSENSE_POWER_EN BIT(2) +#define TAS2562_ISENSE_POWER_EN BIT(3) + +#define TAS2562_TDM_CFG5_VSNS_EN BIT(6) +#define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) + +#define TAS2562_TDM_CFG6_ISNS_EN BIT(6) +#define TAS2562_TDM_CFG6_ISNS_SLOT_MASK GENMASK(5, 0) + +#endif /* __TAS2562_H__ */ -- 2.20.1

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