[alsa-devel] [PATCH] ASoC: Add HP iPAQ H1940 support
This patch depends on gpiolib support for h1940 latch access, here's link to patch in ml archive:
http://www.spinics.net/lists/arm-kernel/msg98042.html
I hope Ben will find some time to merge it into his tree before 2.6.37 merge window :)
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on SND_S3C24XX_SOC && ARCH_H1940 + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..77550fa --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org + * Copyright (c) 2010 Vasily Khoruzhick anarsoul@gmail.com + * + * Based on version from Arnaud Patard arnaud.patard@rtp-net.org + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + printk(KERN_ERR "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +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(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(codec, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + 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(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL");
On Sun, Sep 12, 2010 at 7:55 PM, Vasily Khoruzhick anarsoul@gmail.com wrote:
This patch depends on gpiolib support for h1940 latch access, here's link to patch in ml archive:
http://www.spinics.net/lists/arm-kernel/msg98042.html
I hope Ben will find some time to merge it into his tree before 2.6.37 merge window :)
Do you want this too to go into changelog ? ;)
В сообщении от 12 сентября 2010 15:14:27 автор Jassi Brar написал:
On Sun, Sep 12, 2010 at 7:55 PM, Vasily Khoruzhick anarsoul@gmail.com
wrote:
This patch depends on gpiolib support for h1940 latch access, here's link to patch in ml archive:
http://www.spinics.net/lists/arm-kernel/msg98042.html
I hope Ben will find some time to merge it into his tree before 2.6.37 merge window :)
Do you want this too to go into changelog ? ;)
Nope, will send v2 with cover soon
This patch depends on gpiolib support for h1940 latch access, here's link to patch in ml archive:
http://www.spinics.net/lists/arm-kernel/msg98042.html
I hope Ben will find some time to merge it into his tree before 2.6.37 merge window :)
v2: - printk replaced via dev_err; - added module alias
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 297 +++++++++++++++++++++++++++++++++++++ 3 files changed, 307 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on SND_S3C24XX_SOC && ARCH_H1940 + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..5dbc0ea --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,297 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org + * Copyright (c) 2010 Vasily Khoruzhick anarsoul@gmail.com + * + * Based on version from Arnaud Patard arnaud.patard@rtp-net.org + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + dev_err(&rtd->dev, "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +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(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(codec, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + 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(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:soc-audio");
Dne Ne 12. září 2010 14:18:47 Vasily Khoruzhick napsal(a):
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
have my: Reviewed-by: Marek Vasut marek.vasut@gmail.com
sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 297 +++++++++++++++++++++++++++++++++++++ 3 files changed, 307 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380
- tristate "Audio support for the HP iPAQ H1940"
- depends on SND_S3C24XX_SOC && ARCH_H1940
- select SND_S3C24XX_SOC_I2S
- select SND_SOC_UDA1380
- help
This driver provides audio support for HP iPAQ h1940 PDA.config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..5dbc0ea --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,297 @@ +/*
- h1940-uda1380.c -- ALSA Soc Audio Layer
- Copyright (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org
- Copyright (c) 2010 Vasily Khoruzhick anarsoul@gmail.com
- Based on version from Arnaud Patard arnaud.patard@rtp-net.org
- This program is free software; you can redistribute it and/or modify
it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version.
- */
+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h>
+#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h>
+#include <plat/regs-iis.h>
+#include <mach/h1940-latch.h>
+#include <asm/mach-types.h>
+#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h"
+static unsigned int rates[] = {
- 11025,
- 22050,
- 44100,
+};
+static struct snd_pcm_hw_constraint_list hw_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
+};
+static struct snd_soc_jack hp_jack;
+static struct snd_soc_jack_pin hp_jack_pins[] = {
- {
.pin = "Headphone Jack",.mask = SND_JACK_HEADPHONE,- },
- {
.pin = "Speaker",.mask = SND_JACK_HEADPHONE,.invert = 1,- },
+};
+static struct snd_soc_jack_gpio hp_jack_gpios[] = {
- {
.gpio = S3C2410_GPG(4),.name = "hp-gpio",.report = SND_JACK_HEADPHONE,.invert = 1,.debounce_time = 200,- },
+};
+static int h1940_startup(struct snd_pcm_substream *substream) +{
- struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.rate_min = hw_rates.list[0];
- runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
- return snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,&hw_rates);+}
+static int h1940_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int div;
- int ret;
- unsigned int rate = params_rate(params);
- switch (rate) {
- case 11025:
- case 22050:
- case 44100:
div = s3c24xx_i2s_get_clockrate() / (384 * rate);if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))div++;break;- default:
dev_err(&rtd->dev, "%s: rate %d is not supported\n",__func__, rate);return -EINVAL;- }
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);- if (ret < 0)
return ret;- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);- if (ret < 0)
return ret;- /* select clock source */
- ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
SND_SOC_CLOCK_OUT);- if (ret < 0)
return ret;- /* set MCLK division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
S3C2410_IISMOD_384FS);- if (ret < 0)
return ret;- /* set BCLK division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
S3C2410_IISMOD_32FS);- if (ret < 0)
return ret;- /* set prescaler division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
S3C24XX_PRESCALE(div, div));- if (ret < 0)
return ret;- return 0;
+}
+static struct snd_soc_ops h1940_ops = {
- .startup = h1940_startup,
- .hw_params = h1940_hw_params,
+};
+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(H1940_LATCH_AUDIO_POWER, 1);- else
gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);- return 0;
+}
+/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
+};
+/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = {
- /* headphone connected to VOUTLHP, VOUTRHP */
- {"Headphone Jack", NULL, "VOUTLHP"},
- {"Headphone Jack", NULL, "VOUTRHP"},
- /* ext speaker connected to VOUTL, VOUTR */
- {"Speaker", NULL, "VOUTL"},
- {"Speaker", NULL, "VOUTR"},
- /* mic is connected to VINM */
- {"VINM", NULL, "Mic Jack"},
+};
+static struct platform_device *s3c24xx_snd_device;
+static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{
- struct snd_soc_codec *codec = rtd->codec;
- int err;
- /* Add h1940 specific widgets */
- err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));- if (err)
return err;- /* Set up h1940 specific audio path audio_mapnects */
- err = snd_soc_dapm_add_routes(codec, audio_map,
ARRAY_SIZE(audio_map));- if (err)
return err;- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Speaker");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_sync(codec);
- snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
&hp_jack);- snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
hp_jack_pins);- snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);- return 0;
+}
+/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = {
- {
.name = "uda1380",.stream_name = "UDA1380 Duplex",.cpu_dai_name = "s3c24xx-iis",.codec_dai_name = "uda1380-hifi",.init = h1940_uda1380_init,.platform_name = "s3c24xx-pcm-audio",.codec_name = "uda1380-codec.0-001a",.ops = &h1940_ops,- },
+};
+static struct snd_soc_card h1940_asoc = {
- .name = "h1940",
- .dai_link = h1940_uda1380_dai,
- .num_links = ARRAY_SIZE(h1940_uda1380_dai),
+};
+static int __init h1940_init(void) +{
- int ret;
- if (!machine_is_h1940())
return -ENODEV;- /* configure some gpios */
- ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
- if (ret)
goto err_out;- ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
- if (ret)
goto err_gpio;- s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
- if (!s3c24xx_snd_device) {
ret = -ENOMEM;goto err_gpio;- }
- 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(H1940_LATCH_AUDIO_POWER);
+err_out:
- return ret;
+}
+static void __exit h1940_exit(void) +{
- platform_device_unregister(s3c24xx_snd_device);
- snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);- gpio_free(H1940_LATCH_AUDIO_POWER);
+}
+module_init(h1940_init); +module_exit(h1940_exit);
+/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:soc-audio");
On Sun, Sep 12, 2010 at 9:18 PM, Vasily Khoruzhick anarsoul@gmail.com wrote:
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 297 +++++++++++++++++++++++++++++++++++++ 3 files changed, 307 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380
- tristate "Audio support for the HP iPAQ H1940"
- depends on SND_S3C24XX_SOC && ARCH_H1940
- select SND_S3C24XX_SOC_I2S
- select SND_SOC_UDA1380
- help
- This driver provides audio support for HP iPAQ h1940 PDA.
config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..5dbc0ea --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,297 @@ +/*
- h1940-uda1380.c -- ALSA Soc Audio Layer
- Copyright (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org
- Copyright (c) 2010 Vasily Khoruzhick anarsoul@gmail.com
- Based on version from Arnaud Patard arnaud.patard@rtp-net.org
- This program is free software; you can redistribute it and/or modify it
- under the terms of the GNU General Public License as published by the
- Free Software Foundation; either version 2 of the License, or (at your
- option) any later version.
- */
+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h>
+#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h>
+#include <plat/regs-iis.h>
+#include <mach/h1940-latch.h>
+#include <asm/mach-types.h>
+#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h"
+static unsigned int rates[] = {
- 11025,
- 22050,
- 44100,
+};
+static struct snd_pcm_hw_constraint_list hw_rates = {
- .count = ARRAY_SIZE(rates),
- .list = rates,
- .mask = 0,
+};
+static struct snd_soc_jack hp_jack;
+static struct snd_soc_jack_pin hp_jack_pins[] = {
- {
- .pin = "Headphone Jack",
- .mask = SND_JACK_HEADPHONE,
- },
- {
- .pin = "Speaker",
- .mask = SND_JACK_HEADPHONE,
- .invert = 1,
- },
+};
+static struct snd_soc_jack_gpio hp_jack_gpios[] = {
- {
- .gpio = S3C2410_GPG(4),
- .name = "hp-gpio",
- .report = SND_JACK_HEADPHONE,
- .invert = 1,
- .debounce_time = 200,
- },
+};
+static int h1940_startup(struct snd_pcm_substream *substream) +{
- struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.rate_min = hw_rates.list[0];
- runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
- return snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &hw_rates);
+}
+static int h1940_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int div;
- int ret;
- unsigned int rate = params_rate(params);
- switch (rate) {
- case 11025:
- case 22050:
- case 44100:
- div = s3c24xx_i2s_get_clockrate() / (384 * rate);
- if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
- div++;
- break;
- default:
- dev_err(&rtd->dev, "%s: rate %d is not supported\n",
- __func__, rate);
- return -EINVAL;
- }
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
- /* select clock source */
- ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
- SND_SOC_CLOCK_OUT);
- if (ret < 0)
- return ret;
- /* set MCLK division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
- S3C2410_IISMOD_384FS);
- if (ret < 0)
- return ret;
- /* set BCLK division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
- S3C2410_IISMOD_32FS);
- if (ret < 0)
- return ret;
- /* set prescaler division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
- S3C24XX_PRESCALE(div, div));
- if (ret < 0)
- return ret;
- return 0;
+}
+static struct snd_soc_ops h1940_ops = {
- .startup = h1940_startup,
- .hw_params = h1940_hw_params,
+};
+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(H1940_LATCH_AUDIO_POWER, 1);
- else
- gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
- return 0;
+}
+/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
+};
+/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = {
- /* headphone connected to VOUTLHP, VOUTRHP */
- {"Headphone Jack", NULL, "VOUTLHP"},
- {"Headphone Jack", NULL, "VOUTRHP"},
- /* ext speaker connected to VOUTL, VOUTR */
- {"Speaker", NULL, "VOUTL"},
- {"Speaker", NULL, "VOUTR"},
- /* mic is connected to VINM */
- {"VINM", NULL, "Mic Jack"},
+};
+static struct platform_device *s3c24xx_snd_device;
+static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{
- struct snd_soc_codec *codec = rtd->codec;
- int err;
- /* Add h1940 specific widgets */
- err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
- ARRAY_SIZE(uda1380_dapm_widgets));
- if (err)
- return err;
- /* Set up h1940 specific audio path audio_mapnects */
- err = snd_soc_dapm_add_routes(codec, audio_map,
- ARRAY_SIZE(audio_map));
- if (err)
- return err;
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Speaker");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_sync(codec);
- snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
- &hp_jack);
- snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
- hp_jack_pins);
- snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
- hp_jack_gpios);
- return 0;
+}
+/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = {
- {
- .name = "uda1380",
- .stream_name = "UDA1380 Duplex",
- .cpu_dai_name = "s3c24xx-iis",
- .codec_dai_name = "uda1380-hifi",
- .init = h1940_uda1380_init,
- .platform_name = "s3c24xx-pcm-audio",
- .codec_name = "uda1380-codec.0-001a",
- .ops = &h1940_ops,
- },
+};
+static struct snd_soc_card h1940_asoc = {
- .name = "h1940",
- .dai_link = h1940_uda1380_dai,
- .num_links = ARRAY_SIZE(h1940_uda1380_dai),
+};
+static int __init h1940_init(void) +{
- int ret;
- if (!machine_is_h1940())
- return -ENODEV;
- /* configure some gpios */
- ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
- if (ret)
- goto err_out;
- ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
- if (ret)
- goto err_gpio;
- s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
- if (!s3c24xx_snd_device) {
- ret = -ENOMEM;
- goto err_gpio;
- }
- 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(H1940_LATCH_AUDIO_POWER);
+err_out:
- return ret;
+}
+static void __exit h1940_exit(void) +{
- platform_device_unregister(s3c24xx_snd_device);
- snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
- hp_jack_gpios);
- gpio_free(H1940_LATCH_AUDIO_POWER);
+}
+module_init(h1940_init); +module_exit(h1940_exit);
+/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:soc-audio");
Is platform:soc-audio specific enough in multi-platform builds of the OS? Though it is unlikely there would ever be such a conflict, but still...
В сообщении от 12 сентября 2010 20:07:41 автор Jassi Brar написал:
Is platform:soc-audio specific enough in multi-platform builds of the OS? Though it is unlikely there would ever be such a conflict, but still...
Other glue drivers just can't load, usually they include machine check in module_init.
Regards Vasily
On Mon, Sep 13, 2010 at 3:09 AM, Vasily Khoruzhick anarsoul@gmail.com wrote:
В сообщении от 12 сентября 2010 20:07:41 автор Jassi Brar написал:
Is platform:soc-audio specific enough in multi-platform builds of the OS? Though it is unlikely there would ever be such a conflict, but still...
Other glue drivers just can't load, usually they include machine check in module_init.
Btw, sound/soc/soc-core.c is the real 'soc-audio' driver module not the MACHINE driver. Turns out, sound/soc/soc-core.c already has the MODULE_ALIAS("platform:soc-audio"); So it seems like an immediate conflict.
On Mon, Sep 13, 2010 at 10:09:16AM +0900, Jassi Brar wrote:
Btw, sound/soc/soc-core.c is the real 'soc-audio' driver module not the MACHINE driver. Turns out, sound/soc/soc-core.c already has the MODULE_ALIAS("platform:soc-audio"); So it seems like an immediate conflict.
Indeed - the MODULE_ALIAS() goes along with the device driver, not with the registration of the device.
В сообщении от 13 сентября 2010 12:05:43 автор Mark Brown написал:
Indeed - the MODULE_ALIAS() goes along with the device driver, not with the registration of the device.
Ok, will resend fixed version soon, however as I said on IRC, please, don't merge this patch until Ben review/merge this patch:
This patch depends on gpiolib support for h1940 latch access, here's link to patch in ml archive:
http://www.spinics.net/lists/arm-kernel/msg98042.html
I hope Ben will find some time to merge it into his tree before 2.6.37 merge window :)
v2: - printk replaced via dev_err; - added module alias
v3: - removed module alias
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on SND_S3C24XX_SOC && ARCH_H1940 + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..8e21c5e --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org + * Copyright (c) 2010 Vasily Khoruzhick anarsoul@gmail.com + * + * Based on version from Arnaud Patard arnaud.patard@rtp-net.org + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + dev_err(&rtd->dev, "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +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(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(codec, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + 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(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL");
On Mon, Sep 13, 2010 at 12:47:18PM +0300, Vasily Khoruzhick wrote:
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
This looks fine - please resubmit when the dependencies have been merged.
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org --- This patch can be merged upstream now, since dependencies were merged into Ben's tree
v2: printk replaced via dev_err, added module alias v3: removed module alias
sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on SND_S3C24XX_SOC && ARCH_H1940 + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..8e21c5e --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard arnaud.patard@rtp-net.org + * Copyright (c) 2010 Vasily Khoruzhick anarsoul@gmail.com + * + * Based on version from Arnaud Patard arnaud.patard@rtp-net.org + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + dev_err(&rtd->dev, "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +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(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(codec, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + 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(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL");
On Mon, Sep 27, 2010 at 10:47:48AM +0300, Vasily Khoruzhick wrote:
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
This patch can be merged upstream now, since dependencies were merged into Ben's tree
We need to wait until Ben's tree has been merged into mainline and back up into ASoC - probably after/during the next merge window - unless Ben has a focused branch with this stuff on it which I can pull.
On 27 of September 2010 18:06:10 Mark Brown wrote:
On Mon, Sep 27, 2010 at 10:47:48AM +0300, Vasily Khoruzhick wrote:
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
This patch can be merged upstream now, since dependencies were merged into Ben's tree
We need to wait until Ben's tree has been merged into mainline and back up into ASoC - probably after/during the next merge window - unless Ben has a focused branch with this stuff on it which I can pull.
Well, then it should be discussed with Ben.
Regards Vasily
On Mon, 27 Sep 2010 08:06:10 -0700 Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Mon, Sep 27, 2010 at 10:47:48AM +0300, Vasily Khoruzhick wrote:
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
This patch can be merged upstream now, since dependencies were merged into Ben's tree
We need to wait until Ben's tree has been merged into mainline and back up into ASoC - probably after/during the next merge window - unless Ben has a focused branch with this stuff on it which I can pull.
Dependencies was merged into linux-next tonight, so I suspect this patch can be merged if you pull from linux-next.
Regards, Vasily
On Fri, Oct 15, 2010 at 10:49:27AM +0300, Vasily khoruzhick wrote:
Dependencies was merged into linux-next tonight, so I suspect this patch can be merged if you pull from linux-next.
-next is an integration tree which should never be merged into other trees. I really need either Linus' tree or a specific branch I can merge into ASoC.
On Fri, 15 Oct 2010 12:37:38 +0100 Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Fri, Oct 15, 2010 at 10:49:27AM +0300, Vasily khoruzhick wrote:
Dependencies was merged into linux-next tonight, so I suspect this patch can be merged if you pull from linux-next.
-next is an integration tree which should never be merged into other trees. I really need either Linus' tree or a specific branch I can merge into ASoC.
It's next-samsung from Ben's tree (git://git.fluff.org/bjdooks/linux.git), I suspect it will be merged into Linus' tree during 2.6.37 merge window, not sure if it's safe/ok to merge it into ASoC right now. Let's wait for 2.6.37 merge window then.
Regards Vasily
On Sun, Sep 12, 2010 at 03:18:47PM +0300, Vasily Khoruzhick wrote:
Signed-off-by: Vasily Khoruzhick anarsoul@gmail.com Tested-by: Arnaud Patard arnaud.patard@rtp-net.org
Aside from the MODULE_ALIAS thing Jassi picked up on this looks good.
participants (4)
-
Jassi Brar -
Marek Vasut -
Mark Brown -
Vasily Khoruzhick