diff options
Diffstat (limited to 'sound/soc/pxa')
| -rw-r--r-- | sound/soc/pxa/Kconfig | 196 | ||||
| -rw-r--r-- | sound/soc/pxa/Makefile | 47 | ||||
| -rw-r--r-- | sound/soc/pxa/brownstone.c | 133 | ||||
| -rw-r--r-- | sound/soc/pxa/corgi.c | 317 | ||||
| -rw-r--r-- | sound/soc/pxa/e740_wm9705.c | 167 | ||||
| -rw-r--r-- | sound/soc/pxa/e750_wm9705.c | 150 | ||||
| -rw-r--r-- | sound/soc/pxa/e800_wm9712.c | 150 | ||||
| -rw-r--r-- | sound/soc/pxa/em-x270.c | 92 | ||||
| -rw-r--r-- | sound/soc/pxa/hx4700.c | 214 | ||||
| -rw-r--r-- | sound/soc/pxa/imote2.c | 99 | ||||
| -rw-r--r-- | sound/soc/pxa/magician.c | 433 | ||||
| -rw-r--r-- | sound/soc/pxa/mioa701_wm9713.c | 201 | ||||
| -rw-r--r-- | sound/soc/pxa/mmp-pcm.c | 267 | ||||
| -rw-r--r-- | sound/soc/pxa/mmp-sspa.c | 29 | ||||
| -rw-r--r-- | sound/soc/pxa/palm27x.c | 161 | ||||
| -rw-r--r-- | sound/soc/pxa/poodle.c | 288 | ||||
| -rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 82 | ||||
| -rw-r--r-- | sound/soc/pxa/pxa2xx-ac97.c | 36 | ||||
| -rw-r--r-- | sound/soc/pxa/pxa2xx-i2s.c | 145 | ||||
| -rw-r--r-- | sound/soc/pxa/pxa2xx-pcm.c | 3 | ||||
| -rw-r--r-- | sound/soc/pxa/spitz.c | 74 | ||||
| -rw-r--r-- | sound/soc/pxa/tosa.c | 262 | ||||
| -rw-r--r-- | sound/soc/pxa/ttc-dkb.c | 141 | ||||
| -rw-r--r-- | sound/soc/pxa/z2.c | 219 | ||||
| -rw-r--r-- | sound/soc/pxa/zylonite.c | 266 |
25 files changed, 178 insertions, 3994 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 0ac85eada75c..e026f9912a6d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,4 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only +menu "PXA" + config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA || COMPILE_TEST @@ -8,15 +10,11 @@ config SND_PXA2XX_SOC the PXA2xx AC97, I2S or SSP interface. You will also need to select the audio interfaces to support below. -config SND_MMP_SOC - bool - select MMP_SRAM - -config SND_PXA2XX_AC97 - tristate - config SND_PXA2XX_SOC_AC97 - tristate + tristate "SoC AC97 support for PXA2xx" + depends on SND_PXA2XX_SOC + depends on AC97_BUS=n + default y select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 @@ -36,20 +34,10 @@ config SND_MMP_SOC_SSPA tristate "SoC Audio via MMP SSPA ports" depends on ARCH_MMP select SND_SOC_GENERIC_DMAENGINE_PCM - select SND_ARM help Say Y if you want to add support for codecs attached to the MMP SSPA interface. -config SND_PXA2XX_SOC_CORGI - tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C - select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on Sharp - Zaurus SL-C7x0 models (Corgi, Shepherd, Husky). - config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C @@ -59,101 +47,6 @@ config SND_PXA2XX_SOC_SPITZ Say Y if you want to add support for SoC audio on Sharp Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita). -config SND_PXA2XX_SOC_Z2 - tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C - select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8750 - help - Say Y if you want to add support for SoC audio on Zipit Z2. - -config SND_PXA2XX_SOC_POODLE - tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE && I2C - select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on Sharp - Zaurus SL-5600 model (Poodle). - -config SND_PXA2XX_SOC_TOSA - tristate "SoC AC97 Audio support for Tosa" - depends on SND_PXA2XX_SOC && MACH_TOSA - depends on MFD_TC6393XB - depends on AC97_BUS=n - select REGMAP - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9712 - help - Say Y if you want to add support for SoC audio on Sharp - Zaurus SL-C6000x models (Tosa). - -config SND_PXA2XX_SOC_E740 - tristate "SoC AC97 Audio support for e740" - depends on SND_PXA2XX_SOC && MACH_E740 - depends on AC97_BUS=n - select REGMAP - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_SOC_WM9705 - select SND_PXA2XX_SOC_AC97 - help - Say Y if you want to add support for SoC audio on the - toshiba e740 PDA - -config SND_PXA2XX_SOC_E750 - tristate "SoC AC97 Audio support for e750" - depends on SND_PXA2XX_SOC && MACH_E750 - depends on AC97_BUS=n - select REGMAP - select SND_SOC_WM9705 - select SND_PXA2XX_SOC_AC97 - help - Say Y if you want to add support for SoC audio on the - toshiba e750 PDA - -config SND_PXA2XX_SOC_E800 - tristate "SoC AC97 Audio support for e800" - depends on SND_PXA2XX_SOC && MACH_E800 - depends on AC97_BUS=n - select REGMAP - select SND_SOC_WM9712 - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_PXA2XX_SOC_AC97 - help - Say Y if you want to add support for SoC audio on the - Toshiba e800 PDA - -config SND_PXA2XX_SOC_EM_X270 - tristate "SoC Audio support for CompuLab CM-X300" - depends on SND_PXA2XX_SOC && MACH_CM_X300 - depends on AC97_BUS=n - select REGMAP - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9712 - help - Say Y if you want to add support for SoC audio on - CompuLab EM-x270, eXeda and CM-X300 machines. - -config SND_PXA2XX_SOC_PALM27X - bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" - depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ - MACH_PALMT5 || MACH_PALMTE2) - depends on AC97_BUS=n - select REGMAP - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9712 - help - Say Y if you want to add support for SoC audio on - Palm T|X, T5, E2 or LifeDrive handheld computer. - config SND_PXA910_SOC tristate "SoC Audio for Marvell PXA910 chip" depends on ARCH_MMP && SND @@ -162,79 +55,4 @@ config SND_PXA910_SOC Say Y if you want to add support for SoC audio on the Marvell PXA910 reference platform. -config SND_SOC_TTC_DKB - tristate "SoC Audio support for TTC DKB" - depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y - select PXA_SSP - select SND_PXA_SOC_SSP - select SND_MMP_SOC - select MFD_88PM860X - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on TTC DKB - - -config SND_SOC_ZYLONITE - tristate "SoC Audio support for Marvell Zylonite" - depends on SND_PXA2XX_SOC && MACH_ZYLONITE - depends on AC97_BUS=n - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_PXA2XX_SOC_AC97 - select REGMAP - select SND_PXA_SOC_SSP - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - Marvell Zylonite reference platform. - -config SND_PXA2XX_SOC_HX4700 - tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 && I2C - select SND_PXA2XX_SOC_I2S - select SND_SOC_AK4641 - help - Say Y if you want to add support for SoC audio on the - HP iPAQ hx4700. - -config SND_PXA2XX_SOC_MAGICIAN - tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C - select SND_PXA2XX_SOC_I2S - select SND_PXA_SOC_SSP - select SND_SOC_UDA1380 - help - Say Y if you want to add support for SoC audio on the - HTC Magician. - -config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 - depends on AC97_BUS=n - select REGMAP - select AC97_BUS_NEW - select AC97_BUS_COMPAT - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. - -config SND_PXA2XX_SOC_IMOTE2 - tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C - select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8940 - help - Say Y if you want to add support for SoC audio on the - IMote 2. - -config SND_MMP_SOC_BROWNSTONE - tristate "SoC Audio support for Marvell Brownstone" - depends on SND_MMP_SOC_SSPA && MACH_BROWNSTONE && I2C - select SND_MMP_SOC - select MFD_WM8994 - select SND_SOC_WM8994 - help - Say Y if you want to add support for SoC audio on the - Marvell Brownstone reference platform. +endmenu diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index ea4929d73318..93b4e57eaa5c 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -1,52 +1,17 @@ # SPDX-License-Identifier: GPL-2.0 # PXA Platform Support -snd-soc-pxa2xx-objs := pxa2xx-pcm.o -snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o -snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o -snd-soc-pxa-ssp-objs := pxa-ssp.o -snd-soc-mmp-objs := mmp-pcm.o -snd-soc-mmp-sspa-objs := mmp-sspa.o +snd-soc-pxa2xx-y := pxa2xx-pcm.o +snd-soc-pxa2xx-ac97-y := pxa2xx-ac97.o +snd-soc-pxa2xx-i2s-y := pxa2xx-i2s.o +snd-soc-pxa-ssp-y := pxa-ssp.o +snd-soc-mmp-sspa-y := mmp-sspa.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o -obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o # PXA Machine Support -snd-soc-corgi-objs := corgi.o -snd-soc-poodle-objs := poodle.o -snd-soc-tosa-objs := tosa.o -snd-soc-e740-objs := e740_wm9705.o -snd-soc-e750-objs := e750_wm9705.o -snd-soc-e800-objs := e800_wm9712.o -snd-soc-spitz-objs := spitz.o -snd-soc-em-x270-objs := em-x270.o -snd-soc-palm27x-objs := palm27x.o -snd-soc-zylonite-objs := zylonite.o -snd-soc-hx4700-objs := hx4700.o -snd-soc-magician-objs := magician.o -snd-soc-mioa701-objs := mioa701_wm9713.o -snd-soc-z2-objs := z2.o -snd-soc-imote2-objs := imote2.o -snd-soc-brownstone-objs := brownstone.o -snd-soc-ttc-dkb-objs := ttc-dkb.o - -obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o -obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o -obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o -obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o -obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o -obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o +snd-soc-spitz-y := spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o -obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o -obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o -obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o -obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o -obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o -obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o -obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o -obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o -obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o -obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c deleted file mode 100644 index f310a8e91bbf..000000000000 --- a/sound/soc/pxa/brownstone.c +++ /dev/null @@ -1,133 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * linux/sound/soc/pxa/brownstone.c - * - * Copyright (C) 2011 Marvell International Ltd. - */ - -#include <linux/module.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include "../codecs/wm8994.h" -#include "mmp-sspa.h" - -static const struct snd_kcontrol_new brownstone_dapm_control[] = { - SOC_DAPM_PIN_SWITCH("Ext Spk"), -}; - -static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Main Mic", NULL), -}; - -static const struct snd_soc_dapm_route brownstone_audio_map[] = { - {"Ext Spk", NULL, "SPKOUTLP"}, - {"Ext Spk", NULL, "SPKOUTLN"}, - {"Ext Spk", NULL, "SPKOUTRP"}, - {"Ext Spk", NULL, "SPKOUTRN"}, - - {"Headset Stereophone", NULL, "HPOUT1L"}, - {"Headset Stereophone", NULL, "HPOUT1R"}, - - {"IN1RN", NULL, "Headset Mic"}, - - {"DMIC1DAT", NULL, "MICBIAS1"}, - {"MICBIAS1", NULL, "Main Mic"}, -}; - -static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int freq_out, sspa_mclk, sysclk; - - if (params_rate(params) > 11025) { - freq_out = params_rate(params) * 512; - sysclk = params_rate(params) * 256; - sspa_mclk = params_rate(params) * 64; - } else { - freq_out = params_rate(params) * 1024; - sysclk = params_rate(params) * 512; - sspa_mclk = params_rate(params) * 64; - } - - snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); - snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); - snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk); - - /* set wm8994 sysclk */ - snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0); - - return 0; -} - -/* machine stream operations */ -static const struct snd_soc_ops brownstone_ops = { - .hw_params = brownstone_wm8994_hw_params, -}; - -SND_SOC_DAILINK_DEFS(wm8994, - DAILINK_COMP_ARRAY(COMP_CPU("mmp-sspa-dai.0")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8994-codec", "wm8994-aif1")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("mmp-pcm-audio"))); - -static struct snd_soc_dai_link brownstone_wm8994_dai[] = { -{ - .name = "WM8994", - .stream_name = "WM8994 HiFi", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &brownstone_ops, - SND_SOC_DAILINK_REG(wm8994), -}, -}; - -/* audio machine driver */ -static struct snd_soc_card brownstone = { - .name = "brownstone", - .owner = THIS_MODULE, - .dai_link = brownstone_wm8994_dai, - .num_links = ARRAY_SIZE(brownstone_wm8994_dai), - - .controls = brownstone_dapm_control, - .num_controls = ARRAY_SIZE(brownstone_dapm_control), - .dapm_widgets = brownstone_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), - .dapm_routes = brownstone_audio_map, - .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), - .fully_routed = true, -}; - -static int brownstone_probe(struct platform_device *pdev) -{ - int ret; - - brownstone.dev = &pdev->dev; - ret = devm_snd_soc_register_card(&pdev->dev, &brownstone); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static struct platform_driver mmp_driver = { - .driver = { - .name = "brownstone-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = brownstone_probe, -}; - -module_platform_driver(mmp_driver); - -MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC Brownstone"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:brownstone-audio"); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c deleted file mode 100644 index 8ee2dea25a8d..000000000000 --- a/sound/soc/pxa/corgi.c +++ /dev/null @@ -1,317 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * corgi.c -- SoC audio for Corgi - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <lrg@slimlogic.co.uk> - * Richard Purdie <richard@openedhand.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/i2c.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/corgi.h> -#include <mach/audio.h> - -#include "../codecs/wm8731.h" -#include "pxa2xx-i2s.h" - -#define CORGI_HP 0 -#define CORGI_MIC 1 -#define CORGI_LINE 2 -#define CORGI_HEADSET 3 -#define CORGI_HP_OFF 4 -#define CORGI_SPK_ON 0 -#define CORGI_SPK_OFF 1 - - /* audio clock in Hz - rounded from 12.235MHz */ -#define CORGI_AUDIO_CLOCK 12288000 - -static int corgi_jack_func; -static int corgi_spk_func; - -static void corgi_ext_control(struct snd_soc_dapm_context *dapm) -{ - snd_soc_dapm_mutex_lock(dapm); - - /* set up jack connection */ - switch (corgi_jack_func) { - case CORGI_HP: - /* set = unmute headphone */ - gpio_set_value(CORGI_GPIO_MUTE_L, 1); - gpio_set_value(CORGI_GPIO_MUTE_R, 1); - snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); - break; - case CORGI_MIC: - /* reset = mute headphone */ - gpio_set_value(CORGI_GPIO_MUTE_L, 0); - gpio_set_value(CORGI_GPIO_MUTE_R, 0); - snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); - break; - case CORGI_LINE: - gpio_set_value(CORGI_GPIO_MUTE_L, 0); - gpio_set_value(CORGI_GPIO_MUTE_R, 0); - snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); - break; - case CORGI_HEADSET: - gpio_set_value(CORGI_GPIO_MUTE_L, 0); - gpio_set_value(CORGI_GPIO_MUTE_R, 1); - snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - break; - } - - if (corgi_spk_func == CORGI_SPK_ON) - snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk"); - else - snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk"); - - /* signal a DAPM event */ - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); -} - -static int corgi_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - - /* check the jack status at stream startup */ - corgi_ext_control(&rtd->card->dapm); - - return 0; -} - -/* we need to unmute the HP at shutdown as the mute burns power on corgi */ -static void corgi_shutdown(struct snd_pcm_substream *substream) -{ - /* set = unmute headphone */ - gpio_set_value(CORGI_GPIO_MUTE_L, 1); - gpio_set_value(CORGI_GPIO_MUTE_R, 1); -} - -static int corgi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set the I2S system clock as input (unused) */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops corgi_ops = { - .startup = corgi_startup, - .hw_params = corgi_hw_params, - .shutdown = corgi_shutdown, -}; - -static int corgi_get_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = corgi_jack_func; - return 0; -} - -static int corgi_set_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (corgi_jack_func == ucontrol->value.enumerated.item[0]) - return 0; - - corgi_jack_func = ucontrol->value.enumerated.item[0]; - corgi_ext_control(&card->dapm); - return 1; -} - -static int corgi_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = corgi_spk_func; - return 0; -} - -static int corgi_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (corgi_spk_func == ucontrol->value.enumerated.item[0]) - return 0; - - corgi_spk_func = ucontrol->value.enumerated.item[0]; - corgi_ext_control(&card->dapm); - return 1; -} - -static int corgi_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -static int corgi_mic_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -/* corgi machine dapm widgets */ -static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { -SND_SOC_DAPM_HP("Headphone Jack", NULL), -SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event), -SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event), -SND_SOC_DAPM_LINE("Line Jack", NULL), -SND_SOC_DAPM_HP("Headset Jack", NULL), -}; - -/* Corgi machine audio map (connections to the codec pins) */ -static const struct snd_soc_dapm_route corgi_audio_map[] = { - - /* headset Jack - in = micin, out = LHPOUT*/ - {"Headset Jack", NULL, "LHPOUT"}, - - /* headphone connected to LHPOUT1, RHPOUT1 */ - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - /* speaker connected to LOUT, ROUT */ - {"Ext Spk", NULL, "ROUT"}, - {"Ext Spk", NULL, "LOUT"}, - - /* mic is connected to MICIN (via right channel of headphone jack) */ - {"MICIN", NULL, "Mic Jack"}, - - /* Same as the above but no mic bias for line signals */ - {"MICIN", NULL, "Line Jack"}, -}; - -static const char * const jack_function[] = {"Headphone", "Mic", "Line", - "Headset", "Off"}; -static const char * const spk_function[] = {"On", "Off"}; -static const struct soc_enum corgi_enum[] = { - SOC_ENUM_SINGLE_EXT(5, jack_function), - SOC_ENUM_SINGLE_EXT(2, spk_function), -}; - -static const struct snd_kcontrol_new wm8731_corgi_controls[] = { - SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack, - corgi_set_jack), - SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk, - corgi_set_spk), -}; - -/* corgi digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(wm8731, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link corgi_dai = { - .name = "WM8731", - .stream_name = "WM8731", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &corgi_ops, - SND_SOC_DAILINK_REG(wm8731), -}; - -/* corgi audio machine driver */ -static struct snd_soc_card corgi = { - .name = "Corgi", - .owner = THIS_MODULE, - .dai_link = &corgi_dai, - .num_links = 1, - - .controls = wm8731_corgi_controls, - .num_controls = ARRAY_SIZE(wm8731_corgi_controls), - .dapm_widgets = wm8731_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), - .dapm_routes = corgi_audio_map, - .num_dapm_routes = ARRAY_SIZE(corgi_audio_map), - .fully_routed = true, -}; - -static int corgi_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &corgi; - int ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static struct platform_driver corgi_driver = { - .driver = { - .name = "corgi-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = corgi_probe, -}; - -module_platform_driver(corgi_driver); - -/* Module information */ -MODULE_AUTHOR("Richard Purdie"); -MODULE_DESCRIPTION("ALSA SoC Corgi"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:corgi-audio"); diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c deleted file mode 100644 index eafa1482afbe..000000000000 --- a/sound/soc/pxa/e740_wm9705.c +++ /dev/null @@ -1,167 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * e740-wm9705.c -- SoC audio for e740 - * - * Copyright 2007 (c) Ian Molton <spyro@f2s.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <mach/audio.h> -#include <mach/eseries-gpio.h> - -#include <asm/mach-types.h> - -#define E740_AUDIO_OUT 1 -#define E740_AUDIO_IN 2 - -static int e740_audio_power; - -static void e740_sync_audio_power(int status) -{ - gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status); - gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0); - gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0); -} - -static int e740_mic_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (event & SND_SOC_DAPM_PRE_PMU) - e740_audio_power |= E740_AUDIO_IN; - else if (event & SND_SOC_DAPM_POST_PMD) - e740_audio_power &= ~E740_AUDIO_IN; - - e740_sync_audio_power(e740_audio_power); - - return 0; -} - -static int e740_output_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (event & SND_SOC_DAPM_PRE_PMU) - e740_audio_power |= E740_AUDIO_OUT; - else if (event & SND_SOC_DAPM_POST_PMD) - e740_audio_power &= ~E740_AUDIO_OUT; - - e740_sync_audio_power(e740_audio_power); - - return 0; -} - -static const struct snd_soc_dapm_widget e740_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic (Internal)", NULL), - SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0, - e740_output_amp_event, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0, - e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"Output Amp", NULL, "LOUT"}, - {"Output Amp", NULL, "ROUT"}, - {"Output Amp", NULL, "MONOOUT"}, - - {"Speaker", NULL, "Output Amp"}, - {"Headphone Jack", NULL, "Output Amp"}, - - {"MIC1", NULL, "Mic Amp"}, - {"Mic Amp", NULL, "Mic (Internal)"}, -}; - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link e740_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - SND_SOC_DAILINK_REG(ac97), - }, - { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - SND_SOC_DAILINK_REG(ac97_aux), - }, -}; - -static struct snd_soc_card e740 = { - .name = "Toshiba e740", - .owner = THIS_MODULE, - .dai_link = e740_dai, - .num_links = ARRAY_SIZE(e740_dai), - - .dapm_widgets = e740_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static struct gpio e740_audio_gpios[] = { - { GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp" }, - { GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, "Output amp" }, - { GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, "Audio power" }, -}; - -static int e740_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &e740; - int ret; - - ret = gpio_request_array(e740_audio_gpios, - ARRAY_SIZE(e740_audio_gpios)); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); - } - return ret; -} - -static int e740_remove(struct platform_device *pdev) -{ - gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); - return 0; -} - -static struct platform_driver e740_driver = { - .driver = { - .name = "e740-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = e740_probe, - .remove = e740_remove, -}; - -module_platform_driver(e740_driver); - -/* Module information */ -MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); -MODULE_DESCRIPTION("ALSA SoC driver for e740"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:e740-audio"); diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c deleted file mode 100644 index d75510d7b16b..000000000000 --- a/sound/soc/pxa/e750_wm9705.c +++ /dev/null @@ -1,150 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * e750-wm9705.c -- SoC audio for e750 - * - * Copyright 2007 (c) Ian Molton <spyro@f2s.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <mach/audio.h> -#include <mach/eseries-gpio.h> - -#include <asm/mach-types.h> - -static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (event & SND_SOC_DAPM_PRE_PMU) - gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0); - else if (event & SND_SOC_DAPM_POST_PMD) - gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1); - - return 0; -} - -static int e750_hp_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (event & SND_SOC_DAPM_PRE_PMU) - gpio_set_value(GPIO_E750_HP_AMP_OFF, 0); - else if (event & SND_SOC_DAPM_POST_PMD) - gpio_set_value(GPIO_E750_HP_AMP_OFF, 1); - - return 0; -} - -static const struct snd_soc_dapm_widget e750_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic (Internal)", NULL), - SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, - e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, - e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"Headphone Amp", NULL, "HPOUTL"}, - {"Headphone Amp", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "Headphone Amp"}, - - {"Speaker Amp", NULL, "MONOOUT"}, - {"Speaker", NULL, "Speaker Amp"}, - - {"MIC1", NULL, "Mic (Internal)"}, -}; - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link e750_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - SND_SOC_DAILINK_REG(ac97), - /* use ops to check startup state */ - }, - { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - SND_SOC_DAILINK_REG(ac97_aux), - }, -}; - -static struct snd_soc_card e750 = { - .name = "Toshiba e750", - .owner = THIS_MODULE, - .dai_link = e750_dai, - .num_links = ARRAY_SIZE(e750_dai), - - .dapm_widgets = e750_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static struct gpio e750_audio_gpios[] = { - { GPIO_E750_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Headphone amp" }, - { GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" }, -}; - -static int e750_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &e750; - int ret; - - ret = gpio_request_array(e750_audio_gpios, - ARRAY_SIZE(e750_audio_gpios)); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); - } - return ret; -} - -static int e750_remove(struct platform_device *pdev) -{ - gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); - return 0; -} - -static struct platform_driver e750_driver = { - .driver = { - .name = "e750-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = e750_probe, - .remove = e750_remove, -}; - -module_platform_driver(e750_driver); - -/* Module information */ -MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); -MODULE_DESCRIPTION("ALSA SoC driver for e750"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:e750-audio"); diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c deleted file mode 100644 index 56d543da938a..000000000000 --- a/sound/soc/pxa/e800_wm9712.c +++ /dev/null @@ -1,150 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * e800-wm9712.c -- SoC audio for e800 - * - * Copyright 2007 (c) Ian Molton <spyro@f2s.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/audio.h> -#include <mach/eseries-gpio.h> - -static int e800_spk_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (event & SND_SOC_DAPM_PRE_PMU) - gpio_set_value(GPIO_E800_SPK_AMP_ON, 1); - else if (event & SND_SOC_DAPM_POST_PMD) - gpio_set_value(GPIO_E800_SPK_AMP_ON, 0); - - return 0; -} - -static int e800_hp_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (event & SND_SOC_DAPM_PRE_PMU) - gpio_set_value(GPIO_E800_HP_AMP_OFF, 0); - else if (event & SND_SOC_DAPM_POST_PMD) - gpio_set_value(GPIO_E800_HP_AMP_OFF, 1); - - return 0; -} - -static const struct snd_soc_dapm_widget e800_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic (Internal1)", NULL), - SND_SOC_DAPM_MIC("Mic (Internal2)", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, - e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, - e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"Headphone Jack", NULL, "HPOUTL"}, - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "Headphone Amp"}, - - {"Speaker Amp", NULL, "MONOOUT"}, - {"Speaker", NULL, "Speaker Amp"}, - - {"MIC1", NULL, "Mic (Internal1)"}, - {"MIC2", NULL, "Mic (Internal2)"}, -}; - - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link e800_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - SND_SOC_DAILINK_REG(ac97), - }, - { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - SND_SOC_DAILINK_REG(ac97_aux), - }, -}; - -static struct snd_soc_card e800 = { - .name = "Toshiba e800", - .owner = THIS_MODULE, - .dai_link = e800_dai, - .num_links = ARRAY_SIZE(e800_dai), - - .dapm_widgets = e800_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct gpio e800_audio_gpios[] = { - { GPIO_E800_SPK_AMP_ON, GPIOF_OUT_INIT_HIGH, "Headphone amp" }, - { GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" }, -}; - -static int e800_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &e800; - int ret; - - ret = gpio_request_array(e800_audio_gpios, - ARRAY_SIZE(e800_audio_gpios)); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); - } - return ret; -} - -static int e800_remove(struct platform_device *pdev) -{ - gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); - return 0; -} - -static struct platform_driver e800_driver = { - .driver = { - .name = "e800-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = e800_probe, - .remove = e800_remove, -}; - -module_platform_driver(e800_driver); - -/* Module information */ -MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); -MODULE_DESCRIPTION("ALSA SoC driver for e800"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:e800-audio"); diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c deleted file mode 100644 index 9076ea7e9339..000000000000 --- a/sound/soc/pxa/em-x270.c +++ /dev/null @@ -1,92 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * SoC audio driver for EM-X270, eXeda and CM-X300 - * - * Copyright 2007, 2009 CompuLab, Ltd. - * - * Author: Mike Rapoport <mike@compulab.co.il> - * - * Copied from tosa.c: - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <lrg@slimlogic.co.uk> - * Richard Purdie <richard@openedhand.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/audio.h> - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link em_x270_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - SND_SOC_DAILINK_REG(ac97), - }, - { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - SND_SOC_DAILINK_REG(ac97_aux), - }, -}; - -static struct snd_soc_card em_x270 = { - .name = "EM-X270", - .owner = THIS_MODULE, - .dai_link = em_x270_dai, - .num_links = ARRAY_SIZE(em_x270_dai), -}; - -static struct platform_device *em_x270_snd_device; - -static int __init em_x270_init(void) -{ - int ret; - - if (!(machine_is_em_x270() || machine_is_exeda() - || machine_is_cm_x300())) - return -ENODEV; - - em_x270_snd_device = platform_device_alloc("soc-audio", -1); - if (!em_x270_snd_device) - return -ENOMEM; - - platform_set_drvdata(em_x270_snd_device, &em_x270); - ret = platform_device_add(em_x270_snd_device); - - if (ret) - platform_device_put(em_x270_snd_device); - - return ret; -} - -static void __exit em_x270_exit(void) -{ - platform_device_unregister(em_x270_snd_device); -} - -module_init(em_x270_init); -module_exit(em_x270_exit); - -/* Module information */ -MODULE_AUTHOR("Mike Rapoport"); -MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c deleted file mode 100644 index 7334fac758de..000000000000 --- a/sound/soc/pxa/hx4700.c +++ /dev/null @@ -1,214 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * SoC audio for HP iPAQ hx4700 - * - * Copyright (c) 2009 Philipp Zabel - */ - -#include <linux/module.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/delay.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/jack.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hx4700.h> -#include <asm/mach-types.h> -#include "pxa2xx-i2s.h" - -static struct snd_soc_jack hs_jack; - -/* Headphones jack detection DAPM pin */ -static struct snd_soc_jack_pin hs_jack_pin[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Speaker", - /* disable speaker when hp jack is inserted */ - .mask = SND_JACK_HEADPHONE, - .invert = 1, - }, -}; - -/* Headphones jack detection GPIO */ -static struct snd_soc_jack_gpio hs_jack_gpio = { - .gpio = GPIO75_HX4700_EARPHONE_nDET, - .invert = true, - .name = "hp-gpio", - .report = SND_JACK_HEADPHONE, - .debounce_time = 200, -}; - -/* - * iPAQ hx4700 uses I2S for capture and playback. - */ -static int hx4700_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int ret = 0; - - /* set the I2S system clock as output */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - /* inform codec driver about clock freq * - * (PXA I2S always uses divider 256) */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params), - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops hx4700_ops = { - .hw_params = hx4700_hw_params, -}; - -static int hx4700_spk_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -static int hx4700_hp_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -/* hx4700 machine dapm widgets */ -static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power), - SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power), - SND_SOC_DAPM_MIC("Built-in Microphone", NULL), -}; - -/* hx4700 machine audio_map */ -static const struct snd_soc_dapm_route hx4700_audio_map[] = { - - /* Headphone connected to LOUT, ROUT */ - {"Headphone Jack", NULL, "LOUT"}, - {"Headphone Jack", NULL, "ROUT"}, - - /* Speaker connected to MOUT2 */ - {"Speaker", NULL, "MOUT2"}, - - /* Microphone connected to MICIN */ - {"MICIN", NULL, "Built-in Microphone"}, - {"AIN", NULL, "MICOUT"}, -}; - -/* - * Logic for a ak4641 as connected on a HP iPAQ hx4700 - */ -static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) -{ - int err; - - /* Jack detection API stuff */ - err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin, - ARRAY_SIZE(hs_jack_pin)); - if (err) - return err; - - err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio); - - return err; -} - -/* hx4700 digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(ak4641, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("ak4641.0-0012", "ak4641-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link hx4700_dai = { - .name = "ak4641", - .stream_name = "AK4641", - .init = hx4700_ak4641_init, - .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &hx4700_ops, - SND_SOC_DAILINK_REG(ak4641), -}; - -/* hx4700 audio machine driver */ -static struct snd_soc_card snd_soc_card_hx4700 = { - .name = "iPAQ hx4700", - .owner = THIS_MODULE, - .dai_link = &hx4700_dai, - .num_links = 1, - .dapm_widgets = hx4700_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets), - .dapm_routes = hx4700_audio_map, - .num_dapm_routes = ARRAY_SIZE(hx4700_audio_map), - .fully_routed = true, -}; - -static struct gpio hx4700_audio_gpios[] = { - { GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" }, - { GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" }, -}; - -static int hx4700_audio_probe(struct platform_device *pdev) -{ - int ret; - - if (!machine_is_h4700()) - return -ENODEV; - - ret = gpio_request_array(hx4700_audio_gpios, - ARRAY_SIZE(hx4700_audio_gpios)); - if (ret) - return ret; - - snd_soc_card_hx4700.dev = &pdev->dev; - ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700); - if (ret) - gpio_free_array(hx4700_audio_gpios, - ARRAY_SIZE(hx4700_audio_gpios)); - - return ret; -} - -static int hx4700_audio_remove(struct platform_device *pdev) -{ - gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); - gpio_set_value(GPIO107_HX4700_SPK_nSD, 0); - - gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios)); - return 0; -} - -static struct platform_driver hx4700_audio_driver = { - .driver = { - .name = "hx4700-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = hx4700_audio_probe, - .remove = hx4700_audio_remove, -}; - -module_platform_driver(hx4700_audio_driver); - -MODULE_AUTHOR("Philipp Zabel"); -MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:hx4700-audio"); diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c deleted file mode 100644 index a575676508b3..000000000000 --- a/sound/soc/pxa/imote2.c +++ /dev/null @@ -1,99 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only - -#include <linux/module.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "../codecs/wm8940.h" -#include "pxa2xx-i2s.h" - -static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - unsigned int clk = 0; - int ret; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set the I2S system clock as input (unused) */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk, - SND_SOC_CLOCK_OUT); - - return ret; -} - -static const struct snd_soc_ops imote2_asoc_ops = { - .hw_params = imote2_asoc_hw_params, -}; - -SND_SOC_DAILINK_DEFS(wm8940, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8940-codec.0-0034", - "wm8940-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link imote2_dai = { - .name = "WM8940", - .stream_name = "WM8940", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &imote2_asoc_ops, - SND_SOC_DAILINK_REG(wm8940), -}; - -static struct snd_soc_card imote2 = { - .name = "Imote2", - .owner = THIS_MODULE, - .dai_link = &imote2_dai, - .num_links = 1, -}; - -static int imote2_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &imote2; - int ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static struct platform_driver imote2_driver = { - .driver = { - .name = "imote2-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = imote2_probe, -}; - -module_platform_driver(imote2_driver); - -MODULE_AUTHOR("Jonathan Cameron"); -MODULE_DESCRIPTION("ALSA SoC Imote 2"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:imote2-audio"); diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c deleted file mode 100644 index a5f326c97af2..000000000000 --- a/sound/soc/pxa/magician.c +++ /dev/null @@ -1,433 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * SoC audio for HTC Magician - * - * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com> - * - * based on spitz.c, - * Authors: Liam Girdwood <lrg@slimlogic.co.uk> - * Richard Purdie <richard@openedhand.com> - */ - -#include <linux/module.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <linux/i2c.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/uda1380.h> - -#include <mach/magician.h> -#include <asm/mach-types.h> -#include "../codecs/uda1380.h" -#include "pxa2xx-i2s.h" -#include "pxa-ssp.h" - -#define MAGICIAN_MIC 0 -#define MAGICIAN_MIC_EXT 1 - -static int magician_hp_switch; -static int magician_spk_switch = 1; -static int magician_in_sel = MAGICIAN_MIC; - -static void magician_ext_control(struct snd_soc_dapm_context *dapm) -{ - - snd_soc_dapm_mutex_lock(dapm); - - if (magician_spk_switch) - snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker"); - else - snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker"); - if (magician_hp_switch) - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - else - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); - - switch (magician_in_sel) { - case MAGICIAN_MIC: - snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic"); - break; - case MAGICIAN_MIC_EXT: - snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic"); - break; - } - - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); -} - -static int magician_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - - /* check the jack status at stream startup */ - magician_ext_control(&rtd->card->dapm); - - return 0; -} - -/* - * Magician uses SSP port for playback. - */ -static int magician_playback_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - unsigned int width; - int ret = 0; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | - 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_DSP_A | - SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - width = snd_pcm_format_physical_width(params_format(params)); - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); - if (ret < 0) - return ret; - - /* set audio clock as clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -/* - * Magician uses I2S for capture. - */ -static int magician_capture_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int ret = 0; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_MSB | 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_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the I2S system clock as output */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops magician_capture_ops = { - .startup = magician_startup, - .hw_params = magician_capture_hw_params, -}; - -static const struct snd_soc_ops magician_playback_ops = { - .startup = magician_startup, - .hw_params = magician_playback_hw_params, -}; - -static int magician_get_hp(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = magician_hp_switch; - return 0; -} - -static int magician_set_hp(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (magician_hp_switch == ucontrol->value.integer.value[0]) - return 0; - - magician_hp_switch = ucontrol->value.integer.value[0]; - magician_ext_control(&card->dapm); - return 1; -} - -static int magician_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = magician_spk_switch; - return 0; -} - -static int magician_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (magician_spk_switch == ucontrol->value.integer.value[0]) - return 0; - - magician_spk_switch = ucontrol->value.integer.value[0]; - magician_ext_control(&card->dapm); - return 1; -} - -static int magician_get_input(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = magician_in_sel; - return 0; -} - -static int magician_set_input(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - if (magician_in_sel == ucontrol->value.enumerated.item[0]) - return 0; - - magician_in_sel = ucontrol->value.enumerated.item[0]; - - switch (magician_in_sel) { - case MAGICIAN_MIC: - gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1); - break; - case MAGICIAN_MIC_EXT: - gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0); - } - - return 1; -} - -static int magician_spk_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -static int magician_hp_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -static int magician_mic_bias(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event)); - return 0; -} - -/* magician machine dapm widgets */ -static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power), - SND_SOC_DAPM_SPK("Speaker", magician_spk_power), - SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias), - SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias), -}; - -/* magician machine audio_map */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* Headphone connected to VOUTL, VOUTR */ - {"Headphone Jack", NULL, "VOUTL"}, - {"Headphone Jack", NULL, "VOUTR"}, - - /* Speaker connected to VOUTL, VOUTR */ - {"Speaker", NULL, "VOUTL"}, - {"Speaker", NULL, "VOUTR"}, - - /* Mics are connected to VINM */ - {"VINM", NULL, "Headset Mic"}, - {"VINM", NULL, "Call Mic"}, -}; - -static const char * const input_select[] = {"Call Mic", "Headset Mic"}; -static const struct soc_enum magician_in_sel_enum = - SOC_ENUM_SINGLE_EXT(2, input_select); - -static const struct snd_kcontrol_new uda1380_magician_controls[] = { - SOC_SINGLE_BOOL_EXT("Headphone Switch", - (unsigned long)&magician_hp_switch, - magician_get_hp, magician_set_hp), - SOC_SINGLE_BOOL_EXT("Speaker Switch", - (unsigned long)&magician_spk_switch, - magician_get_spk, magician_set_spk), - SOC_ENUM_EXT("Input Select", magician_in_sel_enum, - magician_get_input, magician_set_input), -}; - -/* magician digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(playback, - DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.0")), - DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-0018", - "uda1380-hifi-playback")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(capture, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-0018", - "uda1380-hifi-capture")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link magician_dai[] = { -{ - .name = "uda1380", - .stream_name = "UDA1380 Playback", - .ops = &magician_playback_ops, - SND_SOC_DAILINK_REG(playback), -}, -{ - .name = "uda1380", - .stream_name = "UDA1380 Capture", - .ops = &magician_capture_ops, - SND_SOC_DAILINK_REG(capture), -} -}; - -/* magician audio machine driver */ -static struct snd_soc_card snd_soc_card_magician = { - .name = "Magician", - .owner = THIS_MODULE, - .dai_link = magician_dai, - .num_links = ARRAY_SIZE(magician_dai), - - .controls = uda1380_magician_controls, - .num_controls = ARRAY_SIZE(uda1380_magician_controls), - .dapm_widgets = uda1380_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static struct platform_device *magician_snd_device; - -/* - * FIXME: move into magician board file once merged into the pxa tree - */ -static struct uda1380_platform_data uda1380_info = { - .gpio_power = EGPIO_MAGICIAN_CODEC_POWER, - .gpio_reset = EGPIO_MAGICIAN_CODEC_RESET, - .dac_clk = UDA1380_DAC_CLK_WSPLL, -}; - -static struct i2c_board_info i2c_board_info[] = { - { - I2C_BOARD_INFO("uda1380", 0x18), - .platform_data = &uda1380_info, - }, -}; - -static int __init magician_init(void) -{ - int ret; - struct i2c_adapter *adapter; - struct i2c_client *client; - - if (!machine_is_magician()) - return -ENODEV; - - adapter = i2c_get_adapter(0); - if (!adapter) - return -ENODEV; - client = i2c_new_client_device(adapter, i2c_board_info); - i2c_put_adapter(adapter); - if (IS_ERR(client)) - return PTR_ERR(client); - - ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); - if (ret) - goto err_request_spk; - ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER"); - if (ret) - goto err_request_ep; - ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER"); - if (ret) - goto err_request_mic; - ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0"); - if (ret) - goto err_request_in_sel0; - ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1"); - if (ret) - goto err_request_in_sel1; - - gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); - - magician_snd_device = platform_device_alloc("soc-audio", -1); - if (!magician_snd_device) { - ret = -ENOMEM; - goto err_pdev; - } - - platform_set_drvdata(magician_snd_device, &snd_soc_card_magician); - ret = platform_device_add(magician_snd_device); - if (ret) { - platform_device_put(magician_snd_device); - goto err_pdev; - } - - return 0; - -err_pdev: - gpio_free(EGPIO_MAGICIAN_IN_SEL1); -err_request_in_sel1: - gpio_free(EGPIO_MAGICIAN_IN_SEL0); -err_request_in_sel0: - gpio_free(EGPIO_MAGICIAN_MIC_POWER); -err_request_mic: - gpio_free(EGPIO_MAGICIAN_EP_POWER); -err_request_ep: - gpio_free(EGPIO_MAGICIAN_SPK_POWER); -err_request_spk: - return ret; -} - -static void __exit magician_exit(void) -{ - platform_device_unregister(magician_snd_device); - - gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); - gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); - gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); - - gpio_free(EGPIO_MAGICIAN_IN_SEL1); - gpio_free(EGPIO_MAGICIAN_IN_SEL0); - gpio_free(EGPIO_MAGICIAN_MIC_POWER); - gpio_free(EGPIO_MAGICIAN_EP_POWER); - gpio_free(EGPIO_MAGICIAN_SPK_POWER); -} - -module_init(magician_init); -module_exit(magician_exit); - -MODULE_AUTHOR("Philipp Zabel"); -MODULE_DESCRIPTION("ALSA SoC Magician"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c deleted file mode 100644 index 763db7bbd9bb..000000000000 --- a/sound/soc/pxa/mioa701_wm9713.c +++ /dev/null @@ -1,201 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Handles the Mitac mioa701 SoC system - * - * Copyright (C) 2008 Robert Jarzmik - * - * This is a little schema of the sound interconnections : - * - * Sagem X200 Wolfson WM9713 - * +--------+ +-------------------+ Rear Speaker - * | | | | /-+ - * | +--->----->---+MONOIN SPKL+--->----+-+ | - * | GSM | | | | | | - * | +--->----->---+PCBEEP SPKR+--->----+-+ | - * | CHIP | | | \-+ - * | +---<-----<---+MONO | - * | | | | Front Speaker - * +--------+ | | /-+ - * | HPL+--->----+-+ | - * | | | | | - * | OUT3+--->----+-+ | - * | | \-+ - * | | - * | | Front Micro - * | | + - * | MIC1+-----<--+o+ - * | | + - * +-------------------+ --- - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/platform_device.h> - -#include <asm/mach-types.h> -#include <mach/audio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/initval.h> -#include <sound/ac97_codec.h> - -#include "../codecs/wm9713.h" - -#define AC97_GPIO_PULL 0x58 - -/* Use GPIO8 for rear speaker amplifier */ -static int rear_amp_power(struct snd_soc_component *component, int power) -{ - unsigned short reg; - - if (power) { - reg = snd_soc_component_read(component, AC97_GPIO_CFG); - snd_soc_component_write(component, AC97_GPIO_CFG, reg | 0x0100); - reg = snd_soc_component_read(component, AC97_GPIO_PULL); - snd_soc_component_write(component, AC97_GPIO_PULL, reg | (1<<15)); - } else { - reg = snd_soc_component_read(component, AC97_GPIO_CFG); - snd_soc_component_write(component, AC97_GPIO_CFG, reg & ~0x0100); - reg = snd_soc_component_read(component, AC97_GPIO_PULL); - snd_soc_component_write(component, AC97_GPIO_PULL, reg & ~(1<<15)); - } - - return 0; -} - -static int rear_amp_event(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kctl, int event) -{ - struct snd_soc_card *card = widget->dapm->card; - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_component *component; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = asoc_rtd_to_codec(rtd, 0)->component; - return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event)); -} - -/* mioa701 machine dapm widgets */ -static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Front Speaker", NULL), - SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event), - SND_SOC_DAPM_MIC("Headset", NULL), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Front Mic", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* Call Mic */ - {"Mic Bias", NULL, "Front Mic"}, - {"MIC1", NULL, "Mic Bias"}, - - /* Headset Mic */ - {"LINEL", NULL, "Headset Mic"}, - {"LINER", NULL, "Headset Mic"}, - - /* GSM Module */ - {"MONOIN", NULL, "GSM Line Out"}, - {"PCBEEP", NULL, "GSM Line Out"}, - {"GSM Line In", NULL, "MONO"}, - - /* headphone connected to HPL, HPR */ - {"Headset", NULL, "HPL"}, - {"Headset", NULL, "HPR"}, - - /* front speaker connected to HPL, OUT3 */ - {"Front Speaker", NULL, "HPL"}, - {"Front Speaker", NULL, "OUT3"}, - - /* rear speaker connected to SPKL, SPKR */ - {"Rear Speaker", NULL, "SPKL"}, - {"Rear Speaker", NULL, "SPKR"}, -}; - -static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; - - /* Prepare GPIO8 for rear speaker amplifier */ - snd_soc_component_update_bits(component, AC97_GPIO_CFG, 0x100, 0x100); - - /* Prepare MIC input */ - snd_soc_component_update_bits(component, AC97_3D_CONTROL, 0xc000, 0xc000); - - return 0; -} - -static struct snd_soc_ops mioa701_ops; - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link mioa701_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - .init = mioa701_wm9713_init, - .ops = &mioa701_ops, - SND_SOC_DAILINK_REG(ac97), - }, - { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - .ops = &mioa701_ops, - SND_SOC_DAILINK_REG(ac97_aux), - }, -}; - -static struct snd_soc_card mioa701 = { - .name = "MioA701", - .owner = THIS_MODULE, - .dai_link = mioa701_dai, - .num_links = ARRAY_SIZE(mioa701_dai), - - .dapm_widgets = mioa701_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static int mioa701_wm9713_probe(struct platform_device *pdev) -{ - int rc; - - if (!machine_is_mioa701()) - return -ENODEV; - - mioa701.dev = &pdev->dev; - rc = devm_snd_soc_register_card(&pdev->dev, &mioa701); - if (!rc) - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will " - "lead to overheating and possible destruction of your device." - " Do not use without a good knowledge of mio's board design!\n"); - return rc; -} - -static struct platform_driver mioa701_wm9713_driver = { - .probe = mioa701_wm9713_probe, - .driver = { - .name = "mioa701-wm9713", - .pm = &snd_soc_pm_ops, - }, -}; - -module_platform_driver(mioa701_wm9713_driver); - -/* Module information */ -MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); -MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:mioa701-wm9713"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c deleted file mode 100644 index 53fc49e32fbc..000000000000 --- a/sound/soc/pxa/mmp-pcm.c +++ /dev/null @@ -1,267 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * linux/sound/soc/pxa/mmp-pcm.c - * - * Copyright (C) 2011 Marvell International Ltd. - */ -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/dmaengine.h> -#include <linux/platform_data/dma-mmp_tdma.h> -#include <linux/platform_data/mmp_audio.h> - -#include <sound/pxa2xx-lib.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/dmaengine_pcm.h> - -#define DRV_NAME "mmp-pcm" - -struct mmp_dma_data { - int ssp_id; - struct resource *dma_res; -}; - -#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \ - SNDRV_PCM_INFO_MMAP_VALID | \ - SNDRV_PCM_INFO_INTERLEAVED | \ - SNDRV_PCM_INFO_PAUSE | \ - SNDRV_PCM_INFO_RESUME | \ - SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) - -static struct snd_pcm_hardware mmp_pcm_hardware[] = { - { - .info = MMP_PCM_INFO, - .period_bytes_min = 1024, - .period_bytes_max = 2048, - .periods_min = 2, - .periods_max = 32, - .buffer_bytes_max = 4096, - .fifo_size = 32, - }, - { - .info = MMP_PCM_INFO, - .period_bytes_min = 1024, - .period_bytes_max = 2048, - .periods_min = 2, - .periods_max = 32, - .buffer_bytes_max = 4096, - .fifo_size = 32, - }, -}; - -static int mmp_pcm_hw_params(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct dma_slave_config slave_config; - int ret; - - ret = - snd_dmaengine_pcm_prepare_slave_config(substream, params, - &slave_config); - if (ret) - return ret; - - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int mmp_pcm_trigger(struct snd_soc_component *component, - struct snd_pcm_substream *substream, int cmd) -{ - return snd_dmaengine_pcm_trigger(substream, cmd); -} - -static snd_pcm_uframes_t mmp_pcm_pointer(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - return snd_dmaengine_pcm_pointer(substream); -} - -static bool filter(struct dma_chan *chan, void *param) -{ - struct mmp_dma_data *dma_data = param; - bool found = false; - char *devname; - - devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, - dma_data->ssp_id); - if ((strcmp(dev_name(chan->device->dev), devname) == 0) && - (chan->chan_id == dma_data->dma_res->start)) { - found = true; - } - - kfree(devname); - return found; -} - -static int mmp_pcm_open(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct platform_device *pdev = to_platform_device(component->dev); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - struct mmp_dma_data dma_data; - struct resource *r; - - r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); - if (!r) - return -EBUSY; - - snd_soc_set_runtime_hwparams(substream, - &mmp_pcm_hardware[substream->stream]); - - dma_data.dma_res = r; - dma_data.ssp_id = cpu_dai->id; - - return snd_dmaengine_pcm_open_request_chan(substream, filter, - &dma_data); -} - -static int mmp_pcm_close(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - return snd_dmaengine_pcm_close_release_chan(substream); -} - -static int mmp_pcm_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long off = vma->vm_pgoff; - - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); - return remap_pfn_range(vma, vma->vm_start, - __phys_to_pfn(runtime->dma_addr) + off, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - -static void mmp_pcm_free_dma_buffers(struct snd_soc_component *component, - struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - struct gen_pool *gpool; - - gpool = sram_get_gpool("asram"); - if (!gpool) - return; - - for (stream = 0; stream < 2; stream++) { - size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; - - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - gen_pool_free(gpool, (unsigned long)buf->area, size); - buf->area = NULL; - } - -} - -static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, - int stream) -{ - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; - struct gen_pool *gpool; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = substream->pcm->card->dev; - buf->private_data = NULL; - - gpool = sram_get_gpool("asram"); - if (!gpool) - return -ENOMEM; - - buf->area = gen_pool_dma_alloc(gpool, size, &buf->addr); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static int mmp_pcm_new(struct snd_soc_component *component, - struct snd_soc_pcm_runtime *rtd) -{ - struct snd_pcm_substream *substream; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0, stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - - ret = mmp_pcm_preallocate_dma_buffer(substream, stream); - if (ret) - goto err; - } - - return 0; - -err: - mmp_pcm_free_dma_buffers(component, pcm); - return ret; -} - -static const struct snd_soc_component_driver mmp_soc_component = { - .name = DRV_NAME, - .open = mmp_pcm_open, - .close = mmp_pcm_close, - .hw_params = mmp_pcm_hw_params, - .trigger = mmp_pcm_trigger, - .pointer = mmp_pcm_pointer, - .mmap = mmp_pcm_mmap, - .pcm_construct = mmp_pcm_new, - .pcm_destruct = mmp_pcm_free_dma_buffers, -}; - -static int mmp_pcm_probe(struct platform_device *pdev) -{ - struct mmp_audio_platdata *pdata = pdev->dev.platform_data; - - if (pdata) { - mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max = - pdata->buffer_max_playback; - mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max = - pdata->period_max_playback; - mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max = - pdata->buffer_max_capture; - mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max = - pdata->period_max_capture; - } - return devm_snd_soc_register_component(&pdev->dev, &mmp_soc_component, - NULL, 0); -} - -static struct platform_driver mmp_pcm_driver = { - .driver = { - .name = "mmp-pcm-audio", - }, - - .probe = mmp_pcm_probe, -}; - -module_platform_driver(mmp_pcm_driver); - -MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); -MODULE_DESCRIPTION("MMP Soc Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:mmp-pcm-audio"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 4255851c71c1..73f36c9dd35c 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -171,11 +171,11 @@ static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, sspa->sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; sspa->ctrl = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: sspa->sp |= SSPA_SP_MSL; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: break; default: return -EINVAL; @@ -239,12 +239,16 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + sspa_ctrl &= ~SSPA_CTL_XPH; if (dev->of_node || params_channels(params) == 2) sspa_ctrl |= SSPA_CTL_XPH; sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; sspa_ctrl |= SSPA_CTL_XWDLEN1(bitval); + sspa_ctrl &= ~SSPA_CTL_XWDLEN2_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN2(bitval); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; sspa_ctrl |= SSPA_CTL_XSSZ1(bitval); @@ -326,7 +330,6 @@ static int mmp_sspa_probe(struct snd_soc_dai *dai) &sspa->playback_dma_data, &sspa->capture_dma_data); - snd_soc_dai_set_drvdata(dai, sspa); return 0; } @@ -337,6 +340,7 @@ static int mmp_sspa_probe(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { + .probe = mmp_sspa_probe, .startup = mmp_sspa_startup, .shutdown = mmp_sspa_shutdown, .trigger = mmp_sspa_trigger, @@ -347,7 +351,6 @@ static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { }; static struct snd_soc_dai_driver mmp_sspa_dai = { - .probe = mmp_sspa_probe, .playback = { .channels_min = 1, .channels_max = 128, @@ -401,7 +404,7 @@ static int mmp_pcm_mmap(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct vm_area_struct *vma) { - vma->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + vm_flags_set(vma, VM_DONTEXPAND | VM_DONTDUMP); vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); return remap_pfn_range(vma, vma->vm_start, substream->dma_buffer.addr >> PAGE_SHIFT, @@ -453,10 +456,11 @@ static int mmp_sspa_close(struct snd_soc_component *component, } static const struct snd_soc_component_driver mmp_sspa_component = { - .name = "mmp-sspa", - .mmap = mmp_pcm_mmap, - .open = mmp_sspa_open, - .close = mmp_sspa_close, + .name = "mmp-sspa", + .mmap = mmp_pcm_mmap, + .open = mmp_sspa_open, + .close = mmp_sspa_close, + .legacy_dai_naming = 1, }; static int asoc_mmp_sspa_probe(struct platform_device *pdev) @@ -541,7 +545,7 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) return 0; } -static int asoc_mmp_sspa_remove(struct platform_device *pdev) +static void asoc_mmp_sspa_remove(struct platform_device *pdev) { struct sspa_priv *sspa = platform_get_drvdata(pdev); @@ -549,11 +553,10 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); if (pdev->dev.of_node) - return 0; + return; clk_put(sspa->audio_clk); clk_put(sspa->sysclk); - return 0; } #ifdef CONFIG_OF diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c deleted file mode 100644 index b92ea1a0453f..000000000000 --- a/sound/soc/pxa/palm27x.c +++ /dev/null @@ -1,161 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * linux/sound/soc/pxa/palm27x.c - * - * SoC Audio driver for Palm T|X, T5 and LifeDrive - * - * based on tosa.c - * - * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <mach/audio.h> -#include <linux/platform_data/asoc-palm27x.h> - -static struct snd_soc_jack hs_jack; - -/* Headphones jack detection DAPM pins */ -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -/* Headphones jack detection gpios */ -static struct snd_soc_jack_gpio hs_jack_gpios[] = { - [0] = { - /* gpio is set on per-platform basis */ - .name = "hp-gpio", - .report = SND_JACK_HEADPHONE, - .debounce_time = 200, - }, -}; - -/* Palm27x machine dapm widgets */ -static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext. Speaker", NULL), - SND_SOC_DAPM_MIC("Ext. Microphone", NULL), -}; - -/* PalmTX audio map */ -static const struct snd_soc_dapm_route audio_map[] = { - /* headphone connected to HPOUTL, HPOUTR */ - {"Headphone Jack", NULL, "HPOUTL"}, - {"Headphone Jack", NULL, "HPOUTR"}, - - /* ext speaker connected to ROUT2, LOUT2 */ - {"Ext. Speaker", NULL, "LOUT2"}, - {"Ext. Speaker", NULL, "ROUT2"}, - - /* mic connected to MIC1 */ - {"MIC1", NULL, "Ext. Microphone"}, -}; - -static struct snd_soc_card palm27x_asoc; - -static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) -{ - int err; - - /* Jack detection API stuff */ - err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins, - ARRAY_SIZE(hs_jack_pins)); - if (err) - return err; - - err = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return err; -} - -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link palm27x_dai[] = { -{ - .name = "AC97 HiFi", - .stream_name = "AC97 HiFi", - .init = palm27x_ac97_init, - SND_SOC_DAILINK_REG(hifi), -}, -{ - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - SND_SOC_DAILINK_REG(aux), -}, -}; - -static struct snd_soc_card palm27x_asoc = { - .name = "Palm/PXA27x", - .owner = THIS_MODULE, - .dai_link = palm27x_dai, - .num_links = ARRAY_SIZE(palm27x_dai), - .dapm_widgets = palm27x_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static int palm27x_asoc_probe(struct platform_device *pdev) -{ - int ret; - - if (!(machine_is_palmtx() || machine_is_palmt5() || - machine_is_palmld() || machine_is_palmte2())) - return -ENODEV; - - if (!pdev->dev.platform_data) { - dev_err(&pdev->dev, "please supply platform_data\n"); - return -ENODEV; - } - - hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *) - (pdev->dev.platform_data))->jack_gpio; - - palm27x_asoc.dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static struct platform_driver palm27x_wm9712_driver = { - .probe = palm27x_asoc_probe, - .driver = { - .name = "palm27x-asoc", - .pm = &snd_soc_pm_ops, - }, -}; - -module_platform_driver(palm27x_wm9712_driver); - -/* Module information */ -MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:palm27x-asoc"); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c deleted file mode 100644 index 323ba3e23039..000000000000 --- a/sound/soc/pxa/poodle.c +++ /dev/null @@ -1,288 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * poodle.c -- SoC audio for Poodle - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <lrg@slimlogic.co.uk> - * Richard Purdie <richard@openedhand.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/i2c.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <asm/hardware/locomo.h> -#include <mach/poodle.h> -#include <mach/audio.h> - -#include "../codecs/wm8731.h" -#include "pxa2xx-i2s.h" - -#define POODLE_HP 1 -#define POODLE_HP_OFF 0 -#define POODLE_SPK_ON 1 -#define POODLE_SPK_OFF 0 - - /* audio clock in Hz - rounded from 12.235MHz */ -#define POODLE_AUDIO_CLOCK 12288000 - -static int poodle_jack_func; -static int poodle_spk_func; - -static void poodle_ext_control(struct snd_soc_dapm_context *dapm) -{ - /* set up jack connection */ - if (poodle_jack_func == POODLE_HP) { - /* set = unmute headphone */ - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_L, 1); - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - } else { - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_L, 0); - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - } - - /* set the endpoints to their new connection states */ - if (poodle_spk_func == POODLE_SPK_ON) - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - else - snd_soc_dapm_disable_pin(dapm, "Ext Spk"); - - /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); -} - -static int poodle_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - - /* check the jack status at stream startup */ - poodle_ext_control(&rtd->card->dapm); - - return 0; -} - -/* we need to unmute the HP at shutdown as the mute burns power on poodle */ -static void poodle_shutdown(struct snd_pcm_substream *substream) -{ - /* set = unmute headphone */ - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_L, 1); - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_R, 1); -} - -static int poodle_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set the I2S system clock as input (unused) */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops poodle_ops = { - .startup = poodle_startup, - .hw_params = poodle_hw_params, - .shutdown = poodle_shutdown, -}; - -static int poodle_get_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = poodle_jack_func; - return 0; -} - -static int poodle_set_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (poodle_jack_func == ucontrol->value.enumerated.item[0]) - return 0; - - poodle_jack_func = ucontrol->value.enumerated.item[0]; - poodle_ext_control(&card->dapm); - return 1; -} - -static int poodle_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = poodle_spk_func; - return 0; -} - -static int poodle_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (poodle_spk_func == ucontrol->value.enumerated.item[0]) - return 0; - - poodle_spk_func = ucontrol->value.enumerated.item[0]; - poodle_ext_control(&card->dapm); - return 1; -} - -static int poodle_amp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - if (SND_SOC_DAPM_EVENT_ON(event)) - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_AMP_ON, 0); - else - locomo_gpio_write(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_AMP_ON, 1); - - return 0; -} - -/* poodle machine dapm widgets */ -static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { -SND_SOC_DAPM_HP("Headphone Jack", NULL), -SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), -SND_SOC_DAPM_MIC("Microphone", NULL), -}; - -/* Corgi machine connections to the codec pins */ -static const struct snd_soc_dapm_route poodle_audio_map[] = { - - /* headphone connected to LHPOUT1, RHPOUT1 */ - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - /* speaker connected to LOUT, ROUT */ - {"Ext Spk", NULL, "ROUT"}, - {"Ext Spk", NULL, "LOUT"}, - - {"MICIN", NULL, "Microphone"}, -}; - -static const char * const jack_function[] = {"Off", "Headphone"}; -static const char * const spk_function[] = {"Off", "On"}; -static const struct soc_enum poodle_enum[] = { - SOC_ENUM_SINGLE_EXT(2, jack_function), - SOC_ENUM_SINGLE_EXT(2, spk_function), -}; - -static const struct snd_kcontrol_new wm8731_poodle_controls[] = { - SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, - poodle_set_jack), - SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, - poodle_set_spk), -}; - -/* poodle digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(wm8731, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link poodle_dai = { - .name = "WM8731", - .stream_name = "WM8731", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &poodle_ops, - SND_SOC_DAILINK_REG(wm8731), -}; - -/* poodle audio machine driver */ -static struct snd_soc_card poodle = { - .name = "Poodle", - .dai_link = &poodle_dai, - .num_links = 1, - .owner = THIS_MODULE, - - .controls = wm8731_poodle_controls, - .num_controls = ARRAY_SIZE(wm8731_poodle_controls), - .dapm_widgets = wm8731_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), - .dapm_routes = poodle_audio_map, - .num_dapm_routes = ARRAY_SIZE(poodle_audio_map), - .fully_routed = true, -}; - -static int poodle_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &poodle; - int ret; - - locomo_gpio_set_dir(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_AMP_ON, 0); - /* should we mute HP at startup - burning power ?*/ - locomo_gpio_set_dir(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_L, 0); - locomo_gpio_set_dir(&poodle_locomo_device.dev, - POODLE_LOCOMO_GPIO_MUTE_R, 0); - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static struct platform_driver poodle_driver = { - .driver = { - .name = "poodle-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = poodle_probe, -}; - -module_platform_driver(poodle_driver); - -/* Module information */ -MODULE_AUTHOR("Richard Purdie"); -MODULE_DESCRIPTION("ALSA SoC Poodle"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:poodle-audio"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c4e7307a4437..b8a3cb8b7597 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -61,22 +61,6 @@ static void dump_registers(struct ssp_device *ssp) pxa_ssp_read_reg(ssp, SSACD)); } -static void pxa_ssp_enable(struct ssp_device *ssp) -{ - uint32_t sscr0; - - sscr0 = __raw_readl(ssp->mmio_base + SSCR0) | SSCR0_SSE; - __raw_writel(sscr0, ssp->mmio_base + SSCR0); -} - -static void pxa_ssp_disable(struct ssp_device *ssp) -{ - uint32_t sscr0; - - sscr0 = __raw_readl(ssp->mmio_base + SSCR0) & ~SSCR0_SSE; - __raw_writel(sscr0, ssp->mmio_base + SSCR0); -} - static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, int out, struct snd_dmaengine_dai_dma_data *dma) { @@ -99,8 +83,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - if (priv->extclk) - clk_prepare_enable(priv->extclk); + clk_prepare_enable(priv->extclk); dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) @@ -124,8 +107,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } - if (priv->extclk) - clk_disable_unprepare(priv->extclk); + clk_disable_unprepare(priv->extclk); kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); @@ -390,10 +372,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -450,14 +432,14 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) sscr1 |= SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -502,9 +484,9 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) pxa_ssp_write_reg(ssp, SSCR1, sscr1); pxa_ssp_write_reg(ssp, SSPSP, sspsp); - switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR; pxa_ssp_write_reg(ssp, SSCR1, scfr); @@ -609,7 +591,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, if (ret < 0) { const struct pxa_ssp_clock_mode *m; - int ssacd, acds; + int ssacd; for (m = pxa_ssp_clock_modes; m->rate; m++) { if (m->rate == rate) @@ -619,12 +601,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, if (!m->rate) return -EINVAL; - acds = m->acds; - - /* The values in the table are for 16 bits */ - if (width == 32) - acds--; - ret = pxa_ssp_set_pll(priv, bclk); if (ret < 0) return ret; @@ -797,7 +773,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (IS_ERR(priv->extclk)) { ret = PTR_ERR(priv->extclk); if (ret == -EPROBE_DEFER) - return ret; + goto err_priv; priv->extclk = NULL; } @@ -837,6 +813,8 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) #define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, .startup = pxa_ssp_startup, .shutdown = pxa_ssp_shutdown, .trigger = pxa_ssp_trigger, @@ -848,8 +826,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { }; static struct snd_soc_dai_driver pxa_ssp_dai = { - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, .playback = { .channels_min = 1, .channels_max = 8, @@ -866,19 +842,17 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { }; static const struct snd_soc_component_driver pxa_ssp_component = { - .name = "pxa-ssp", - .pcm_construct = pxa2xx_soc_pcm_new, - .pcm_destruct = pxa2xx_soc_pcm_free, - .open = pxa2xx_soc_pcm_open, - .close = pxa2xx_soc_pcm_close, - .hw_params = pxa2xx_soc_pcm_hw_params, - .hw_free = pxa2xx_soc_pcm_hw_free, - .prepare = pxa2xx_soc_pcm_prepare, - .trigger = pxa2xx_soc_pcm_trigger, - .pointer = pxa2xx_soc_pcm_pointer, - .mmap = pxa2xx_soc_pcm_mmap, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, + .name = "pxa-ssp", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .legacy_dai_naming = 1, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4240fde6aae8..78f50032afc5 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -21,9 +21,11 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/hardware.h> -#include <mach/regs-ac97.h> -#include <mach/audio.h> +#include <linux/platform_data/asoc-pxa.h> + +#define PCDR 0x0040 /* PCM FIFO Data Register */ +#define MODR 0x0140 /* Modem FIFO Data Register */ +#define MCDR 0x0060 /* Mic-in FIFO Data Register */ static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { @@ -59,35 +61,30 @@ static struct ac97_controller_ops pxa2xx_ac97_ops = { }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { - .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { - .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { - .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, .chan_name = "pcm_aux_mono_out", .maxburst = 16, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { - .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, .chan_name = "pcm_aux_mono_in", .maxburst = 16, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { - .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, .chan_name = "pcm_aux_mic_mono", .maxburst = 16, @@ -202,15 +199,12 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", .pcm_construct = pxa2xx_soc_pcm_new, - .pcm_destruct = pxa2xx_soc_pcm_free, .open = pxa2xx_soc_pcm_open, .close = pxa2xx_soc_pcm_close, .hw_params = pxa2xx_soc_pcm_hw_params, - .hw_free = pxa2xx_soc_pcm_hw_free, .prepare = pxa2xx_soc_pcm_prepare, .trigger = pxa2xx_soc_pcm_trigger, .pointer = pxa2xx_soc_pcm_pointer, - .mmap = pxa2xx_soc_pcm_mmap, }; #ifdef CONFIG_OF @@ -229,6 +223,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) int ret; struct ac97_controller *ctrl; pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + struct resource *regs; void **codecs_pdata; if (pdev->id != -1) { @@ -236,6 +231,16 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return -ENXIO; } + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) + return -ENXIO; + + pxa2xx_ac97_pcm_stereo_in.addr = regs->start + PCDR; + pxa2xx_ac97_pcm_stereo_out.addr = regs->start + PCDR; + pxa2xx_ac97_pcm_aux_mono_out.addr = regs->start + MODR; + pxa2xx_ac97_pcm_aux_mono_in.addr = regs->start + MODR; + pxa2xx_ac97_pcm_mic_mono_in.addr = regs->start + MCDR; + ret = pxa2xx_ac97_hw_probe(pdev); if (ret) { dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret); @@ -258,16 +263,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver)); } -static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) +static void pxa2xx_ac97_dev_remove(struct platform_device *pdev) { struct ac97_controller *ctrl = platform_get_drvdata(pdev); snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); - return 0; } -#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_dev_suspend(struct device *dev) { return pxa2xx_ac97_hw_suspend(); @@ -278,18 +281,15 @@ static int pxa2xx_ac97_dev_resume(struct device *dev) return pxa2xx_ac97_hw_resume(); } -static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, +static DEFINE_SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume); -#endif static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, .remove = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", -#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, -#endif .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), }, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 5301859a8453..849fbf176a70 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -21,21 +21,20 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/hardware.h> -#include <mach/audio.h> +#include <linux/platform_data/asoc-pxa.h> #include "pxa2xx-i2s.h" /* * I2S Controller Register and Bit Definitions */ -#define SACR0 __REG(0x40400000) /* Global Control Register */ -#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */ -#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */ -#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */ -#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */ -#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */ -#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */ +#define SACR0 (0x0000) /* Global Control Register */ +#define SACR1 (0x0004) /* Serial Audio I 2 S/MSB-Justified Control Register */ +#define SASR0 (0x000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */ +#define SAIMR (0x0014) /* Serial Audio Interrupt Mask Register */ +#define SAICR (0x0018) /* Serial Audio Interrupt Clear Register */ +#define SADIV (0x0060) /* Audio Clock Divider Register. */ +#define SADR (0x0080) /* Serial Audio Data Register (TX and RX FIFO access Register). */ #define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */ #define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */ @@ -77,16 +76,15 @@ struct pxa_i2s_port { static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; +static void __iomem *i2s_reg_base; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { - .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, .chan_name = "tx", .maxburst = 32, }; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { - .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, .chan_name = "rx", .maxburst = 32, @@ -95,14 +93,14 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); if (!snd_soc_dai_active(cpu_dai)) - SACR0 = 0; + writel(0, i2s_reg_base + SACR0); return 0; } @@ -114,7 +112,7 @@ static int pxa_i2s_wait(void) /* flush the Rx FIFO */ for (i = 0; i < 16; i++) - SADR; + readl(i2s_reg_base + SADR); return 0; } @@ -131,11 +129,11 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: pxa_i2s.master = 1; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: pxa_i2s.master = 0; break; default: @@ -174,39 +172,39 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { - SACR0 = 0; + writel(0, i2s_reg_base + SACR0); if (pxa_i2s.master) - SACR0 |= SACR0_BCKD; + writel(readl(i2s_reg_base + SACR0) | (SACR0_BCKD), i2s_reg_base + SACR0); - SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1); - SACR1 |= pxa_i2s.fmt; + writel(readl(i2s_reg_base + SACR0) | (SACR0_RFTH(14) | SACR0_TFTH(1)), i2s_reg_base + SACR0); + writel(readl(i2s_reg_base + SACR1) | (pxa_i2s.fmt), i2s_reg_base + SACR1); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - SAIMR |= SAIMR_TFS; + writel(readl(i2s_reg_base + SAIMR) | (SAIMR_TFS), i2s_reg_base + SAIMR); else - SAIMR |= SAIMR_RFS; + writel(readl(i2s_reg_base + SAIMR) | (SAIMR_RFS), i2s_reg_base + SAIMR); switch (params_rate(params)) { case 8000: - SADIV = 0x48; + writel(0x48, i2s_reg_base + SADIV); break; case 11025: - SADIV = 0x34; + writel(0x34, i2s_reg_base + SADIV); break; case 16000: - SADIV = 0x24; + writel(0x24, i2s_reg_base + SADIV); break; case 22050: - SADIV = 0x1a; + writel(0x1a, i2s_reg_base + SADIV); break; case 44100: - SADIV = 0xd; + writel(0xd, i2s_reg_base + SADIV); break; case 48000: - SADIV = 0xc; + writel(0xc, i2s_reg_base + SADIV); break; case 96000: /* not in manual and possibly slightly inaccurate */ - SADIV = 0x6; + writel(0x6, i2s_reg_base + SADIV); break; } @@ -221,10 +219,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - SACR1 &= ~SACR1_DRPL; + writel(readl(i2s_reg_base + SACR1) & (~SACR1_DRPL), i2s_reg_base + SACR1); else - SACR1 &= ~SACR1_DREC; - SACR0 |= SACR0_ENB; + writel(readl(i2s_reg_base + SACR1) & (~SACR1_DREC), i2s_reg_base + SACR1); + writel(readl(i2s_reg_base + SACR0) | (SACR0_ENB), i2s_reg_base + SACR0); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -243,15 +241,15 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - SACR1 |= SACR1_DRPL; - SAIMR &= ~SAIMR_TFS; + writel(readl(i2s_reg_base + SACR1) | (SACR1_DRPL), i2s_reg_base + SACR1); + writel(readl(i2s_reg_base + SAIMR) & (~SAIMR_TFS), i2s_reg_base + SAIMR); } else { - SACR1 |= SACR1_DREC; - SAIMR &= ~SAIMR_RFS; + writel(readl(i2s_reg_base + SACR1) | (SACR1_DREC), i2s_reg_base + SACR1); + writel(readl(i2s_reg_base + SAIMR) & (~SAIMR_RFS), i2s_reg_base + SAIMR); } - if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { - SACR0 &= ~SACR0_ENB; + if ((readl(i2s_reg_base + SACR1) & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { + writel(readl(i2s_reg_base + SACR0) & (~SACR0_ENB), i2s_reg_base + SACR0); pxa_i2s_wait(); if (clk_ena) { clk_disable_unprepare(clk_i2s); @@ -264,13 +262,13 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, static int pxa2xx_soc_pcm_suspend(struct snd_soc_component *component) { /* store registers */ - pxa_i2s.sacr0 = SACR0; - pxa_i2s.sacr1 = SACR1; - pxa_i2s.saimr = SAIMR; - pxa_i2s.sadiv = SADIV; + pxa_i2s.sacr0 = readl(i2s_reg_base + SACR0); + pxa_i2s.sacr1 = readl(i2s_reg_base + SACR1); + pxa_i2s.saimr = readl(i2s_reg_base + SAIMR); + pxa_i2s.sadiv = readl(i2s_reg_base + SADIV); /* deactivate link */ - SACR0 &= ~SACR0_ENB; + writel(readl(i2s_reg_base + SACR0) & (~SACR0_ENB), i2s_reg_base + SACR0); pxa_i2s_wait(); return 0; } @@ -279,12 +277,12 @@ static int pxa2xx_soc_pcm_resume(struct snd_soc_component *component) { pxa_i2s_wait(); - SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB; - SACR1 = pxa_i2s.sacr1; - SAIMR = pxa_i2s.saimr; - SADIV = pxa_i2s.sadiv; + writel(pxa_i2s.sacr0 & ~SACR0_ENB, i2s_reg_base + SACR0); + writel(pxa_i2s.sacr1, i2s_reg_base + SACR1); + writel(pxa_i2s.saimr, i2s_reg_base + SAIMR); + writel(pxa_i2s.sadiv, i2s_reg_base + SADIV); - SACR0 = pxa_i2s.sacr0; + writel(pxa_i2s.sacr0, i2s_reg_base + SACR0); return 0; } @@ -306,12 +304,12 @@ static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) * the SACR0[RST] bit must also be set and cleared to reset all * I2S controller registers. */ - SACR0 = SACR0_RST; - SACR0 = 0; + writel(SACR0_RST, i2s_reg_base + SACR0); + writel(0, i2s_reg_base + SACR0); /* Make sure RPL and REC are disabled */ - SACR1 = SACR1_DRPL | SACR1_DREC; + writel(SACR1_DRPL | SACR1_DREC, i2s_reg_base + SACR1); /* Along with FIFO servicing */ - SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + writel(readl(i2s_reg_base + SAIMR) & (~(SAIMR_RFS | SAIMR_TFS)), i2s_reg_base + SAIMR); snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out, &pxa2xx_i2s_pcm_stereo_in); @@ -331,6 +329,8 @@ static int pxa2xx_i2s_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .probe = pxa2xx_i2s_probe, + .remove = pxa2xx_i2s_remove, .startup = pxa2xx_i2s_startup, .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, @@ -340,8 +340,6 @@ static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { }; static struct snd_soc_dai_driver pxa_i2s_dai = { - .probe = pxa2xx_i2s_probe, - .remove = pxa2xx_i2s_remove, .playback = { .channels_min = 2, .channels_max = 2, @@ -353,27 +351,34 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { .rates = PXA2XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = &pxa_i2s_dai_ops, - .symmetric_rates = 1, + .symmetric_rate = 1, }; static const struct snd_soc_component_driver pxa_i2s_component = { - .name = "pxa-i2s", - .pcm_construct = pxa2xx_soc_pcm_new, - .pcm_destruct = pxa2xx_soc_pcm_free, - .open = pxa2xx_soc_pcm_open, - .close = pxa2xx_soc_pcm_close, - .hw_params = pxa2xx_soc_pcm_hw_params, - .hw_free = pxa2xx_soc_pcm_hw_free, - .prepare = pxa2xx_soc_pcm_prepare, - .trigger = pxa2xx_soc_pcm_trigger, - .pointer = pxa2xx_soc_pcm_pointer, - .mmap = pxa2xx_soc_pcm_mmap, - .suspend = pxa2xx_soc_pcm_suspend, - .resume = pxa2xx_soc_pcm_resume, + .name = "pxa-i2s", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, + .suspend = pxa2xx_soc_pcm_suspend, + .resume = pxa2xx_soc_pcm_resume, + .legacy_dai_naming = 1, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) { + struct resource *res; + + i2s_reg_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); + if (IS_ERR(i2s_reg_base)) + return PTR_ERR(i2s_reg_base); + + pxa2xx_i2s_pcm_stereo_out.addr = res->start + SADR; + pxa2xx_i2s_pcm_stereo_in.addr = res->start + SADR; + return devm_snd_soc_register_component(&pdev->dev, &pxa_i2s_component, &pxa_i2s_dai, 1); } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2b7839715dd5..9d6c41f775e5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -19,15 +19,12 @@ static const struct snd_soc_component_driver pxa2xx_soc_platform = { .pcm_construct = pxa2xx_soc_pcm_new, - .pcm_destruct = pxa2xx_soc_pcm_free, .open = pxa2xx_soc_pcm_open, .close = pxa2xx_soc_pcm_close, .hw_params = pxa2xx_soc_pcm_hw_params, - .hw_free = pxa2xx_soc_pcm_hw_free, .prepare = pxa2xx_soc_pcm_prepare, .trigger = pxa2xx_soc_pcm_trigger, .pointer = pxa2xx_soc_pcm_pointer, - .mmap = pxa2xx_soc_pcm_mmap, }; static int pxa2xx_soc_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 7c1384a869ca..6ca0a6bcd34c 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -14,13 +14,12 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-i2s.h" @@ -37,7 +36,7 @@ static int spitz_jack_func; static int spitz_spk_func; -static int spitz_mic_gpio; +static struct gpio_desc *gpiod_mic, *gpiod_mute_l, *gpiod_mute_r; static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { @@ -56,8 +55,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 1); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 1); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -65,8 +64,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -74,8 +73,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -83,8 +82,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_HP_OFF: @@ -93,8 +92,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; } @@ -105,7 +104,7 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) static int spitz_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); /* check the jack status at stream startup */ spitz_ext_control(&rtd->card->dapm); @@ -116,9 +115,9 @@ static int spitz_startup(struct snd_pcm_substream *substream) static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; @@ -199,7 +198,7 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(gpiod_mic, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -261,7 +260,7 @@ static struct snd_soc_dai_link spitz_dai = { .name = "wm8750", .stream_name = "WM8750", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ops = &spitz_ops, SND_SOC_DAILINK_REG(wm8750), }; @@ -287,49 +286,32 @@ static int spitz_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (machine_is_akita()) - spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; - else - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - - ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); - if (ret) - goto err1; - - ret = gpio_direction_output(spitz_mic_gpio, 0); - if (ret) - goto err2; + gpiod_mic = devm_gpiod_get(&pdev->dev, "mic", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mic)) + return PTR_ERR(gpiod_mic); + gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_l)) + return PTR_ERR(gpiod_mute_l); + gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_r)) + return PTR_ERR(gpiod_mute_r); card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err2; - } - return 0; - -err2: - gpio_free(spitz_mic_gpio); -err1: return ret; } -static int spitz_remove(struct platform_device *pdev) -{ - gpio_free(spitz_mic_gpio); - return 0; -} - static struct platform_driver spitz_driver = { .driver = { .name = "spitz-audio", .pm = &snd_soc_pm_ops, }, .probe = spitz_probe, - .remove = spitz_remove, }; module_platform_driver(spitz_driver); diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c deleted file mode 100644 index 3b40b5fa5de7..000000000000 --- a/sound/soc/pxa/tosa.c +++ /dev/null @@ -1,262 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * tosa.c -- SoC audio for Tosa - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <lrg@slimlogic.co.uk> - * Richard Purdie <richard@openedhand.com> - * - * GPIO's - * 1 - Jack Insertion - * 5 - Hookswitch (headset answer/hang up switch) - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/tosa.h> -#include <mach/audio.h> - -#define TOSA_HP 0 -#define TOSA_MIC_INT 1 -#define TOSA_HEADSET 2 -#define TOSA_HP_OFF 3 -#define TOSA_SPK_ON 0 -#define TOSA_SPK_OFF 1 - -static int tosa_jack_func; -static int tosa_spk_func; - -static void tosa_ext_control(struct snd_soc_dapm_context *dapm) -{ - - snd_soc_dapm_mutex_lock(dapm); - - /* set up jack connection */ - switch (tosa_jack_func) { - case TOSA_HP: - snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); - break; - case TOSA_MIC_INT: - snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); - break; - case TOSA_HEADSET: - snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - break; - } - - if (tosa_spk_func == TOSA_SPK_ON) - snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker"); - else - snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker"); - - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); -} - -static int tosa_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - - /* check the jack status at stream startup */ - tosa_ext_control(&rtd->card->dapm); - - return 0; -} - -static const struct snd_soc_ops tosa_ops = { - .startup = tosa_startup, -}; - -static int tosa_get_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = tosa_jack_func; - return 0; -} - -static int tosa_set_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (tosa_jack_func == ucontrol->value.enumerated.item[0]) - return 0; - - tosa_jack_func = ucontrol->value.enumerated.item[0]; - tosa_ext_control(&card->dapm); - return 1; -} - -static int tosa_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = tosa_spk_func; - return 0; -} - -static int tosa_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - - if (tosa_spk_func == ucontrol->value.enumerated.item[0]) - return 0; - - tosa_spk_func = ucontrol->value.enumerated.item[0]; - tosa_ext_control(&card->dapm); - return 1; -} - -/* tosa dapm event handlers */ -static int tosa_hp_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); - return 0; -} - -/* tosa machine dapm widgets */ -static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = { -SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event), -SND_SOC_DAPM_HP("Headset Jack", NULL), -SND_SOC_DAPM_MIC("Mic (Internal)", NULL), -SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -/* tosa audio map */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* headphone connected to HPOUTL, HPOUTR */ - {"Headphone Jack", NULL, "HPOUTL"}, - {"Headphone Jack", NULL, "HPOUTR"}, - - /* ext speaker connected to LOUT2, ROUT2 */ - {"Speaker", NULL, "LOUT2"}, - {"Speaker", NULL, "ROUT2"}, - - /* internal mic is connected to mic1, mic2 differential - with bias */ - {"MIC1", NULL, "Mic Bias"}, - {"MIC2", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Mic (Internal)"}, - - /* headset is connected to HPOUTR, and LINEINR with bias */ - {"Headset Jack", NULL, "HPOUTR"}, - {"LINEINR", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Jack"}, -}; - -static const char * const jack_function[] = {"Headphone", "Mic", "Line", - "Headset", "Off"}; -static const char * const spk_function[] = {"On", "Off"}; -static const struct soc_enum tosa_enum[] = { - SOC_ENUM_SINGLE_EXT(5, jack_function), - SOC_ENUM_SINGLE_EXT(2, spk_function), -}; - -static const struct snd_kcontrol_new tosa_controls[] = { - SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack, - tosa_set_jack), - SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk, - tosa_set_spk), -}; - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link tosa_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .ops = &tosa_ops, - SND_SOC_DAILINK_REG(ac97), -}, -{ - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - .ops = &tosa_ops, - SND_SOC_DAILINK_REG(ac97_aux), -}, -}; - -static struct snd_soc_card tosa = { - .name = "Tosa", - .owner = THIS_MODULE, - .dai_link = tosa_dai, - .num_links = ARRAY_SIZE(tosa_dai), - - .controls = tosa_controls, - .num_controls = ARRAY_SIZE(tosa_controls), - .dapm_widgets = tosa_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static int tosa_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = ⤩ - int ret; - - ret = gpio_request_one(TOSA_GPIO_L_MUTE, GPIOF_OUT_INIT_LOW, - "Headphone Jack"); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - gpio_free(TOSA_GPIO_L_MUTE); - } - return ret; -} - -static int tosa_remove(struct platform_device *pdev) -{ - gpio_free(TOSA_GPIO_L_MUTE); - return 0; -} - -static struct platform_driver tosa_driver = { - .driver = { - .name = "tosa-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = tosa_probe, - .remove = tosa_remove, -}; - -module_platform_driver(tosa_driver); - -/* Module information */ -MODULE_AUTHOR("Richard Purdie"); -MODULE_DESCRIPTION("ALSA SoC Tosa"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:tosa-audio"); diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c deleted file mode 100644 index d5f2961b1a3e..000000000000 --- a/sound/soc/pxa/ttc-dkb.c +++ /dev/null @@ -1,141 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * linux/sound/soc/pxa/ttc_dkb.c - * - * Copyright (C) 2012 Marvell International Ltd. - */ -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> -#include <asm/mach-types.h> -#include <sound/pcm_params.h> -#include "../codecs/88pm860x-codec.h" - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* ttc machine dapm widgets */ -static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic 2", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* ttc machine audio map */ -static const struct snd_soc_dapm_route ttc_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; - - /* Headset jack detection */ - snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); - snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack, mic_jack_pins, - ARRAY_SIZE(mic_jack_pins)); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(component, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(component, &hs_jack, SND_JACK_MICROPHONE); - - return 0; -} - -/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(i2s, - DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.1")), - DAILINK_COMP_ARRAY(COMP_CODEC("88pm860x-codec", "88pm860x-i2s")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("mmp-pcm-audio"))); - -static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { -{ - .name = "88pm860x i2s", - .stream_name = "audio playback", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = ttc_pm860x_init, - SND_SOC_DAILINK_REG(i2s), -}, -}; - -/* ttc/td audio machine driver */ -static struct snd_soc_card ttc_dkb_card = { - .name = "ttc-dkb-hifi", - .owner = THIS_MODULE, - .dai_link = ttc_pm860x_hifi_dai, - .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), - - .dapm_widgets = ttc_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets), - .dapm_routes = ttc_audio_map, - .num_dapm_routes = ARRAY_SIZE(ttc_audio_map), -}; - -static int ttc_dkb_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &ttc_dkb_card; - int ret; - - card->dev = &pdev->dev; - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - - return ret; -} - -static struct platform_driver ttc_dkb_driver = { - .driver = { - .name = "ttc-dkb-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = ttc_dkb_probe, -}; - -module_platform_driver(ttc_dkb_driver); - -/* Module information */ -MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC TTC DKB"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ttc-dkb-audio"); diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c deleted file mode 100644 index edf2b9eec5b8..000000000000 --- a/sound/soc/pxa/z2.c +++ /dev/null @@ -1,219 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * linux/sound/soc/pxa/z2.c - * - * SoC Audio driver for Aeronix Zipit Z2 - * - * Copyright (C) 2009 Ken McGuire <kenm@desertweyr.com> - * Copyright (C) 2010 Marek Vasut <marek.vasut@gmail.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <mach/hardware.h> -#include <mach/audio.h> -#include <mach/z2.h> - -#include "../codecs/wm8750.h" -#include "pxa2xx-i2s.h" - -static struct snd_soc_card snd_soc_z2; - -static int z2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set the I2S system clock as input (unused) */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_jack hs_jack; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Ext Spk", - .mask = SND_JACK_HEADPHONE, - .invert = 1 - }, -}; - -/* Headset jack detection gpios */ -static struct snd_soc_jack_gpio hs_jack_gpios[] = { - { - .gpio = GPIO37_ZIPITZ2_HEADSET_DETECT, - .name = "hsdet-gpio", - .report = SND_JACK_HEADSET, - .debounce_time = 200, - .invert = 1, - }, -}; - -/* z2 machine dapm widgets */ -static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - - /* headset is a mic and mono headphone */ - SND_SOC_DAPM_HP("Headset Jack", NULL), -}; - -/* Z2 machine audio_map */ -static const struct snd_soc_dapm_route z2_audio_map[] = { - - /* headphone connected to LOUT1, ROUT1 */ - {"Headphone Jack", NULL, "LOUT1"}, - {"Headphone Jack", NULL, "ROUT1"}, - - /* ext speaker connected to LOUT2, ROUT2 */ - {"Ext Spk", NULL, "ROUT2"}, - {"Ext Spk", NULL, "LOUT2"}, - - /* mic is connected to R input 2 - with bias */ - {"RINPUT2", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Mic Jack"}, -}; - -/* - * Logic for a wm8750 as connected on a Z2 Device - */ -static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - /* Jack detection API stuff */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, - &hs_jack, hs_jack_pins, - ARRAY_SIZE(hs_jack_pins)); - if (ret) - goto err; - - ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - if (ret) - goto err; - - return 0; - -err: - return ret; -} - -static const struct snd_soc_ops z2_ops = { - .hw_params = z2_hw_params, -}; - -/* z2 digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(wm8750, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8750.0-001b", "wm8750-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link z2_dai = { - .name = "wm8750", - .stream_name = "WM8750", - .init = z2_wm8750_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &z2_ops, - SND_SOC_DAILINK_REG(wm8750), -}; - -/* z2 audio machine driver */ -static struct snd_soc_card snd_soc_z2 = { - .name = "Z2", - .owner = THIS_MODULE, - .dai_link = &z2_dai, - .num_links = 1, - - .dapm_widgets = wm8750_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), - .dapm_routes = z2_audio_map, - .num_dapm_routes = ARRAY_SIZE(z2_audio_map), - .fully_routed = true, -}; - -static struct platform_device *z2_snd_device; - -static int __init z2_init(void) -{ - int ret; - - if (!machine_is_zipit2()) - return -ENODEV; - - z2_snd_device = platform_device_alloc("soc-audio", -1); - if (!z2_snd_device) - return -ENOMEM; - - platform_set_drvdata(z2_snd_device, &snd_soc_z2); - ret = platform_device_add(z2_snd_device); - - if (ret) - platform_device_put(z2_snd_device); - - return ret; -} - -static void __exit z2_exit(void) -{ - platform_device_unregister(z2_snd_device); -} - -module_init(z2_init); -module_exit(z2_exit); - -MODULE_AUTHOR("Ken McGuire <kenm@desertweyr.com>, " - "Marek Vasut <marek.vasut@gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC ZipitZ2"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c deleted file mode 100644 index bb89a53f4ab1..000000000000 --- a/sound/soc/pxa/zylonite.c +++ /dev/null @@ -1,266 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * zylonite.c -- SoC audio for Zylonite - * - * Copyright 2008 Wolfson Microelectronics PLC. - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "../codecs/wm9713.h" -#include "pxa-ssp.h" - -/* - * There is a physical switch SW15 on the board which changes the MCLK - * for the WM9713 between the standard AC97 master clock and the - * output of the CLK_POUT signal from the PXA. - */ -static int clk_pout; -module_param(clk_pout, int, 0); -MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board)."); - -static struct clk *pout; - -static struct snd_soc_card zylonite; - -static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Microphone", NULL), - SND_SOC_DAPM_MIC("Handset Microphone", NULL), - SND_SOC_DAPM_SPK("Multiactor", NULL), - SND_SOC_DAPM_SPK("Headset Earpiece", NULL), -}; - -/* Currently supported audio map */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* Headphone output connected to HPL/HPR */ - { "Headphone", NULL, "HPL" }, - { "Headphone", NULL, "HPR" }, - - /* On-board earpiece */ - { "Headset Earpiece", NULL, "OUT3" }, - - /* Headphone mic */ - { "MIC2A", NULL, "Mic Bias" }, - { "Mic Bias", NULL, "Headset Microphone" }, - - /* On-board mic */ - { "MIC1", NULL, "Mic Bias" }, - { "Mic Bias", NULL, "Handset Microphone" }, - - /* Multiactor differentially connected over SPKL/SPKR */ - { "Multiactor", NULL, "SPKL" }, - { "Multiactor", NULL, "SPKR" }, -}; - -static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) -{ - if (clk_pout) - snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, - clk_get_rate(pout), 0); - - return 0; -} - -static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - unsigned int wm9713_div = 0; - int ret = 0; - int rate = params_rate(params); - - /* Only support ratios that we can generate neatly from the AC97 - * based master clock - in particular, this excludes 44.1kHz. - * In most applications the voice DAC will be used for telephony - * data so multiples of 8kHz will be the common case. - */ - switch (rate) { - case 8000: - wm9713_div = 12; - break; - case 16000: - wm9713_div = 6; - break; - case 48000: - wm9713_div = 2; - break; - default: - /* Don't support OSS emulation */ - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); - if (ret < 0) - return ret; - - if (clk_pout) - ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, - WM9713_PCMDIV(wm9713_div)); - else - ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, - WM9713_PCMDIV(wm9713_div)); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops zylonite_voice_ops = { - .hw_params = zylonite_voice_hw_params, -}; - -SND_SOC_DAILINK_DEFS(ac97, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(ac97_aux, - DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-aux")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -SND_SOC_DAILINK_DEFS(voice, - DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.2")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-voice")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio"))); - -static struct snd_soc_dai_link zylonite_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .init = zylonite_wm9713_init, - SND_SOC_DAILINK_REG(ac97), -}, -{ - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - SND_SOC_DAILINK_REG(ac97_aux), -}, -{ - .name = "WM9713 Voice", - .stream_name = "WM9713 Voice", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &zylonite_voice_ops, - SND_SOC_DAILINK_REG(voice), -}, -}; - -static int zylonite_probe(struct snd_soc_card *card) -{ - int ret; - - if (clk_pout) { - pout = clk_get(NULL, "CLK_POUT"); - if (IS_ERR(pout)) { - dev_err(card->dev, "Unable to obtain CLK_POUT: %ld\n", - PTR_ERR(pout)); - return PTR_ERR(pout); - } - - ret = clk_enable(pout); - if (ret != 0) { - dev_err(card->dev, "Unable to enable CLK_POUT: %d\n", - ret); - clk_put(pout); - return ret; - } - - dev_dbg(card->dev, "MCLK enabled at %luHz\n", - clk_get_rate(pout)); - } - - return 0; -} - -static int zylonite_remove(struct snd_soc_card *card) -{ - if (clk_pout) { - clk_disable(pout); - clk_put(pout); - } - - return 0; -} - -static int zylonite_suspend_post(struct snd_soc_card *card) -{ - if (clk_pout) - clk_disable(pout); - - return 0; -} - -static int zylonite_resume_pre(struct snd_soc_card *card) -{ - int ret = 0; - - if (clk_pout) { - ret = clk_enable(pout); - if (ret != 0) - dev_err(card->dev, "Unable to enable CLK_POUT: %d\n", - ret); - } - - return ret; -} - -static struct snd_soc_card zylonite = { - .name = "Zylonite", - .owner = THIS_MODULE, - .probe = &zylonite_probe, - .remove = &zylonite_remove, - .suspend_post = &zylonite_suspend_post, - .resume_pre = &zylonite_resume_pre, - .dai_link = zylonite_dai, - .num_links = ARRAY_SIZE(zylonite_dai), - - .dapm_widgets = zylonite_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *zylonite_snd_ac97_device; - -static int __init zylonite_init(void) -{ - int ret; - - zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!zylonite_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(zylonite_snd_ac97_device, &zylonite); - - ret = platform_device_add(zylonite_snd_ac97_device); - if (ret != 0) - platform_device_put(zylonite_snd_ac97_device); - - return ret; -} - -static void __exit zylonite_exit(void) -{ - platform_device_unregister(zylonite_snd_ac97_device); -} - -module_init(zylonite_init); -module_exit(zylonite_exit); - -MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite"); -MODULE_LICENSE("GPL"); |