diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2023-02-20 15:28:57 -0800 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2023-02-20 15:28:57 -0800 |
| commit | ff0c7e18629b8bd64681313a88ce55e182c9fee6 (patch) | |
| tree | 70efa650f6458514e1d7e5e3f72cc8f87fc5cf2c /sound | |
| parent | 5b0ed5964928b0aaf0d644c17c886c7f5ea4bb3f (diff) | |
| parent | a1f925bc4fa899b3c0f2dcbc432d572c36e74e71 (diff) | |
Merge tag 'arm-boardfile-remove-6.3' of git://git.kernel.org/pub/scm/linux/kernel/git/soc/soc
Pull ARM SoC boardfile updates from Arnd Bergmann
"Unused boardfile removal for 6.3
This is a follow-up to the deprecation of most of the old-style board
files that was merged in linux-6.0, removing them for good.
This branch is almost exclusively dead code removal based on those
annotations. Some device driver removals went through separate
subsystem trees, but the majority is in the same branch, in order to
better handle dependencies between the patches and avoid breaking
bisection.
Unfortunately that leads to merge conflicts against other changes in
the subsystem trees, but they should all be trivial to resolve by
removing the files.
See commit 7d0d3fa7339e ("Merge tag 'arm-boardfiles-6.0' of
git://git.kernel.org/pub/scm/linux/kernel/git/soc/soc") for the
description of which machines were marked unused and are now removed.
The only removals that got postponed are Terastation WXL (mv78xx0) and
Jornada720 (StrongARM1100), which turned out to still have potential
users"
* tag 'arm-boardfile-remove-6.3' of git://git.kernel.org/pub/scm/linux/kernel/git/soc/soc: (91 commits)
mmc: omap: drop TPS65010 dependency
ARM: pxa: restore mfp-pxa320.h
usb: ohci-omap: avoid unused-variable warning
ARM: debug: remove references in DEBUG_UART_8250_SHIFT to removed configs
ARM: s3c: remove obsolete s3c-cpu-freq header
MAINTAINERS: adjust SAMSUNG SOC CLOCK DRIVERS after s3c24xx support removal
MAINTAINERS: update file entries after arm multi-platform rework and mach-pxa removal
ARM: remove CONFIG_UNUSED_BOARD_FILES
mfd: remove htc-pasic3 driver
w1: remove ds1wm driver
usb: remove ohci-tmio driver
fbdev: remove w100fb driver
fbdev: remove tmiofb driver
mmc: remove tmio_mmc driver
mfd: remove ucb1400 support
mfd: remove toshiba tmio drivers
rtc: remove v3020 driver
power: remove pda_power supply driver
ASoC: pxa: remove unused board support
pcmcia: remove unused pxa/sa1100 drivers
...
Diffstat (limited to 'sound')
51 files changed, 3 insertions, 9001 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index e56d96d2b11c..0ddfb717b81d 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -107,7 +107,6 @@ endif # !UML endif # SOUND -# AC97_BUS is used from both sound and ucb1400 config AC97_BUS tristate help diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index ff685321f1a1..9afc5906d662 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -152,7 +152,6 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, // only guess --jk { 0x4e534331, 0xffffffff, "LM4549", NULL, NULL }, { 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix -{ 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x53544d02, 0xffffffff, "ST7597", NULL, NULL }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 025c1666c1fc..4b5f33de70d5 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3937,43 +3937,3 @@ static int patch_lm4550(struct snd_ac97 *ac97) ac97->res_table = lm4550_restbl; return 0; } - -/* - * UCB1400 codec (http://www.semiconductors.philips.com/acrobat_download/datasheets/UCB1400-02.pdf) - */ -static const struct snd_kcontrol_new snd_ac97_controls_ucb1400[] = { -/* enable/disable headphone driver which allows direct connection to - stereo headphone without the use of external DC blocking - capacitors */ -AC97_SINGLE("Headphone Driver", 0x6a, 6, 1, 0), -/* Filter used to compensate the DC offset is added in the ADC to remove idle - tones from the audio band. */ -AC97_SINGLE("DC Filter", 0x6a, 4, 1, 0), -/* Control smart-low-power mode feature. Allows automatic power down - of unused blocks in the ADC analog front end and the PLL. */ -AC97_SINGLE("Smart Low Power Mode", 0x6c, 4, 3, 0), -}; - -static int patch_ucb1400_specific(struct snd_ac97 * ac97) -{ - int idx, err; - for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_ucb1400); idx++) { - err = snd_ctl_add(ac97->bus->card, snd_ctl_new1(&snd_ac97_controls_ucb1400[idx], ac97)); - if (err < 0) - return err; - } - return 0; -} - -static const struct snd_ac97_build_ops patch_ucb1400_ops = { - .build_specific = patch_ucb1400_specific, -}; - -static int patch_ucb1400(struct snd_ac97 * ac97) -{ - ac97->build_ops = &patch_ucb1400_ops; - /* enable headphone driver and smart low power mode by default */ - snd_ac97_write_cache(ac97, 0x6a, 0x0050); - snd_ac97_write_cache(ac97, 0x6c, 0x0030); - return 0; -} diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 8039a8febefa..34870c2d0cba 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -27,29 +27,6 @@ config SND_EP93XX_SOC_I2S_WATCHDOG endif # if SND_EP93XX_SOC_I2S -config SND_EP93XX_SOC_AC97 - tristate - select AC97_BUS - select SND_SOC_AC97_BUS - -config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. - -config SND_EP93XX_SOC_SIMONE - tristate "SoC Audio support for Simplemachines Sim.One board" - depends on SND_EP93XX_SOC && MACH_SIM_ONE - select SND_EP93XX_SOC_AC97 - select SND_SOC_AC97_CODEC - help - Say Y or M here if you want to add support for AC97 audio on the - Simplemachines Sim.One board. - config SND_EP93XX_SOC_EDB93XX tristate "SoC Audio support for Cirrus Logic EDB93xx boards" depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) diff --git a/sound/soc/cirrus/Makefile b/sound/soc/cirrus/Makefile index bfb8dc409f53..19a86daad660 100644 --- a/sound/soc/cirrus/Makefile +++ b/sound/soc/cirrus/Makefile @@ -2,17 +2,11 @@ # EP93xx Platform Support snd-soc-ep93xx-objs := ep93xx-pcm.o snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o -snd-soc-ep93xx-ac97-objs := ep93xx-ac97.o obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o -obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support -snd-soc-snappercl15-objs := snappercl15.o -snd-soc-simone-objs := simone.o snd-soc-edb93xx-objs := edb93xx.o -obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o -obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c deleted file mode 100644 index 37593abe6053..000000000000 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ /dev/null @@ -1,446 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * ASoC driver for Cirrus Logic EP93xx AC97 controller. - * - * Copyright (c) 2010 Mika Westerberg - * - * Based on s3c-ac97 ASoC driver by Jaswinder Singh. - */ - -#include <linux/delay.h> -#include <linux/err.h> -#include <linux/io.h> -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/slab.h> - -#include <sound/core.h> -#include <sound/dmaengine_pcm.h> -#include <sound/ac97_codec.h> -#include <sound/soc.h> - -#include <linux/platform_data/dma-ep93xx.h> -#include <linux/soc/cirrus/ep93xx.h> - -#include "ep93xx-pcm.h" - -/* - * Per channel (1-4) registers. - */ -#define AC97CH(n) (((n) - 1) * 0x20) - -#define AC97DR(n) (AC97CH(n) + 0x0000) - -#define AC97RXCR(n) (AC97CH(n) + 0x0004) -#define AC97RXCR_REN BIT(0) -#define AC97RXCR_RX3 BIT(3) -#define AC97RXCR_RX4 BIT(4) -#define AC97RXCR_CM BIT(15) - -#define AC97TXCR(n) (AC97CH(n) + 0x0008) -#define AC97TXCR_TEN BIT(0) -#define AC97TXCR_TX3 BIT(3) -#define AC97TXCR_TX4 BIT(4) -#define AC97TXCR_CM BIT(15) - -#define AC97SR(n) (AC97CH(n) + 0x000c) -#define AC97SR_TXFE BIT(1) -#define AC97SR_TXUE BIT(6) - -#define AC97RISR(n) (AC97CH(n) + 0x0010) -#define AC97ISR(n) (AC97CH(n) + 0x0014) -#define AC97IE(n) (AC97CH(n) + 0x0018) - -/* - * Global AC97 controller registers. - */ -#define AC97S1DATA 0x0080 -#define AC97S2DATA 0x0084 -#define AC97S12DATA 0x0088 - -#define AC97RGIS 0x008c -#define AC97GIS 0x0090 -#define AC97IM 0x0094 -/* - * Common bits for RGIS, GIS and IM registers. - */ -#define AC97_SLOT2RXVALID BIT(1) -#define AC97_CODECREADY BIT(5) -#define AC97_SLOT2TXCOMPLETE BIT(6) - -#define AC97EOI 0x0098 -#define AC97EOI_WINT BIT(0) -#define AC97EOI_CODECREADY BIT(1) - -#define AC97GCR 0x009c -#define AC97GCR_AC97IFE BIT(0) - -#define AC97RESET 0x00a0 -#define AC97RESET_TIMEDRESET BIT(0) - -#define AC97SYNC 0x00a4 -#define AC97SYNC_TIMEDSYNC BIT(0) - -#define AC97_TIMEOUT msecs_to_jiffies(5) - -/** - * struct ep93xx_ac97_info - EP93xx AC97 controller info structure - * @lock: mutex serializing access to the bus (slot 1 & 2 ops) - * @dev: pointer to the platform device dev structure - * @regs: mapped AC97 controller registers - * @done: bus ops wait here for an interrupt - */ -struct ep93xx_ac97_info { - struct mutex lock; - struct device *dev; - void __iomem *regs; - struct completion done; - struct snd_dmaengine_dai_dma_data dma_params_rx; - struct snd_dmaengine_dai_dma_data dma_params_tx; -}; - -/* currently ALSA only supports a single AC97 device */ -static struct ep93xx_ac97_info *ep93xx_ac97_info; - -static struct ep93xx_dma_data ep93xx_ac97_pcm_out = { - .name = "ac97-pcm-out", - .port = EP93XX_DMA_AAC1, - .direction = DMA_MEM_TO_DEV, -}; - -static struct ep93xx_dma_data ep93xx_ac97_pcm_in = { - .name = "ac97-pcm-in", - .port = EP93XX_DMA_AAC1, - .direction = DMA_DEV_TO_MEM, -}; - -static inline unsigned ep93xx_ac97_read_reg(struct ep93xx_ac97_info *info, - unsigned reg) -{ - return __raw_readl(info->regs + reg); -} - -static inline void ep93xx_ac97_write_reg(struct ep93xx_ac97_info *info, - unsigned reg, unsigned val) -{ - __raw_writel(val, info->regs + reg); -} - -static unsigned short ep93xx_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - unsigned short val; - - mutex_lock(&info->lock); - - ep93xx_ac97_write_reg(info, AC97S1DATA, reg); - ep93xx_ac97_write_reg(info, AC97IM, AC97_SLOT2RXVALID); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) { - dev_warn(info->dev, "timeout reading register %x\n", reg); - mutex_unlock(&info->lock); - return -ETIMEDOUT; - } - val = (unsigned short)ep93xx_ac97_read_reg(info, AC97S2DATA); - - mutex_unlock(&info->lock); - return val; -} - -static void ep93xx_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, - unsigned short val) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - - mutex_lock(&info->lock); - - /* - * Writes to the codec need to be done so that slot 2 is filled in - * before slot 1. - */ - ep93xx_ac97_write_reg(info, AC97S2DATA, val); - ep93xx_ac97_write_reg(info, AC97S1DATA, reg); - - ep93xx_ac97_write_reg(info, AC97IM, AC97_SLOT2TXCOMPLETE); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) - dev_warn(info->dev, "timeout writing register %x\n", reg); - - mutex_unlock(&info->lock); -} - -static void ep93xx_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - - mutex_lock(&info->lock); - - /* - * We are assuming that before this functions gets called, the codec - * BIT_CLK is stopped by forcing the codec into powerdown mode. We can - * control the SYNC signal directly via AC97SYNC register. Using - * TIMEDSYNC the controller will keep the SYNC high > 1us. - */ - ep93xx_ac97_write_reg(info, AC97SYNC, AC97SYNC_TIMEDSYNC); - ep93xx_ac97_write_reg(info, AC97IM, AC97_CODECREADY); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) - dev_warn(info->dev, "codec warm reset timeout\n"); - - mutex_unlock(&info->lock); -} - -static void ep93xx_ac97_cold_reset(struct snd_ac97 *ac97) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - - mutex_lock(&info->lock); - - /* - * For doing cold reset, we disable the AC97 controller interface, clear - * WINT and CODECREADY bits, and finally enable the interface again. - */ - ep93xx_ac97_write_reg(info, AC97GCR, 0); - ep93xx_ac97_write_reg(info, AC97EOI, AC97EOI_CODECREADY | AC97EOI_WINT); - ep93xx_ac97_write_reg(info, AC97GCR, AC97GCR_AC97IFE); - - /* - * Now, assert the reset and wait for the codec to become ready. - */ - ep93xx_ac97_write_reg(info, AC97RESET, AC97RESET_TIMEDRESET); - ep93xx_ac97_write_reg(info, AC97IM, AC97_CODECREADY); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) - dev_warn(info->dev, "codec cold reset timeout\n"); - - /* - * Give the codec some time to come fully out from the reset. This way - * we ensure that the subsequent reads/writes will work. - */ - usleep_range(15000, 20000); - - mutex_unlock(&info->lock); -} - -static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) -{ - struct ep93xx_ac97_info *info = dev_id; - unsigned status, mask; - - /* - * Just mask out the interrupt and wake up the waiting thread. - * Interrupts are cleared via reading/writing to slot 1 & 2 registers by - * the waiting thread. - */ - status = ep93xx_ac97_read_reg(info, AC97GIS); - mask = ep93xx_ac97_read_reg(info, AC97IM); - mask &= ~status; - ep93xx_ac97_write_reg(info, AC97IM, mask); - - complete(&info->done); - return IRQ_HANDLED; -} - -static struct snd_ac97_bus_ops ep93xx_ac97_ops = { - .read = ep93xx_ac97_read, - .write = ep93xx_ac97_write, - .reset = ep93xx_ac97_cold_reset, - .warm_reset = ep93xx_ac97_warm_reset, -}; - -static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); - unsigned v = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* - * Enable compact mode, TX slots 3 & 4, and the TX FIFO - * itself. - */ - v |= AC97TXCR_CM; - v |= AC97TXCR_TX3 | AC97TXCR_TX4; - v |= AC97TXCR_TEN; - ep93xx_ac97_write_reg(info, AC97TXCR(1), v); - } else { - /* - * Enable compact mode, RX slots 3 & 4, and the RX FIFO - * itself. - */ - v |= AC97RXCR_CM; - v |= AC97RXCR_RX3 | AC97RXCR_RX4; - v |= AC97RXCR_REN; - ep93xx_ac97_write_reg(info, AC97RXCR(1), v); - } - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* - * As per Cirrus EP93xx errata described below: - * - * https://www.cirrus.com/en/pubs/errata/ER667E2B.pdf - * - * we will wait for the TX FIFO to be empty before - * clearing the TEN bit. - */ - unsigned long timeout = jiffies + AC97_TIMEOUT; - - do { - v = ep93xx_ac97_read_reg(info, AC97SR(1)); - if (time_after(jiffies, timeout)) { - dev_warn(info->dev, "TX timeout\n"); - break; - } - } while (!(v & (AC97SR_TXFE | AC97SR_TXUE))); - - /* disable the TX FIFO */ - ep93xx_ac97_write_reg(info, AC97TXCR(1), 0); - } else { - /* disable the RX FIFO */ - ep93xx_ac97_write_reg(info, AC97RXCR(1), 0); - } - break; - - default: - dev_warn(info->dev, "unknown command %d\n", cmd); - return -EINVAL; - } - - return 0; -} - -static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai) -{ - struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); - - info->dma_params_tx.filter_data = &ep93xx_ac97_pcm_out; - info->dma_params_rx.filter_data = &ep93xx_ac97_pcm_in; - - dai->playback_dma_data = &info->dma_params_tx; - dai->capture_dma_data = &info->dma_params_rx; - - return 0; -} - -static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { - .trigger = ep93xx_ac97_trigger, -}; - -static struct snd_soc_dai_driver ep93xx_ac97_dai = { - .name = "ep93xx-ac97", - .id = 0, - .probe = ep93xx_ac97_dai_probe, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &ep93xx_ac97_dai_ops, -}; - -static const struct snd_soc_component_driver ep93xx_ac97_component = { - .name = "ep93xx-ac97", - .legacy_dai_naming = 1, -}; - -static int ep93xx_ac97_probe(struct platform_device *pdev) -{ - struct ep93xx_ac97_info *info; - int irq; - int ret; - - info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); - if (!info) - return -ENOMEM; - - info->regs = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(info->regs)) - return PTR_ERR(info->regs); - - irq = platform_get_irq(pdev, 0); - if (irq <= 0) - return irq < 0 ? irq : -ENODEV; - - ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, - IRQF_TRIGGER_HIGH, pdev->name, info); - if (ret) - goto fail; - - dev_set_drvdata(&pdev->dev, info); - - mutex_init(&info->lock); - init_completion(&info->done); - info->dev = &pdev->dev; - - ep93xx_ac97_info = info; - platform_set_drvdata(pdev, info); - - ret = snd_soc_set_ac97_ops(&ep93xx_ac97_ops); - if (ret) - goto fail; - - ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component, - &ep93xx_ac97_dai, 1); - if (ret) - goto fail; - - ret = devm_ep93xx_pcm_platform_register(&pdev->dev); - if (ret) - goto fail_unregister; - - return 0; - -fail_unregister: - snd_soc_unregister_component(&pdev->dev); -fail: - ep93xx_ac97_info = NULL; - snd_soc_set_ac97_ops(NULL); - return ret; -} - -static int ep93xx_ac97_remove(struct platform_device *pdev) -{ - struct ep93xx_ac97_info *info = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - - /* disable the AC97 controller */ - ep93xx_ac97_write_reg(info, AC97GCR, 0); - - ep93xx_ac97_info = NULL; - - snd_soc_set_ac97_ops(NULL); - - return 0; -} - -static struct platform_driver ep93xx_ac97_driver = { - .probe = ep93xx_ac97_probe, - .remove = ep93xx_ac97_remove, - .driver = { - .name = "ep93xx-ac97", - }, -}; - -module_platform_driver(ep93xx_ac97_driver); - -MODULE_DESCRIPTION("EP93xx AC97 ASoC Driver"); -MODULE_AUTHOR("Mika Westerberg <mika.westerberg@iki.fi>"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-ac97"); diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c deleted file mode 100644 index 801c90877d77..000000000000 --- a/sound/soc/cirrus/simone.c +++ /dev/null @@ -1,86 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * simone.c -- ASoC audio for Simplemachines Sim.One board - * - * Copyright (c) 2010 Mika Westerberg - * - * Based on snappercl15 machine driver by Ryan Mallon. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/soc/cirrus/ep93xx.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_CPU("ep93xx-ac97")), - DAILINK_COMP_ARRAY(COMP_CODEC("ac97-codec", "ac97-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("ep93xx-ac97"))); - -static struct snd_soc_dai_link simone_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - SND_SOC_DAILINK_REG(hifi), -}; - -static struct snd_soc_card snd_soc_simone = { - .name = "Sim.One", - .owner = THIS_MODULE, - .dai_link = &simone_dai, - .num_links = 1, -}; - -static struct platform_device *simone_snd_ac97_device; - -static int simone_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_simone; - int ret; - - simone_snd_ac97_device = platform_device_register_simple("ac97-codec", - -1, NULL, 0); - if (IS_ERR(simone_snd_ac97_device)) - return PTR_ERR(simone_snd_ac97_device); - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - platform_device_unregister(simone_snd_ac97_device); - } - - return ret; -} - -static int simone_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - platform_device_unregister(simone_snd_ac97_device); - - return 0; -} - -static struct platform_driver simone_driver = { - .driver = { - .name = "simone-audio", - }, - .probe = simone_probe, - .remove = simone_remove, -}; - -module_platform_driver(simone_driver); - -MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); -MODULE_AUTHOR("Mika Westerberg <mika.westerberg@iki.fi>"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:simone-audio"); diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c deleted file mode 100644 index a286f5beeaeb..000000000000 --- a/sound/soc/cirrus/snappercl15.c +++ /dev/null @@ -1,134 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module - * - * Copyright (C) 2008 Bluewater Systems Ltd - * Author: Ryan Mallon - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <linux/soc/cirrus/ep93xx.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "../codecs/tlv320aic23.h" - -#define CODEC_CLOCK 5644800 - -static int snappercl15_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 err; - - err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, - SND_SOC_CLOCK_IN); - if (err) - return err; - - err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK, - SND_SOC_CLOCK_OUT); - if (err) - return err; - - return 0; -} - -static const struct snd_soc_ops snappercl15_ops = { - .hw_params = snappercl15_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - -SND_SOC_DAILINK_DEFS(aic23, - DAILINK_COMP_ARRAY(COMP_CPU("ep93xx-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic23-codec.0-001a", - "tlv320aic23-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("ep93xx-i2s"))); - -static struct snd_soc_dai_link snappercl15_dai = { - .name = "tlv320aic23", - .stream_name = "AIC23", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ops = &snappercl15_ops, - SND_SOC_DAILINK_REG(aic23), -}; - -static struct snd_soc_card snd_soc_snappercl15 = { - .name = "Snapper CL15", - .owner = THIS_MODULE, - .dai_link = &snappercl15_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static int snappercl15_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_snappercl15; - int ret; - - ret = ep93xx_i2s_acquire(); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - ep93xx_i2s_release(); - } - - return ret; -} - -static int snappercl15_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - ep93xx_i2s_release(); - - return 0; -} - -static struct platform_driver snappercl15_driver = { - .driver = { - .name = "snappercl15-audio", - }, - .probe = snappercl15_probe, - .remove = snappercl15_remove, -}; - -module_platform_driver(snappercl15_driver); - -MODULE_AUTHOR("Ryan Mallon"); -MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:snappercl15-audio"); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index a045693d5bc2..c26d1b36e8f7 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,10 +8,6 @@ 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 @@ -41,15 +37,6 @@ config SND_MMP_SOC_SSPA 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_I2C - 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 +46,6 @@ 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_I2C - 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 @@ -161,71 +53,3 @@ config SND_PXA910_SOC help 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_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. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index b712eb894a61..406605fc7414 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -4,47 +4,14 @@ 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 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-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 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-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o -obj-$(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 4489d2c8b124..000000000000 --- a/sound/soc/pxa/corgi.c +++ /dev/null @@ -1,332 +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 <linux/platform_data/asoc-pxa.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 struct gpio_desc *gpiod_mute_l, *gpiod_mute_r, - *gpiod_apm_on, *gpiod_mic_bias; - -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 */ - gpiod_set_value(gpiod_mute_l, 1); - gpiod_set_value(gpiod_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 */ - gpiod_set_value(gpiod_mute_l, 0); - gpiod_set_value(gpiod_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: - gpiod_set_value(gpiod_mute_l, 0); - gpiod_set_value(gpiod_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: - gpiod_set_value(gpiod_mute_l, 0); - gpiod_set_value(gpiod_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 */ - gpiod_set_value(gpiod_mute_l, 1); - gpiod_set_value(gpiod_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) -{ - gpiod_set_value(gpiod_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) -{ - gpiod_set_value(gpiod_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; - - gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_HIGH); - if (IS_ERR(gpiod_mute_l)) - return PTR_ERR(gpiod_mute_l); - gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_HIGH); - if (IS_ERR(gpiod_mute_r)) - return PTR_ERR(gpiod_mute_r); - gpiod_apm_on = devm_gpiod_get(&pdev->dev, "apm-on", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_apm_on)) - return PTR_ERR(gpiod_apm_on); - gpiod_mic_bias = devm_gpiod_get(&pdev->dev, "mic-bias", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_mic_bias)) - return PTR_ERR(gpiod_mic_bias); - - 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 4e0e9b778d4c..000000000000 --- a/sound/soc/pxa/e740_wm9705.c +++ /dev/null @@ -1,168 +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/consumer.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <linux/platform_data/asoc-pxa.h> - -#include <asm/mach-types.h> - -static struct gpio_desc *gpiod_output_amp, *gpiod_input_amp; -static struct gpio_desc *gpiod_audio_power; - -#define E740_AUDIO_OUT 1 -#define E740_AUDIO_IN 2 - -static int e740_audio_power; - -static void e740_sync_audio_power(int status) -{ - gpiod_set_value(gpiod_audio_power, !status); - gpiod_set_value(gpiod_output_amp, (status & E740_AUDIO_OUT) ? 1 : 0); - gpiod_set_value(gpiod_input_amp, (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 int e740_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &e740; - int ret; - - gpiod_input_amp = devm_gpiod_get(&pdev->dev, "Mic amp", GPIOD_OUT_LOW); - ret = PTR_ERR_OR_ZERO(gpiod_input_amp); - if (ret) - return ret; - gpiod_output_amp = devm_gpiod_get(&pdev->dev, "Output amp", GPIOD_OUT_LOW); - ret = PTR_ERR_OR_ZERO(gpiod_output_amp); - if (ret) - return ret; - gpiod_audio_power = devm_gpiod_get(&pdev->dev, "Audio power", GPIOD_OUT_HIGH); - ret = PTR_ERR_OR_ZERO(gpiod_audio_power); - 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); - return ret; -} - -static int e740_remove(struct platform_device *pdev) -{ - 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 7a1e0d8bfd11..000000000000 --- a/sound/soc/pxa/e750_wm9705.c +++ /dev/null @@ -1,147 +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/consumer.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <linux/platform_data/asoc-pxa.h> - -#include <asm/mach-types.h> - -static struct gpio_desc *gpiod_spk_amp, *gpiod_hp_amp; - -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) - gpiod_set_value(gpiod_spk_amp, 1); - else if (event & SND_SOC_DAPM_POST_PMD) - gpiod_set_value(gpiod_spk_amp, 0); - - 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) - gpiod_set_value(gpiod_hp_amp, 1); - else if (event & SND_SOC_DAPM_POST_PMD) - gpiod_set_value(gpiod_hp_amp, 0); - - 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 int e750_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &e750; - int ret; - - gpiod_hp_amp = devm_gpiod_get(&pdev->dev, "Headphone amp", GPIOD_OUT_LOW); - ret = PTR_ERR_OR_ZERO(gpiod_hp_amp); - if (ret) - return ret; - gpiod_spk_amp = devm_gpiod_get(&pdev->dev, "Speaker amp", GPIOD_OUT_LOW); - ret = PTR_ERR_OR_ZERO(gpiod_spk_amp); - 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); - return ret; -} - -static int e750_remove(struct platform_device *pdev) -{ - 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 a39c494127cf..000000000000 --- a/sound/soc/pxa/e800_wm9712.c +++ /dev/null @@ -1,147 +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/consumer.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <linux/platform_data/asoc-pxa.h> - -static struct gpio_desc *gpiod_spk_amp, *gpiod_hp_amp; - -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) - gpiod_set_value(gpiod_spk_amp, 1); - else if (event & SND_SOC_DAPM_POST_PMD) - gpiod_set_value(gpiod_spk_amp, 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) - gpiod_set_value(gpiod_hp_amp, 1); - else if (event & SND_SOC_DAPM_POST_PMD) - gpiod_set_value(gpiod_hp_amp, 0); - - 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 int e800_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &e800; - int ret; - - gpiod_hp_amp = devm_gpiod_get(&pdev->dev, "Headphone amp", GPIOD_OUT_LOW); - ret = PTR_ERR_OR_ZERO(gpiod_hp_amp); - if (ret) - return ret; - gpiod_spk_amp = devm_gpiod_get(&pdev->dev, "Speaker amp", GPIOD_OUT_LOW); - ret = PTR_ERR_OR_ZERO(gpiod_spk_amp); - 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); - return ret; -} - -static int e800_remove(struct platform_device *pdev) -{ - 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 b59ec22e1e7e..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 <linux/platform_data/asoc-pxa.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 a323ddb8fc3e..000000000000 --- a/sound/soc/pxa/hx4700.c +++ /dev/null @@ -1,207 +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/consumer.h> - -#include <sound/core.h> -#include <sound/jack.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include "pxa2xx-i2s.h" - -static struct gpio_desc *gpiod_hp_driver, *gpiod_spk_sd; -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, - .invert = 1, - }, - { - .pin = "Speaker", - /* disable speaker when hp jack is inserted */ - .mask = SND_JACK_HEADPHONE, - }, -}; - -/* Headphones jack detection GPIO */ -static struct snd_soc_jack_gpio hs_jack_gpio = { - .name = "earphone-det", - .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) -{ - gpiod_set_value(gpiod_spk_sd, !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) -{ - gpiod_set_value(gpiod_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_pins(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 int hx4700_audio_probe(struct platform_device *pdev) -{ - int ret; - - if (!machine_is_h4700()) - return -ENODEV; - - gpiod_hp_driver = devm_gpiod_get(&pdev->dev, "hp-driver", GPIOD_ASIS); - ret = PTR_ERR_OR_ZERO(gpiod_hp_driver); - if (ret) - return ret; - gpiod_spk_sd = devm_gpiod_get(&pdev->dev, "spk-sd", GPIOD_ASIS); - ret = PTR_ERR_OR_ZERO(gpiod_spk_sd); - if (ret) - return ret; - - hs_jack_gpio.gpiod_dev = &pdev->dev; - snd_soc_card_hx4700.dev = &pdev->dev; - ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700); - - return ret; -} - -static int hx4700_audio_remove(struct platform_device *pdev) -{ - gpiod_set_value(gpiod_hp_driver, 0); - gpiod_set_value(gpiod_spk_sd, 0); - 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/magician.c b/sound/soc/pxa/magician.c deleted file mode 100644 index b791a2ba5ce5..000000000000 --- a/sound/soc/pxa/magician.c +++ /dev/null @@ -1,366 +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/consumer.h> -#include <linux/i2c.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.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 struct gpio_desc *gpiod_spk_power, *gpiod_ep_power, *gpiod_mic_power; -static struct gpio_desc *gpiod_in_sel0, *gpiod_in_sel1; - -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_BC_FC); - 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_BP_FP); - 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_BC_FC); - 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_BP_FP); - 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: - gpiod_set_value(gpiod_in_sel1, 1); - break; - case MAGICIAN_MIC_EXT: - gpiod_set_value(gpiod_in_sel1, 0); - } - - return 1; -} - -static int magician_spk_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpiod_set_value(gpiod_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) -{ - gpiod_set_value(gpiod_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) -{ - gpiod_set_value(gpiod_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 int magician_audio_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - - gpiod_spk_power = devm_gpiod_get(dev, "SPK_POWER", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_spk_power)) - return PTR_ERR(gpiod_spk_power); - gpiod_ep_power = devm_gpiod_get(dev, "EP_POWER", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_ep_power)) - return PTR_ERR(gpiod_ep_power); - gpiod_mic_power = devm_gpiod_get(dev, "MIC_POWER", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_mic_power)) - return PTR_ERR(gpiod_mic_power); - gpiod_in_sel0 = devm_gpiod_get(dev, "IN_SEL0", GPIOD_OUT_HIGH); - if (IS_ERR(gpiod_in_sel0)) - return PTR_ERR(gpiod_in_sel0); - gpiod_in_sel1 = devm_gpiod_get(dev, "IN_SEL1", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_in_sel1)) - return PTR_ERR(gpiod_in_sel1); - - snd_soc_card_magician.dev = &pdev->dev; - return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_magician); -} - -static struct platform_driver magician_audio_driver = { - .driver.name = "magician-audio", - .driver.pm = &snd_soc_pm_ops, - .probe = magician_audio_probe, -}; -module_platform_driver(magician_audio_driver); - -MODULE_AUTHOR("Philipp Zabel"); -MODULE_DESCRIPTION("ALSA SoC Magician"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:magician-audio"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c deleted file mode 100644 index 0fa37637eca9..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 <linux/platform_data/asoc-pxa.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 99b245e3079a..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 (devname && (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, 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/palm27x.c b/sound/soc/pxa/palm27x.c deleted file mode 100644 index a2321c01c160..000000000000 --- a/sound/soc/pxa/palm27x.c +++ /dev/null @@ -1,162 +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 <linux/platform_data/asoc-pxa.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_pins(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 5fdaa477e85d..000000000000 --- a/sound/soc/pxa/poodle.c +++ /dev/null @@ -1,291 +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 <linux/platform_data/asoc-pxa.h> -#include <linux/platform_data/asoc-poodle.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 struct poodle_audio_platform_data *poodle_pdata; - -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_pdata->locomo_dev, - poodle_pdata->gpio_mute_l, 1); - locomo_gpio_write(poodle_pdata->locomo_dev, - poodle_pdata->gpio_mute_r, 1); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - } else { - locomo_gpio_write(poodle_pdata->locomo_dev, - poodle_pdata->gpio_mute_l, 0); - locomo_gpio_write(poodle_pdata->locomo_dev, - poodle_pdata->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_pdata->locomo_dev, - poodle_pdata->gpio_mute_l, 1); - locomo_gpio_write(poodle_pdata->locomo_dev, - poodle_pdata->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_pdata->locomo_dev, - poodle_pdata->gpio_amp_on, 0); - else - locomo_gpio_write(poodle_pdata->locomo_dev, - poodle_pdata->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; - - poodle_pdata = pdev->dev.platform_data; - locomo_gpio_set_dir(poodle_pdata->locomo_dev, - poodle_pdata->gpio_amp_on, 0); - /* should we mute HP at startup - burning power ?*/ - locomo_gpio_set_dir(poodle_pdata->locomo_dev, - poodle_pdata->gpio_mute_l, 0); - locomo_gpio_set_dir(poodle_pdata->locomo_dev, - poodle_pdata->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/tosa.c b/sound/soc/pxa/tosa.c deleted file mode 100644 index 30f83cab0c32..000000000000 --- a/sound/soc/pxa/tosa.c +++ /dev/null @@ -1,255 +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/consumer.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <linux/platform_data/asoc-pxa.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 struct gpio_desc *tosa_mute; -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) -{ - gpiod_set_value(tosa_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; - - tosa_mute = devm_gpiod_get(&pdev->dev, NULL, GPIOD_OUT_LOW); - if (IS_ERR(tosa_mute)) - return dev_err_probe(&pdev->dev, PTR_ERR(tosa_mute), - "failed to get L_MUTE GPIO\n"); - gpiod_set_consumer_name(tosa_mute, "Headphone Jack"); - - 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 tosa_driver = { - .driver = { - .name = "tosa-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = tosa_probe, -}; - -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 6cc970bb2aac..000000000000 --- a/sound/soc/pxa/ttc-dkb.c +++ /dev/null @@ -1,143 +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_pins(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_pins(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 020dcce1df1f..000000000000 --- a/sound/soc/pxa/z2.c +++ /dev/null @@ -1,218 +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/consumer.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <linux/platform_data/asoc-pxa.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[] = { - { - .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_pins(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; - - hs_jack_gpios[0].gpiod_dev = &z2_snd_device->dev; - 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"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 2a61e620cd3b..93c2b1b08d0a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -11,16 +11,6 @@ menuconfig SND_SOC_SAMSUNG if SND_SOC_SAMSUNG -config SND_S3C24XX_I2S - tristate - -config SND_S3C_I2SV2_SOC - tristate - -config SND_S3C2412_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - config SND_SAMSUNG_PCM tristate "Samsung PCM interface support" @@ -31,35 +21,6 @@ config SND_SAMSUNG_SPDIF config SND_SAMSUNG_I2S tristate "Samsung I2S interface support" -config SND_SOC_SAMSUNG_NEO1973_WM8753 - tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" - depends on MACH_NEO1973_GTA02 || COMPILE_TEST - depends on SND_SOC_I2C_AND_SPI - select SND_S3C24XX_I2S - select SND_SOC_WM8753 - select SND_SOC_BT_SCO - help - Say Y here to enable audio support for the Openmoko Neo1973 - Smartphones. - -config SND_SOC_SAMSUNG_JIVE_WM8750 - tristate "SoC I2S Audio support for Jive" - depends on MACH_JIVE && I2C || COMPILE_TEST && ARM - depends on SND_SOC_I2C_AND_SPI - select SND_SOC_WM8750 - select SND_S3C2412_SOC_I2S - help - Say Y if you want to add support for SoC audio on the Jive. - -config SND_SOC_SAMSUNG_SMDK_WM8580 - tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on MACH_SMDK6410 || COMPILE_TEST - depends on I2C - select SND_SOC_WM8580 - select SND_SAMSUNG_I2S - help - Say Y if you want to add support for SoC audio on the SMDKs. - config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" depends on I2C=y @@ -69,60 +30,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 help Say Y if you want to add support for SoC audio on the SMDKs. -config SND_SOC_SAMSUNG_S3C24XX_UDA134X - tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on ARCH_S3C24XX || COMPILE_TEST - select SND_S3C24XX_I2S - select SND_SOC_L3 - select SND_SOC_UDA134X - -config SND_SOC_SAMSUNG_SIMTEC - tristate - help - Internal node for common S3C24XX/Simtec support. - -config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23 - tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on ARCH_S3C24XX || COMPILE_TEST - depends on I2C - select SND_S3C24XX_I2S - select SND_SOC_TLV320AIC23_I2C - select SND_SOC_SAMSUNG_SIMTEC - -config SND_SOC_SAMSUNG_SIMTEC_HERMES - tristate "SoC I2S Audio support for Simtec Hermes board" - depends on ARCH_S3C24XX || COMPILE_TEST - depends on I2C - select SND_S3C24XX_I2S - select SND_SOC_TLV320AIC3X - select SND_SOC_SAMSUNG_SIMTEC - -config SND_SOC_SAMSUNG_H1940_UDA1380 - tristate "Audio support for the HP iPAQ H1940" - depends on ARCH_H1940 || COMPILE_TEST - depends on I2C - select SND_S3C24XX_I2S - select SND_SOC_UDA1380 - help - This driver provides audio support for HP iPAQ h1940 PDA. - -config SND_SOC_SAMSUNG_RX1950_UDA1380 - tristate "Audio support for the HP iPAQ RX1950" - depends on MACH_RX1950 || COMPILE_TEST - depends on I2C - select SND_S3C24XX_I2S - select SND_SOC_UDA1380 - help - This driver provides audio support for HP iPAQ RX1950 PDA. - -config SND_SOC_SMARTQ - tristate "SoC I2S Audio support for SmartQ board" - depends on MACH_SMARTQ || COMPILE_TEST - depends on GPIOLIB || COMPILE_TEST - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_WM8750 - config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" select SND_SAMSUNG_SPDIF diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 398e843f388c..f5d327b90a4e 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -2,35 +2,19 @@ # S3c24XX Platform Support snd-soc-s3c-dma-objs := dmaengine.o snd-soc-idma-objs := idma.o -snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o -snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o -snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o -obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o -obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support -snd-soc-jive-wm8750-objs := jive_wm8750.o -snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o -snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o -snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o -snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o -snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o -snd-soc-h1940-uda1380-objs := h1940_uda1380.o -snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o -snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm8994-objs := smdk_wm8994.o snd-soc-snow-objs := snow.o -snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o @@ -44,18 +28,8 @@ snd-soc-tm2-wm5110-objs := tm2_wm5110.o snd-soc-aries-wm8994-objs := aries_wm8994.o snd-soc-midas-wm1811-objs := midas_wm1811.o -obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o -obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o -obj-$(CONFIG_SND_SOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o -obj-$(CONFIG_SND_SOC_SAMSUNG_SIMTEC) += snd-soc-s3c24xx-simtec.o -obj-$(CONFIG_SND_SOC_SAMSUNG_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o -obj-$(CONFIG_SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o -obj-$(CONFIG_SND_SOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o -obj-$(CONFIG_SND_SOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o -obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o -obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c deleted file mode 100644 index fa45a54ab18f..000000000000 --- a/sound/soc/samsung/h1940_uda1380.c +++ /dev/null @@ -1,224 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// h1940_uda1380.c - ALSA SoC Audio Layer -// -// Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> -// Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com> -// -// Based on version from Arnaud Patard <arnaud.patard@rtp-net.org> - -#include <linux/types.h> -#include <linux/gpio/consumer.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/jack.h> - -#include "regs-iis.h" -#include "s3c24xx-i2s.h" - -static const unsigned int rates[] = { - 11025, - 22050, - 44100, -}; - -static const struct snd_pcm_hw_constraint_list hw_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, -}; - -static struct gpio_desc *gpiod_speaker_power; - -static struct snd_soc_jack hp_jack; - -static struct snd_soc_jack_pin hp_jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Speaker", - .mask = SND_JACK_HEADPHONE, - .invert = 1, - }, -}; - -static struct snd_soc_jack_gpio hp_jack_gpios[] = { - { - .name = "hp-gpio", - .report = SND_JACK_HEADPHONE, - .invert = 1, - .debounce_time = 200, - }, -}; - -static int h1940_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_rates); -} - -static int h1940_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int div; - int ret; - unsigned int rate = params_rate(params); - - switch (rate) { - case 11025: - case 22050: - case 44100: - div = s3c24xx_i2s_get_clockrate() / (384 * rate); - if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) - div++; - break; - default: - dev_err(rtd->dev, "%s: rate %d is not supported\n", - __func__, rate); - return -EINVAL; - } - - /* select clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_384FS); - if (ret < 0) - return ret; - - /* set BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(div, div)); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops h1940_ops = { - .startup = h1940_startup, - .hw_params = h1940_hw_params, -}; - -static int h1940_spk_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (SND_SOC_DAPM_EVENT_ON(event)) - gpiod_set_value(gpiod_speaker_power, 1); - else - gpiod_set_value(gpiod_speaker_power, 0); - - return 0; -} - -/* h1940 machine dapm widgets */ -static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), -}; - -/* h1940 machine audio_map */ -static const struct snd_soc_dapm_route audio_map[] = { - /* headphone connected to VOUTLHP, VOUTRHP */ - {"Headphone Jack", NULL, "VOUTLHP"}, - {"Headphone Jack", NULL, "VOUTRHP"}, - - /* ext speaker connected to VOUTL, VOUTR */ - {"Speaker", NULL, "VOUTL"}, - {"Speaker", NULL, "VOUTR"}, - - /* mic is connected to VINM */ - {"VINM", NULL, "Mic Jack"}, -}; - -static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) -{ - snd_soc_card_jack_new_pins(rtd->card, "Headphone Jack", - SND_JACK_HEADPHONE, - &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); - - snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); - - return 0; -} - -/* s3c24xx digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(uda1380, - DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")), - DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-001a", "uda1380-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis"))); - -static struct snd_soc_dai_link h1940_uda1380_dai[] = { - { - .name = "uda1380", - .stream_name = "UDA1380 Duplex", - .init = h1940_uda1380_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &h1940_ops, - SND_SOC_DAILINK_REG(uda1380), - }, -}; - -static struct snd_soc_card h1940_asoc = { - .name = "h1940", - .owner = THIS_MODULE, - .dai_link = h1940_uda1380_dai, - .num_links = ARRAY_SIZE(h1940_uda1380_dai), - - .dapm_widgets = uda1380_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static int h1940_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - - h1940_asoc.dev = dev; - hp_jack_gpios[0].gpiod_dev = dev; - gpiod_speaker_power = devm_gpiod_get(&pdev->dev, "speaker-power", - GPIOD_OUT_LOW); - - if (IS_ERR(gpiod_speaker_power)) { - dev_err(dev, "Could not get gpio\n"); - return PTR_ERR(gpiod_speaker_power); - } - - return devm_snd_soc_register_card(dev, &h1940_asoc); -} - -static struct platform_driver h1940_audio_driver = { - .driver = { - .name = "h1940-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = h1940_probe, -}; -module_platform_driver(h1940_audio_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); -MODULE_DESCRIPTION("ALSA SoC H1940"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:h1940-audio"); diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c deleted file mode 100644 index 40a85f539509..000000000000 --- a/sound/soc/samsung/jive_wm8750.c +++ /dev/null @@ -1,143 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Copyright 2007,2008 Simtec Electronics -// -// Based on sound/soc/pxa/spitz.c -// Copyright 2005 Wolfson Microelectronics PLC. -// Copyright 2005 Openedhand Ltd. - -#include <linux/module.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "s3c2412-i2s.h" -#include "../codecs/wm8750.h" - -static const struct snd_soc_dapm_route audio_map[] = { - { "Headphone Jack", NULL, "LOUT1" }, - { "Headphone Jack", NULL, "ROUT1" }, - { "Internal Speaker", NULL, "LOUT2" }, - { "Internal Speaker", NULL, "ROUT2" }, - { "LINPUT1", NULL, "Line Input" }, - { "RINPUT1", NULL, "Line Input" }, -}; - -static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Internal Speaker", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static int jive_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 s3c_i2sv2_rate_calc div; - 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; - } - - s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), - s3c_i2sv2_get_clock(cpu_dai)); - - /* 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; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, - div.clk_div - 1); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops jive_ops = { - .hw_params = jive_hw_params, -}; - -SND_SOC_DAILINK_DEFS(wm8750, - DAILINK_COMP_ARRAY(COMP_CPU("s3c2412-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8750.0-001a", "wm8750-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c2412-i2s"))); - -static struct snd_soc_dai_link jive_dai = { - .name = "wm8750", - .stream_name = "WM8750", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &jive_ops, - SND_SOC_DAILINK_REG(wm8750), -}; - -/* jive audio machine driver */ -static struct snd_soc_card snd_soc_machine_jive = { - .name = "Jive", - .owner = THIS_MODULE, - .dai_link = &jive_dai, - .num_links = 1, - - .dapm_widgets = wm8750_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, -}; - -static struct platform_device *jive_snd_device; - -static int __init jive_init(void) -{ - int ret; - - if (!machine_is_jive()) - return 0; - - printk("JIVE WM8750 Audio support\n"); - - jive_snd_device = platform_device_alloc("soc-audio", -1); - if (!jive_snd_device) - return -ENOMEM; - - platform_set_drvdata(jive_snd_device, &snd_soc_machine_jive); - ret = platform_device_add(jive_snd_device); - - if (ret) - platform_device_put(jive_snd_device); - - return ret; -} - -static void __exit jive_exit(void) -{ - platform_device_unregister(jive_snd_device); -} - -module_init(jive_init); -module_exit(jive_exit); - -MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c deleted file mode 100644 index e9f2334028bf..000000000000 --- a/sound/soc/samsung/neo1973_wm8753.c +++ /dev/null @@ -1,360 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// neo1973_wm8753.c - SoC audio for Openmoko Neo1973 and Freerunner devices -// -// Copyright 2007 Openmoko Inc -// Author: Graeme Gregory <graeme@openmoko.org> -// Copyright 2007 Wolfson Microelectronics PLC. -// Author: Graeme Gregory -// graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com -// Copyright 2009 Wolfson Microelectronics - -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/gpio/consumer.h> - -#include <sound/soc.h> - -#include "regs-iis.h" -#include "../codecs/wm8753.h" -#include "s3c24xx-i2s.h" - -static int neo1973_hifi_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 pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static const struct snd_soc_ops neo1973_hifi_ops = { - .hw_params = neo1973_hifi_hw_params, - .hw_free = neo1973_hifi_hw_free, -}; - -static int neo1973_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); - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); - if (ret < 0) - return ret; - - /* configure and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static const struct snd_soc_ops neo1973_voice_ops = { - .hw_params = neo1973_voice_hw_params, - .hw_free = neo1973_voice_hw_free, -}; - -static struct gpio_desc *gpiod_hp_in, *gpiod_amp_shut; -static int gta02_speaker_enabled; - -static int lm4853_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - gta02_speaker_enabled = ucontrol->value.integer.value[0]; - - gpiod_set_value(gpiod_hp_in, !gta02_speaker_enabled); - - return 0; -} - -static int lm4853_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = gta02_speaker_enabled; - return 0; -} - -static int lm4853_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpiod_set_value(gpiod_amp_shut, SND_SOC_DAPM_EVENT_OFF(event)); - - return 0; -} - -static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { - 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("Handset Mic", NULL), - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), -}; - -static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, - - /* Connections to the amp */ - {"Stereo Out", NULL, "LOUT1"}, - {"Stereo Out", NULL, "ROUT1"}, - - /* Call Speaker */ - {"Handset Spk", NULL, "LOUT2"}, - {"Handset Spk", NULL, "ROUT2"}, -}; - -static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Spk"), - SOC_DAPM_PIN_SWITCH("Stereo Out"), - - SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0, - lm4853_get_spk, - lm4853_set_spk), -}; - -static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - - /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic"); - snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out"); - snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk"); - - /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk"); - - return 0; -} - -SND_SOC_DAILINK_DEFS(wm8753, - DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8753.0-001a", "wm8753-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis"))); - -SND_SOC_DAILINK_DEFS(bluetooth, - DAILINK_COMP_ARRAY(COMP_CPU("bt-sco-pcm")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8753.0-001a", "wm8753-voice"))); - -static struct snd_soc_dai_link neo1973_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = neo1973_wm8753_init, - .ops = &neo1973_hifi_ops, - SND_SOC_DAILINK_REG(wm8753), -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &neo1973_voice_ops, - SND_SOC_DAILINK_REG(bluetooth), -}, -}; - -static struct snd_soc_aux_dev neo1973_aux_devs[] = { - { - .dlc = COMP_AUX("dfbmcs320.0"), - }, -}; - -static struct snd_soc_codec_conf neo1973_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF("lm4857.0-007c"), - .name_prefix = "Amp", - }, -}; - -static struct snd_soc_card neo1973 = { - .name = "neo1973gta02", - .owner = THIS_MODULE, - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), - .aux_dev = neo1973_aux_devs, - .num_aux_devs = ARRAY_SIZE(neo1973_aux_devs), - .codec_conf = neo1973_codec_conf, - .num_configs = ARRAY_SIZE(neo1973_codec_conf), - - .controls = neo1973_wm8753_controls, - .num_controls = ARRAY_SIZE(neo1973_wm8753_controls), - .dapm_widgets = neo1973_wm8753_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets), - .dapm_routes = neo1973_wm8753_routes, - .num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes), - .fully_routed = true, -}; - -static int neo1973_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - - gpiod_hp_in = devm_gpiod_get(dev, "hp", GPIOD_OUT_HIGH); - if (IS_ERR(gpiod_hp_in)) { - dev_err(dev, "missing gpio %s\n", "hp"); - return PTR_ERR(gpiod_hp_in); - } - gpiod_amp_shut = devm_gpiod_get(dev, "amp-shut", GPIOD_OUT_HIGH); - if (IS_ERR(gpiod_amp_shut)) { - dev_err(dev, "missing gpio %s\n", "amp-shut"); - return PTR_ERR(gpiod_amp_shut); - } - - neo1973.dev = dev; - return devm_snd_soc_register_card(dev, &neo1973); -} - -static struct platform_driver neo1973_audio = { - .driver = { - .name = "neo1973-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = neo1973_probe, -}; -module_platform_driver(neo1973_audio); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:neo1973-audio"); diff --git a/sound/soc/samsung/regs-i2s-v2.h b/sound/soc/samsung/regs-i2s-v2.h deleted file mode 100644 index 867984e75709..000000000000 --- a/sound/soc/samsung/regs-i2s-v2.h +++ /dev/null @@ -1,111 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - * Copyright 2007 Simtec Electronics <linux@simtec.co.uk> - * http://armlinux.simtec.co.uk/ - * - * S3C2412 IIS register definition - */ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S5PC1XX_IISFICS 0x18 -#define S5PC1XX_IISTXDS 0x1C - -#define S5PC1XX_IISCON_SW_RST (1 << 31) -#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) -#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) -#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) -#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) -#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) -#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) - -#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) -#define S3C64XX_IISCON_FTXURINTEN (1 << 16) -#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) -#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) -#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) -#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) -#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) -#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) -#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) -#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) -#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ -#define S5PC1XX_IISMOD_BLCS_MASK 0x3 -#define S5PC1XX_IISMOD_BLCS_SHIFT 26 -#define S5PC1XX_IISMOD_BLCP_MASK 0x3 -#define S5PC1XX_IISMOD_BLCP_SHIFT 24 - -#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ -#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ -#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) -#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) -#define S3C64XX_IISMOD_DC2_EN (1 << 17) -#define S3C64XX_IISMOD_DC1_EN (1 << 16) -#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) -#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) -#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) -#define S3C64XX_IISMOD_BLC_MASK (3 << 13) - -#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) -#define S3C2412_IISMOD_SLAVE (1 << 11) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (1 << 5) -#define S3C2412_IISMOD_SDF_LSB (2 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C64XX_IISMOD_CDCLKCON (1 << 12) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) -#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - -#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) -#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h deleted file mode 100644 index 253e172ad3b6..000000000000 --- a/sound/soc/samsung/regs-iis.h +++ /dev/null @@ -1,66 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - * Copyright (c) 2003 Simtec Electronics <linux@simtec.co.uk> - * http://www.simtec.co.uk/products/SWLINUX/ - * - * S3C2410 IIS register definition - */ - -#ifndef __SAMSUNG_REGS_IIS_H__ -#define __SAMSUNG_REGS_IIS_H__ - -#define S3C2410_IISCON (0x00) - -#define S3C2410_IISCON_LRINDEX (1 << 8) -#define S3C2410_IISCON_TXFIFORDY (1 << 7) -#define S3C2410_IISCON_RXFIFORDY (1 << 6) -#define S3C2410_IISCON_TXDMAEN (1 << 5) -#define S3C2410_IISCON_RXDMAEN (1 << 4) -#define S3C2410_IISCON_TXIDLE (1 << 3) -#define S3C2410_IISCON_RXIDLE (1 << 2) -#define S3C2410_IISCON_PSCEN (1 << 1) -#define S3C2410_IISCON_IISEN (1 << 0) - -#define S3C2410_IISMOD (0x04) - -#define S3C2440_IISMOD_MPLL (1 << 9) -#define S3C2410_IISMOD_SLAVE (1 << 8) -#define S3C2410_IISMOD_NOXFER (0 << 6) -#define S3C2410_IISMOD_RXMODE (1 << 6) -#define S3C2410_IISMOD_TXMODE (2 << 6) -#define S3C2410_IISMOD_TXRXMODE (3 << 6) -#define S3C2410_IISMOD_LR_LLOW (0 << 5) -#define S3C2410_IISMOD_LR_RLOW (1 << 5) -#define S3C2410_IISMOD_IIS (0 << 4) -#define S3C2410_IISMOD_MSB (1 << 4) -#define S3C2410_IISMOD_8BIT (0 << 3) -#define S3C2410_IISMOD_16BIT (1 << 3) -#define S3C2410_IISMOD_BITMASK (1 << 3) -#define S3C2410_IISMOD_256FS (0 << 2) -#define S3C2410_IISMOD_384FS (1 << 2) -#define S3C2410_IISMOD_16FS (0 << 0) -#define S3C2410_IISMOD_32FS (1 << 0) -#define S3C2410_IISMOD_48FS (2 << 0) -#define S3C2410_IISMOD_FS_MASK (3 << 0) - -#define S3C2410_IISPSR (0x08) - -#define S3C2410_IISPSR_INTMASK (31 << 5) -#define S3C2410_IISPSR_INTSHIFT (5) -#define S3C2410_IISPSR_EXTMASK (31 << 0) -#define S3C2410_IISPSR_EXTSHFIT (0) - -#define S3C2410_IISFCON (0x0c) - -#define S3C2410_IISFCON_TXDMA (1 << 15) -#define S3C2410_IISFCON_RXDMA (1 << 14) -#define S3C2410_IISFCON_TXENABLE (1 << 13) -#define S3C2410_IISFCON_RXENABLE (1 << 12) -#define S3C2410_IISFCON_TXMASK (0x3f << 6) -#define S3C2410_IISFCON_TXSHIFT (6) -#define S3C2410_IISFCON_RXMASK (0x3f) -#define S3C2410_IISFCON_RXSHIFT (0) - -#define S3C2410_IISFIFO (0x10) - -#endif /* __SAMSUNG_REGS_IIS_H__ */ diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c deleted file mode 100644 index abf28321f7d7..000000000000 --- a/sound/soc/samsung/rx1950_uda1380.c +++ /dev/null @@ -1,245 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// rx1950.c - ALSA SoC Audio Layer -// -// Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com> -// -// Based on smdk2440.c and magician.c -// -// Authors: Graeme Gregory graeme.gregory@wolfsonmicro.com -// Philipp Zabel <philipp.zabel@gmail.com> -// Denis Grigoriev <dgreenday@gmail.com> -// Vasily Khoruzhick <anarsoul@gmail.com> - -#include <linux/types.h> -#include <linux/gpio/consumer.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/jack.h> - -#include "regs-iis.h" -#include "s3c24xx-i2s.h" - -static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); -static int rx1950_startup(struct snd_pcm_substream *substream); -static int rx1950_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params); -static int rx1950_spk_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event); - -static const unsigned int rates[] = { - 16000, - 44100, - 48000, -}; - -static const struct snd_pcm_hw_constraint_list hw_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, -}; - -static struct snd_soc_jack hp_jack; - -static struct snd_soc_jack_pin hp_jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Speaker", - .mask = SND_JACK_HEADPHONE, - .invert = 1, - }, -}; - -static struct snd_soc_jack_gpio hp_jack_gpios[] = { - [0] = { - .name = "hp-gpio", - .report = SND_JACK_HEADPHONE, - .invert = 1, - .debounce_time = 200, - }, -}; - -static const struct snd_soc_ops rx1950_ops = { - .startup = rx1950_startup, - .hw_params = rx1950_hw_params, -}; - -/* s3c24xx digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(uda1380, - DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")), - DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-001a", - "uda1380-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis"))); - -static struct snd_soc_dai_link rx1950_uda1380_dai[] = { - { - .name = "uda1380", - .stream_name = "UDA1380 Duplex", - .init = rx1950_uda1380_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &rx1950_ops, - SND_SOC_DAILINK_REG(uda1380), - }, -}; - -/* rx1950 machine dapm widgets */ -static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", rx1950_spk_power), -}; - -/* rx1950 machine audio_map */ -static const struct snd_soc_dapm_route audio_map[] = { - /* headphone connected to VOUTLHP, VOUTRHP */ - {"Headphone Jack", NULL, "VOUTLHP"}, - {"Headphone Jack", NULL, "VOUTRHP"}, - - /* ext speaker connected to VOUTL, VOUTR */ - {"Speaker", NULL, "VOUTL"}, - {"Speaker", NULL, "VOUTR"}, - - /* mic is connected to VINM */ - {"VINM", NULL, "Mic Jack"}, -}; - -static struct snd_soc_card rx1950_asoc = { - .name = "rx1950", - .owner = THIS_MODULE, - .dai_link = rx1950_uda1380_dai, - .num_links = ARRAY_SIZE(rx1950_uda1380_dai), - - .dapm_widgets = uda1380_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static int rx1950_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_rates); -} - -static struct gpio_desc *gpiod_speaker_power; - -static int rx1950_spk_power(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - if (SND_SOC_DAPM_EVENT_ON(event)) - gpiod_set_value(gpiod_speaker_power, 1); - else - gpiod_set_value(gpiod_speaker_power, 0); - - return 0; -} - -static int rx1950_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 *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int div; - int ret; - unsigned int rate = params_rate(params); - int clk_source, fs_mode; - - switch (rate) { - case 16000: - case 48000: - clk_source = S3C24XX_CLKSRC_PCLK; - fs_mode = S3C2410_IISMOD_256FS; - div = s3c24xx_i2s_get_clockrate() / (256 * rate); - if (s3c24xx_i2s_get_clockrate() % (256 * rate) > (128 * rate)) - div++; - break; - case 44100: - case 88200: - clk_source = S3C24XX_CLKSRC_MPLL; - fs_mode = S3C2410_IISMOD_384FS; - div = 1; - break; - default: - printk(KERN_ERR "%s: rate %d is not supported\n", - __func__, rate); - return -EINVAL; - } - - /* select clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source, rate, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - fs_mode); - if (ret < 0) - return ret; - - /* set BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(div, div)); - if (ret < 0) - return ret; - - return 0; -} - -static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) -{ - snd_soc_card_jack_new_pins(rtd->card, "Headphone Jack", - SND_JACK_HEADPHONE, - &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); - - snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); - - return 0; -} - -static int rx1950_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - - /* configure some gpios */ - gpiod_speaker_power = devm_gpiod_get(dev, "speaker-power", GPIOD_OUT_LOW); - if (IS_ERR(gpiod_speaker_power)) { - dev_err(dev, "cannot get gpio\n"); - return PTR_ERR(gpiod_speaker_power); - } - - hp_jack_gpios[0].gpiod_dev = dev; - rx1950_asoc.dev = dev; - - return devm_snd_soc_register_card(dev, &rx1950_asoc); -} - -static struct platform_driver rx1950_audio = { - .driver = { - .name = "rx1950-audio", - .pm = &snd_soc_pm_ops, - }, - .probe = rx1950_probe, -}; - -module_platform_driver(rx1950_audio); - -/* Module information */ -MODULE_AUTHOR("Vasily Khoruzhick"); -MODULE_DESCRIPTION("ALSA SoC RX1950"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:rx1950-audio"); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c deleted file mode 100644 index 2b221cb0ed03..000000000000 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ /dev/null @@ -1,670 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. -// -// Copyright (c) 2006 Wolfson Microelectronics PLC. -// Graeme Gregory graeme.gregory@wolfsonmicro.com -// linux@wolfsonmicro.com -// -// Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics -// http://armlinux.simtec.co.uk/ -// Ben Dooks <ben@simtec.co.uk> - -#include <linux/module.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "regs-i2s-v2.h" -#include "s3c-i2s-v2.h" - -#define S3C2412_I2S_DEBUG_CON 0 - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return snd_soc_dai_get_drvdata(cpu_dai); -} - -#define bit_set(v, b) (((v) & (b)) ? 1 : 0) - -#if S3C2412_I2S_DEBUG_CON -static void dbg_showcon(const char *fn, u32 con) -{ - printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, - bit_set(con, S3C2412_IISCON_LRINDEX), - bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_TXFIFO_FULL), - bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); - - printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", - fn, - bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_TXCH_PAUSE), - bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); - printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, - bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); -} -#else -static inline void dbg_showcon(const char *fn, u32 con) -{ -} -#endif - -/* Turn on or off the transmission path. */ -static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_TXDMA_PAUSE; - con &= ~S3C2412_IISCON_TXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXONLY: - case S3C2412_IISMOD_MODE_TXRX: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_RXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } else { - /* Note, we do not have any indication that the FIFO problems - * tha the S3C2410/2440 had apply here, so we should be able - * to disable the DMA and TX without resetting the FIFOS. - */ - - con |= S3C2412_IISCON_TXDMA_PAUSE; - con |= S3C2412_IISCON_TXCH_PAUSE; - con &= ~S3C2412_IISCON_TXDMA_ACTIVE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_RXONLY; - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - con &= ~S3C2412_IISCON_IIS_ACTIVE; - break; - - default: - dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } - - fic = readl(regs + S3C2412_IISFIC); - dbg_showcon(__func__, con); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_RXDMA_PAUSE; - con &= ~S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - case S3C2412_IISMOD_MODE_RXONLY: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } else { - /* See txctrl notes on FIFOs. */ - - con &= ~S3C2412_IISCON_RXDMA_ACTIVE; - con |= S3C2412_IISCON_RXDMA_PAUSE; - con |= S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_RXONLY: - con &= ~S3C2412_IISCON_IIS_ACTIVE; - mod &= ~S3C2412_IISMOD_MODE_MASK; - break; - - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXONLY; - break; - - default: - dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } - - fic = readl(regs + S3C2412_IISFIC); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) -{ - u32 iiscon; - unsigned long loops = msecs_to_loops(5); - - pr_debug("Entered %s\n", __func__); - - while (--loops) { - iiscon = readl(i2s->regs + S3C2412_IISCON); - if (iiscon & S3C2412_IISCON_LRINDEX) - break; - - cpu_relax(); - } - - if (!loops) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } - - return 0; -} - -/* - * Set S3C2412 I2S DAI format - */ -static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BC_FC: - i2s->master = 0; - iismod |= S3C2412_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_BP_FP: - i2s->master = 1; - iismod &= ~S3C2412_IISMOD_SLAVE; - break; - default: - pr_err("unknown master/slave format\n"); - return -EINVAL; - } - - iismod &= ~S3C2412_IISMOD_SDF_MASK; - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_RIGHT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_MSB; - break; - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_LSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_IIS; - break; - default: - pr_err("Unknown data format\n"); - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - struct snd_dmaengine_dai_dma_data *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - /* Sample size */ - switch (params_width(params)) { - case 8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case 16: - break; - case 24: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - pr_debug("Entered %s\n", __func__); - pr_debug("%s r: IISMOD: %x\n", __func__, iismod); - - switch (clk_id) { - case S3C_I2SV2_CLKSRC_PCLK: - iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_AUDIOBUS: - iismod |= S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_CDCLK: - /* Error if controller doesn't have the CDCLKCON bit */ - if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) - return -EINVAL; - - switch (dir) { - case SND_SOC_CLOCK_IN: - iismod |= S3C64XX_IISMOD_CDCLKCON; - break; - case SND_SOC_CLOCK_OUT: - iismod &= ~S3C64XX_IISMOD_CDCLKCON; - break; - default: - return -EINVAL; - } - break; - - default: - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct s3c_i2sv2_info *i2s = to_info(asoc_rtd_to_cpu(rtd, 0)); - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - unsigned long irqs; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* On start, ensure that the FIFOs are cleared and reset. */ - - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - /* clear again, just in case */ - writel(0x0, i2s->regs + S3C2412_IISFIC); - - fallthrough; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!i2s->master) { - ret = s3c2412_snd_lrsync(i2s); - if (ret) - goto exit_err; - } - - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 1); - else - s3c2412_snd_txctrl(i2s, 1); - - local_irq_restore(irqs); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 0); - else - s3c2412_snd_txctrl(i2s, 0); - - local_irq_restore(irqs); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C2412 Clock dividers - */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 reg; - - pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); - - switch (div_id) { - case S3C_I2SV2_DIV_BCLK: - switch (div) { - case 16: - div = S3C2412_IISMOD_BCLK_16FS; - break; - - case 32: - div = S3C2412_IISMOD_BCLK_32FS; - break; - - case 24: - div = S3C2412_IISMOD_BCLK_24FS; - break; - - case 48: - div = S3C2412_IISMOD_BCLK_48FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_BCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_RCLK: - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; - - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; - - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; - - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_RCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_PRESCALER: - if (div >= 0) { - writel((div << 8) | S3C2412_IISPSR_PSREN, - i2s->regs + S3C2412_IISPSR); - } else { - writel(0x0, i2s->regs + S3C2412_IISPSR); - } - pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 reg = readl(i2s->regs + S3C2412_IISFIC); - snd_pcm_sframes_t delay; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - delay = S3C2412_IISFIC_TXCOUNT(reg); - else - delay = S3C2412_IISFIC_RXCOUNT(reg); - - return delay; -} - -struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISMOD_IMS_SYSMUX) - return i2s->iis_cclk; - else - return i2s->iis_pclk; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); - -/* default table of all avaialable root fs divisors */ -static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; - -int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) -{ - unsigned long clkrate = clk_get_rate(clk); - unsigned int div; - unsigned int fsclk; - unsigned int actual; - unsigned int fs; - unsigned int fsdiv; - signed int deviation = 0; - unsigned int best_fs = 0; - unsigned int best_div = 0; - unsigned int best_rate = 0; - unsigned int best_deviation = INT_MAX; - - pr_debug("Input clock rate %ldHz\n", clkrate); - - if (fstab == NULL) - fstab = iis_fs_tab; - - for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { - fsdiv = iis_fs_tab[fs]; - - fsclk = clkrate / fsdiv; - div = fsclk / rate; - - if ((fsclk % rate) > (rate / 2)) - div++; - - if (div <= 1) - continue; - - actual = clkrate / (fsdiv * div); - deviation = actual - rate; - - printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", - fsdiv, div, actual, deviation); - - deviation = abs(deviation); - - if (deviation < best_deviation) { - best_fs = fsdiv; - best_div = div; - best_rate = actual; - best_deviation = deviation; - } - - if (deviation == 0) - break; - } - - printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", - best_fs, best_div, best_rate); - - info->fs_div = best_fs; - info->clk_div = best_div; - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); - -int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s) -{ - struct device *dev = dai->dev; - unsigned int iismod; - - i2s->dev = dev; - - /* record our i2s structure for later use in the callbacks */ - snd_soc_dai_set_drvdata(dai, i2s); - - i2s->iis_pclk = clk_get(dev, "iis"); - if (IS_ERR(i2s->iis_pclk)) { - dev_err(dev, "failed to get iis_clock\n"); - return -ENOENT; - } - - clk_prepare_enable(i2s->iis_pclk); - - /* Mark ourselves as in TXRX mode so we can run through our cleanup - * process without warnings. */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - iismod |= S3C2412_IISMOD_MODE_TXRX; - writel(iismod, i2s->regs + S3C2412_IISMOD); - s3c2412_snd_txctrl(i2s, 0); - s3c2412_snd_rxctrl(i2s, 0); - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); - -void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s) -{ - clk_disable_unprepare(i2s->iis_pclk); - clk_put(i2s->iis_pclk); - i2s->iis_pclk = NULL; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup); - -int s3c_i2sv2_register_component(struct device *dev, int id, - const struct snd_soc_component_driver *cmp_drv, - struct snd_soc_dai_driver *dai_drv) -{ - struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; - - ops->trigger = s3c2412_i2s_trigger; - if (!ops->hw_params) - ops->hw_params = s3c_i2sv2_hw_params; - ops->set_fmt = s3c2412_i2s_set_fmt; - ops->set_clkdiv = s3c2412_i2s_set_clkdiv; - ops->set_sysclk = s3c_i2sv2_set_sysclk; - - /* Allow overriding by (for example) IISv4 */ - if (!ops->delay) - ops->delay = s3c2412_i2s_delay; - - return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1); -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); - -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h deleted file mode 100644 index 8c6fc0d3d77e..000000000000 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ /dev/null @@ -1,108 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0+ */ -/* - * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben@simtec.co.uk> - */ - -/* This code is the core support for the I2S block found in a number of - * Samsung SoC devices which is unofficially named I2S-V2. Currently the - * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S - * channels via configurable GPIO. - */ - -#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H -#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ - -#define S3C_I2SV2_DIV_BCLK (1) -#define S3C_I2SV2_DIV_RCLK (2) -#define S3C_I2SV2_DIV_PRESCALER (3) - -#define S3C_I2SV2_CLKSRC_PCLK 0 -#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 -#define S3C_I2SV2_CLKSRC_CDCLK 2 - -/* Set this flag for I2S controllers that have the bit IISMOD[12] - * bridge/break RCLK signal and external Xi2sCDCLK pin. - */ -#define S3C_FEATURE_CDCLKCON (1 << 0) - -/** - * struct s3c_i2sv2_info - S3C I2S-V2 information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device registe block. - * @feature: Set of bit-flags indicating features of the controller. - * @master: True if the I2S core is the I2S bit clock master. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - * @suspend_iismod: PM save for the IISMOD register. - * @suspend_iiscon: PM save for the IISCON register. - * @suspend_iispsr: PM save for the IISPSR register. - * - * This is the private codec state for the hardware associated with an - * I2S channel such as the register mappings and clock sources. - */ -struct s3c_i2sv2_info { - struct device *dev; - void __iomem *regs; - - u32 feature; - - struct clk *iis_pclk; - struct clk *iis_cclk; - - unsigned char master; - - struct snd_dmaengine_dai_dma_data *dma_playback; - struct snd_dmaengine_dai_dma_data *dma_capture; - - u32 suspend_iismod; - u32 suspend_iiscon; - u32 suspend_iispsr; - - unsigned long base; -}; - -extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); - -struct s3c_i2sv2_rate_calc { - unsigned int clk_div; /* for prescaler */ - unsigned int fs_div; /* for root frame clock */ -}; - -extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk); - -/** - * s3c_i2sv2_probe - probe for i2s device helper - * @dai: The ASoC DAI structure supplied to the original probe. - * @i2s: Our local i2s structure to fill in. - * @base: The base address for the registers. - */ -extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s); - -/** - * s3c_i2sv2_cleanup - cleanup resources allocated in s3c_i2sv2_probe - * @dai: The ASoC DAI structure supplied to the original probe. - * @i2s: Our local i2s structure to fill in. - */ -extern void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s); -/** - * s3c_i2sv2_register_component - register component and dai with soc core - * @dev: DAI device - * @id: DAI ID - * @drv: The driver structure to register - * - * Fill in any missing fields and then register the given dai with the - * soc core. - */ -extern int s3c_i2sv2_register_component(struct device *dev, int id, - const struct snd_soc_component_driver *cmp_drv, - struct snd_soc_dai_driver *dai_drv); - -#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c deleted file mode 100644 index 0579a352961c..000000000000 --- a/sound/soc/samsung/s3c2412-i2s.c +++ /dev/null @@ -1,251 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// ALSA Soc Audio Layer - S3C2412 I2S driver -// -// Copyright (c) 2006 Wolfson Microelectronics PLC. -// Graeme Gregory graeme.gregory@wolfsonmicro.com -// linux@wolfsonmicro.com -// -// Copyright (c) 2007, 2004-2005 Simtec Electronics -// http://armlinux.simtec.co.uk/ -// Ben Dooks <ben@simtec.co.uk> - -#include <linux/delay.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "dma.h" -#include "regs-i2s-v2.h" -#include "s3c2412-i2s.h" - -#include <linux/platform_data/asoc-s3c.h> - -static struct snd_dmaengine_dai_dma_data s3c2412_i2s_pcm_stereo_out = { - .chan_name = "tx", - .addr_width = 4, -}; - -static struct snd_dmaengine_dai_dma_data s3c2412_i2s_pcm_stereo_in = { - .chan_name = "rx", - .addr_width = 4, -}; - -static struct s3c_i2sv2_info s3c2412_i2s; - -static int s3c2412_i2s_probe(struct snd_soc_dai *dai) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - snd_soc_dai_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, - &s3c2412_i2s_pcm_stereo_in); - - ret = s3c_i2sv2_probe(dai, &s3c2412_i2s); - if (ret) - return ret; - - s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; - s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - - s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); - if (IS_ERR(s3c2412_i2s.iis_cclk)) { - pr_err("failed to get i2sclk clock\n"); - ret = PTR_ERR(s3c2412_i2s.iis_cclk); - goto err; - } - - /* Set MPLL as the source for IIS CLK */ - - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - ret = clk_prepare_enable(s3c2412_i2s.iis_cclk); - if (ret) - goto err; - - return 0; - -err: - s3c_i2sv2_cleanup(dai, &s3c2412_i2s); - - return ret; -} - -static int s3c2412_i2s_remove(struct snd_soc_dai *dai) -{ - clk_disable_unprepare(s3c2412_i2s.iis_cclk); - s3c_i2sv2_cleanup(dai, &s3c2412_i2s); - - return 0; -} - -static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - switch (params_width(params)) { - case 8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case 16: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_component *component) -{ - struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); - u32 iismod; - - if (component->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warn("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warn("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warn("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_component *component) -{ - struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); - - pr_info("component_active %d, IISMOD %08x, IISCON %08x\n", - component->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (component->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - -#define S3C2412_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static const struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { - .hw_params = s3c2412_i2s_hw_params, -}; - -static struct snd_soc_dai_driver s3c2412_i2s_dai = { - .probe = s3c2412_i2s_probe, - .remove = s3c2412_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &s3c2412_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver s3c2412_i2s_component = { - .name = "s3c2412-i2s", - .suspend = s3c2412_i2s_suspend, - .resume = s3c2412_i2s_resume, - .legacy_dai_naming = 1, -}; - -static int s3c2412_iis_dev_probe(struct platform_device *pdev) -{ - int ret = 0; - struct resource *res; - struct s3c_audio_pdata *pdata = dev_get_platdata(&pdev->dev); - - if (!pdata) { - dev_err(&pdev->dev, "missing platform data"); - return -ENXIO; - } - - s3c2412_i2s.regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); - if (IS_ERR(s3c2412_i2s.regs)) - return PTR_ERR(s3c2412_i2s.regs); - - s3c2412_i2s_pcm_stereo_out.addr = res->start + S3C2412_IISTXD; - s3c2412_i2s_pcm_stereo_out.filter_data = pdata->dma_playback; - s3c2412_i2s_pcm_stereo_in.addr = res->start + S3C2412_IISRXD; - s3c2412_i2s_pcm_stereo_in.filter_data = pdata->dma_capture; - - ret = samsung_asoc_dma_platform_register(&pdev->dev, - pdata->dma_filter, - "tx", "rx", NULL); - if (ret) { - pr_err("failed to register the DMA: %d\n", ret); - return ret; - } - - ret = s3c_i2sv2_register_component(&pdev->dev, -1, - &s3c2412_i2s_component, - &s3c2412_i2s_dai); - if (ret) - pr_err("failed to register the dai\n"); - - return ret; -} - -static struct platform_driver s3c2412_iis_driver = { - .probe = s3c2412_iis_dev_probe, - .driver = { - .name = "s3c2412-iis", - }, -}; - -module_platform_driver(s3c2412_iis_driver); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:s3c2412-iis"); diff --git a/sound/soc/samsung/s3c2412-i2s.h b/sound/soc/samsung/s3c2412-i2s.h deleted file mode 100644 index bff2a797cb08..000000000000 --- a/sound/soc/samsung/s3c2412-i2s.h +++ /dev/null @@ -1,22 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0+ */ -/* - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben@simtec.co.uk> - */ - -#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H -#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ - -#include "s3c-i2s-v2.h" - -#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS - -#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c deleted file mode 100644 index 7b7bbe007acd..000000000000 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ /dev/null @@ -1,463 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// s3c24xx-i2s.c -- ALSA Soc Audio Layer -// -// (c) 2006 Wolfson Microelectronics PLC. -// Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com -// -// Copyright 2004-2005 Simtec Electronics -// http://armlinux.simtec.co.uk/ -// Ben Dooks <ben@simtec.co.uk> - -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "regs-iis.h" -#include "dma.h" -#include "s3c24xx-i2s.h" - -static struct snd_dmaengine_dai_dma_data s3c24xx_i2s_pcm_stereo_out = { - .chan_name = "tx", - .addr_width = 2, -}; - -static struct snd_dmaengine_dai_dma_data s3c24xx_i2s_pcm_stereo_in = { - .chan_name = "rx", - .addr_width = 2, -}; - -struct s3c24xx_i2s_info { - void __iomem *regs; - struct clk *iis_clk; - u32 iiscon; - u32 iismod; - u32 iisfcon; - u32 iispsr; -}; -static struct s3c24xx_i2s_info s3c24xx_i2s; - -static void s3c24xx_snd_txctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; - iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_TXIDLE; - iismod |= S3C2410_IISMOD_TXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_TXENABLE; - iisfcon &= ~S3C2410_IISFCON_TXDMA; - iiscon |= S3C2410_IISCON_TXIDLE; - iiscon &= ~S3C2410_IISCON_TXDMAEN; - iismod &= ~S3C2410_IISMOD_TXMODE; - - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -static void s3c24xx_snd_rxctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; - iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_RXIDLE; - iismod |= S3C2410_IISMOD_RXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_RXENABLE; - iisfcon &= ~S3C2410_IISFCON_RXDMA; - iiscon |= S3C2410_IISCON_RXIDLE; - iiscon &= ~S3C2410_IISCON_RXDMAEN; - iismod &= ~S3C2410_IISMOD_RXMODE; - - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c24xx_snd_lrsync(void) -{ - u32 iiscon; - int timeout = 50; /* 5ms */ - - while (1) { - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - if (iiscon & S3C2410_IISCON_LRINDEX) - break; - - if (!timeout--) - return -ETIMEDOUT; - udelay(100); - } - - return 0; -} - -/* - * Check whether CPU is the master or slave - */ -static inline int s3c24xx_snd_is_clkmaster(void) -{ - return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; -} - -/* - * Set S3C24xx I2S DAI format - */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 iismod; - - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BC_FC: - iismod |= S3C2410_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_BP_FP: - iismod &= ~S3C2410_IISMOD_SLAVE; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2410_IISMOD_MSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2410_IISMOD_MSB; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - - return 0; -} - -static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_dmaengine_dai_dma_data *dma_data; - u32 iismod; - - dma_data = snd_soc_dai_get_dma_data(dai, substream); - - /* Working copies of register */ - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x\n", iismod); - - switch (params_width(params)) { - case 8: - iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->addr_width = 1; - break; - case 16: - iismod |= S3C2410_IISMOD_16BIT; - dma_data->addr_width = 2; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x\n", iismod); - - return 0; -} - -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!s3c24xx_snd_is_clkmaster()) { - ret = s3c24xx_snd_lrsync(); - if (ret) - goto exit_err; - } - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(1); - else - s3c24xx_snd_txctrl(1); - - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(0); - else - s3c24xx_snd_txctrl(0); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C24xx Clock source - */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - iismod &= ~S3C2440_IISMOD_MPLL; - - switch (clk_id) { - case S3C24XX_CLKSRC_PCLK: - break; - case S3C24XX_CLKSRC_MPLL: - iismod |= S3C2440_IISMOD_MPLL; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - return 0; -} - -/* - * Set S3C24xx Clock dividers - */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - u32 reg; - - switch (div_id) { - case S3C24XX_DIV_BCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_MCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_PRESCALER: - writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); - reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * To avoid duplicating clock code, allow machine driver to - * get the clockrate from here. - */ -u32 s3c24xx_i2s_get_clockrate(void) -{ - return clk_get_rate(s3c24xx_i2s.iis_clk); -} -EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); - -static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) -{ - int ret; - snd_soc_dai_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, - &s3c24xx_i2s_pcm_stereo_in); - - s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); - if (IS_ERR(s3c24xx_i2s.iis_clk)) { - pr_err("failed to get iis_clock\n"); - return PTR_ERR(s3c24xx_i2s.iis_clk); - } - ret = clk_prepare_enable(s3c24xx_i2s.iis_clk); - if (ret) - return ret; - - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); - - s3c24xx_snd_txctrl(0); - s3c24xx_snd_rxctrl(0); - - return 0; -} - -#ifdef CONFIG_PM -static int s3c24xx_i2s_suspend(struct snd_soc_component *component) -{ - s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); - - clk_disable_unprepare(s3c24xx_i2s.iis_clk); - - return 0; -} - -static int s3c24xx_i2s_resume(struct snd_soc_component *component) -{ - int ret; - - ret = clk_prepare_enable(s3c24xx_i2s.iis_clk); - if (ret) - return ret; - - writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); - - return 0; -} -#else -#define s3c24xx_i2s_suspend NULL -#define s3c24xx_i2s_resume NULL -#endif - -#define S3C24XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static const struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, -}; - -static struct snd_soc_dai_driver s3c24xx_i2s_dai = { - .probe = s3c24xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c24xx_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver s3c24xx_i2s_component = { - .name = "s3c24xx-i2s", - .suspend = s3c24xx_i2s_suspend, - .resume = s3c24xx_i2s_resume, - .legacy_dai_naming = 1, -}; - -static int s3c24xx_iis_dev_probe(struct platform_device *pdev) -{ - struct resource *res; - int ret; - - s3c24xx_i2s.regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); - if (IS_ERR(s3c24xx_i2s.regs)) - return PTR_ERR(s3c24xx_i2s.regs); - - s3c24xx_i2s_pcm_stereo_out.addr = res->start + S3C2410_IISFIFO; - s3c24xx_i2s_pcm_stereo_in.addr = res->start + S3C2410_IISFIFO; - - ret = samsung_asoc_dma_platform_register(&pdev->dev, NULL, - "tx", "rx", NULL); - if (ret) { - dev_err(&pdev->dev, "Failed to register the DMA: %d\n", ret); - return ret; - } - - ret = devm_snd_soc_register_component(&pdev->dev, - &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); - if (ret) - dev_err(&pdev->dev, "Failed to register the DAI\n"); - - return ret; -} - -static struct platform_driver s3c24xx_iis_driver = { - .probe = s3c24xx_iis_dev_probe, - .driver = { - .name = "s3c24xx-iis", - }, -}; - -module_platform_driver(s3c24xx_iis_driver); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:s3c24xx-iis"); diff --git a/sound/soc/samsung/s3c24xx-i2s.h b/sound/soc/samsung/s3c24xx-i2s.h deleted file mode 100644 index e073e31855d0..000000000000 --- a/sound/soc/samsung/s3c24xx-i2s.h +++ /dev/null @@ -1,31 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0+ */ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Revision history - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXI2S_H_ -#define S3C24XXI2S_H_ - -/* clock sources */ -#define S3C24XX_CLKSRC_PCLK 0 -#define S3C24XX_CLKSRC_MPLL 1 - -/* Clock dividers */ -#define S3C24XX_DIV_MCLK 0 -#define S3C24XX_DIV_BCLK 1 -#define S3C24XX_DIV_PRESCALER 2 - -/* prescaler */ -#define S3C24XX_PRESCALE(a,b) \ - (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) - -u32 s3c24xx_i2s_get_clockrate(void); - -#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c deleted file mode 100644 index 0cc66774b85d..000000000000 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ /dev/null @@ -1,372 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Copyright 2009 Simtec Electronics - -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/module.h> - -#include <sound/soc.h> - -#include <linux/platform_data/asoc-s3c24xx_simtec.h> - -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -static struct s3c24xx_audio_simtec_pdata *pdata; -static struct clk *xtal_clk; - -static int spk_gain; -static int spk_unmute; - -/** - * speaker_gain_get - read the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_gain_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_gain; - return 0; -} - -/** - * speaker_gain_set - set the value of the speaker amp gain - * @value: The value to write. - */ -static void speaker_gain_set(int value) -{ - gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); - gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); -} - -/** - * speaker_gain_put - set the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - * - * Note, if the speaker amp is muted, then we do not set a gain value - * as at-least one of the ICs that is fitted will try and power up even - * if the main control is set to off. - */ -static int speaker_gain_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int value = ucontrol->value.integer.value[0]; - - spk_gain = value; - - if (!spk_unmute) - speaker_gain_set(value); - - return 0; -} - -static const struct snd_kcontrol_new amp_gain_controls[] = { - SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, - speaker_gain_get, speaker_gain_put), -}; - -/** - * spk_unmute_state - set the unmute state of the speaker - * @to: zero to unmute, non-zero to ununmute. - */ -static void spk_unmute_state(int to) -{ - pr_debug("%s: to=%d\n", __func__, to); - - spk_unmute = to; - gpio_set_value(pdata->amp_gpio, to); - - /* if we're umuting, also re-set the gain */ - if (to && pdata->amp_gain[0] > 0) - speaker_gain_set(spk_gain); -} - -/** - * speaker_unmute_get - read the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_unmute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_unmute; - return 0; -} - -/** - * speaker_unmute_put - set the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - */ -static int speaker_unmute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - spk_unmute_state(ucontrol->value.integer.value[0]); - return 0; -} - -/* This is added as a manual control as the speaker amps create clicks - * when their power state is changed, which are far more noticeable than - * anything produced by the CODEC itself. - */ -static const struct snd_kcontrol_new amp_unmute_controls[] = { - SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, - speaker_unmute_get, speaker_unmute_put), -}; - -void simtec_audio_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - - if (pdata->amp_gpio > 0) { - pr_debug("%s: adding amp routes\n", __func__); - - snd_soc_add_card_controls(card, amp_unmute_controls, - ARRAY_SIZE(amp_unmute_controls)); - } - - if (pdata->amp_gain[0] > 0) { - pr_debug("%s: adding amp controls\n", __func__); - snd_soc_add_card_controls(card, amp_gain_controls, - ARRAY_SIZE(amp_gain_controls)); - } -} -EXPORT_SYMBOL_GPL(simtec_audio_init); - -#define CODEC_CLOCK 12000000 - -/** - * simtec_hw_params - update hardware parameters - * @substream: The audio substream instance. - * @params: The parameters requested. - * - * Update the codec data routing and configuration settings - * from the supplied data. - */ -static int simtec_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; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret) { - pr_err( "%s: failed setting codec sysclk\n", __func__); - return ret; - } - - if (pdata->use_mpllin) { - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, - 0, SND_SOC_CLOCK_OUT); - - if (ret) { - pr_err("%s: failed to set MPLLin as clksrc\n", - __func__); - return ret; - } - } - - if (pdata->output_cdclk) { - int cdclk_scale; - - cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; - cdclk_scale--; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - cdclk_scale); - if (ret) { - pr_err("%s: failed to set clock div\n", - __func__); - return ret; - } - } - - return 0; -} - -static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) -{ - /* call any board supplied startup code, this currently only - * covers the bast/vr1000 which have a CPLD in the way of the - * LRCLK */ - if (pd->startup) - pd->startup(); - - return 0; -} - -static const struct snd_soc_ops simtec_snd_ops = { - .hw_params = simtec_hw_params, -}; - -/** - * attach_gpio_amp - get and configure the necessary gpios - * @dev: The device we're probing. - * @pd: The platform data supplied by the board. - * - * If there is a GPIO based amplifier attached to the board, claim - * the necessary GPIO lines for it, and set default values. - */ -static int attach_gpio_amp(struct device *dev, - struct s3c24xx_audio_simtec_pdata *pd) -{ - int ret; - - /* attach gpio amp gain (if any) */ - if (pdata->amp_gain[0] > 0) { - ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain0\n"); - return ret; - } - - ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain1\n"); - gpio_free(pdata->amp_gain[0]); - return ret; - } - - gpio_direction_output(pd->amp_gain[0], 0); - gpio_direction_output(pd->amp_gain[1], 0); - } - - /* note, currently we assume GPA0 isn't valid amp */ - if (pdata->amp_gpio > 0) { - ret = gpio_request(pd->amp_gpio, "gpio-amp"); - if (ret) { - dev_err(dev, "cannot get amp gpio %d (%d)\n", - pd->amp_gpio, ret); - goto err_amp; - } - - /* set the amp off at startup */ - spk_unmute_state(0); - } - - return 0; - -err_amp: - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - return ret; -} - -static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) -{ - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - if (pd->amp_gpio > 0) - gpio_free(pd->amp_gpio); -} - -#ifdef CONFIG_PM -static int simtec_audio_resume(struct device *dev) -{ - simtec_call_startup(pdata); - return 0; -} - -const struct dev_pm_ops simtec_audio_pmops = { - .resume = simtec_audio_resume, -}; -EXPORT_SYMBOL_GPL(simtec_audio_pmops); -#endif - -int simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_card *card) -{ - struct platform_device *snd_dev; - int ret; - - card->dai_link->ops = &simtec_snd_ops; - card->dai_link->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - pdata = pdev->dev.platform_data; - if (!pdata) { - dev_err(&pdev->dev, "no platform data supplied\n"); - return -EINVAL; - } - - simtec_call_startup(pdata); - - xtal_clk = clk_get(&pdev->dev, "xtal"); - if (IS_ERR(xtal_clk)) { - dev_err(&pdev->dev, "could not get clkout0\n"); - return -EINVAL; - } - - dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); - - ret = attach_gpio_amp(&pdev->dev, pdata); - if (ret) - goto err_clk; - - snd_dev = platform_device_alloc("soc-audio", -1); - if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio device\n"); - ret = -ENOMEM; - goto err_gpio; - } - - platform_set_drvdata(snd_dev, card); - - ret = platform_device_add(snd_dev); - if (ret) { - dev_err(&pdev->dev, "failed to add soc-audio dev\n"); - goto err_pdev; - } - - platform_set_drvdata(pdev, snd_dev); - return 0; - -err_pdev: - platform_device_put(snd_dev); - -err_gpio: - detach_gpio_amp(pdata); - -err_clk: - clk_put(xtal_clk); - return ret; -} -EXPORT_SYMBOL_GPL(simtec_audio_core_probe); - -int simtec_audio_remove(struct platform_device *pdev) -{ - struct platform_device *snd_dev = platform_get_drvdata(pdev); - - platform_device_unregister(snd_dev); - - detach_gpio_amp(pdata); - clk_put(xtal_clk); - return 0; -} -EXPORT_SYMBOL_GPL(simtec_audio_remove); - -MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec.h b/sound/soc/samsung/s3c24xx_simtec.h deleted file mode 100644 index 38d8384755cd..000000000000 --- a/sound/soc/samsung/s3c24xx_simtec.h +++ /dev/null @@ -1,18 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - * Copyright 2009 Simtec Electronics - */ - -extern void simtec_audio_init(struct snd_soc_pcm_runtime *rtd); - -extern int simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_card *card); - -extern int simtec_audio_remove(struct platform_device *pdev); - -#ifdef CONFIG_PM -extern const struct dev_pm_ops simtec_audio_pmops; -#define simtec_audio_pm &simtec_audio_pmops -#else -#define simtec_audio_pm NULL -#endif diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c deleted file mode 100644 index ed0d1b8fa2d4..000000000000 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ /dev/null @@ -1,112 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Copyright 2009 Simtec Electronics - -#include <linux/module.h> -#include <sound/soc.h> - -#include "s3c24xx_simtec.h" - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Out", NULL), - SND_SOC_DAPM_LINE("GSM In", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("ZV", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ - - { "Headphone Jack", NULL, "HPLOUT" }, - { "Headphone Jack", NULL, "HPLCOM" }, - { "Headphone Jack", NULL, "HPROUT" }, - { "Headphone Jack", NULL, "HPRCOM" }, - - /* ZV connected to Line1 */ - - { "LINE1L", NULL, "ZV" }, - { "LINE1R", NULL, "ZV" }, - - /* Line In connected to Line2 */ - - { "LINE2L", NULL, "Line In" }, - { "LINE2R", NULL, "Line In" }, - - /* Microphone connected to MIC3R and MIC_BIAS */ - - { "MIC3L", NULL, "Mic Jack" }, - - /* GSM connected to MONO_LOUT and MIC3L (in) */ - - { "GSM Out", NULL, "MONO_LOUT" }, - { "MIC3L", NULL, "GSM In" }, - - /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are - * not using the DAPM to power it up and down as there it makes - * a click when powering up. */ -}; - -/** - * simtec_hermes_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) -{ - simtec_audio_init(rtd); - - return 0; -} - -SND_SOC_DAILINK_DEFS(tlv320aic33, - DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.0-001a", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis"))); - -static struct snd_soc_dai_link simtec_dai_aic33 = { - .name = "tlv320aic33", - .stream_name = "TLV320AIC33", - .init = simtec_hermes_init, - SND_SOC_DAILINK_REG(tlv320aic33), -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic33 = { - .name = "Simtec-Hermes", - .owner = THIS_MODULE, - .dai_link = &simtec_dai_aic33, - .num_links = 1, - - .dapm_widgets = dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), - .dapm_routes = base_map, - .num_dapm_routes = ARRAY_SIZE(base_map), -}; - -static int simtec_audio_hermes_probe(struct platform_device *pd) -{ - dev_info(&pd->dev, "probing....\n"); - return simtec_audio_core_probe(pd, &snd_soc_machine_simtec_aic33); -} - -static struct platform_driver simtec_audio_hermes_platdrv = { - .driver = { - .name = "s3c24xx-simtec-hermes-snd", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_hermes_probe, - .remove = simtec_audio_remove, -}; - -module_platform_driver(simtec_audio_hermes_platdrv); - -MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); -MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c deleted file mode 100644 index c03d52990267..000000000000 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ /dev/null @@ -1,100 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Copyright 2009 Simtec Electronics - -#include <linux/module.h> -#include <sound/soc.h> - -#include "s3c24xx_simtec.h" - -/* supported machines: - * - * Machine Connections AMP - * ------- ----------- --- - * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) - * VR1000 HPOUT, LIN None - * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) - */ - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - { "Headphone Jack", NULL, "LHPOUT"}, - { "Headphone Jack", NULL, "RHPOUT"}, - - { "Line Out", NULL, "LOUT" }, - { "Line Out", NULL, "ROUT" }, - - { "LLINEIN", NULL, "Line In"}, - { "RLINEIN", NULL, "Line In"}, - - { "MICIN", NULL, "Mic Jack"}, -}; - -/** - * simtec_tlv320aic23_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - simtec_audio_init(rtd); - - return 0; -} - -SND_SOC_DAILINK_DEFS(tlv320aic23, - DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.0-001a", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis"))); - -static struct snd_soc_dai_link simtec_dai_aic23 = { - .name = "tlv320aic23", - .stream_name = "TLV320AIC23", - .init = simtec_tlv320aic23_init, - SND_SOC_DAILINK_REG(tlv320aic23), -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic23 = { - .name = "Simtec", - .owner = THIS_MODULE, - .dai_link = &simtec_dai_aic23, - .num_links = 1, - - .dapm_widgets = dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dapm_widgets), - .dapm_routes = base_map, - .num_dapm_routes = ARRAY_SIZE(base_map), -}; - -static int simtec_audio_tlv320aic23_probe(struct platform_device *pd) -{ - return simtec_audio_core_probe(pd, &snd_soc_machine_simtec_aic23); -} - -static struct platform_driver simtec_audio_tlv320aic23_driver = { - .driver = { - .name = "s3c24xx-simtec-tlv320aic23", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_tlv320aic23_probe, - .remove = simtec_audio_remove, -}; - -module_platform_driver(simtec_audio_tlv320aic23_driver); - -MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); -MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c deleted file mode 100644 index 6272070dcd92..000000000000 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ /dev/null @@ -1,257 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Modifications by Christian Pellegrin <chripell@evolware.org> -// -// s3c24xx_uda134x.c - S3C24XX_UDA134X ALSA SoC Audio board driver -// -// Copyright 2007 Dension Audio Systems Ltd. -// Author: Zoltan Devai - -#include <linux/clk.h> -#include <linux/gpio.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/s3c24xx_uda134x.h> - -#include "regs-iis.h" -#include "s3c24xx-i2s.h" - -struct s3c24xx_uda134x { - struct clk *xtal; - struct clk *pclk; - struct mutex clk_lock; - int clk_users; -}; - -/* #define ENFORCE_RATES 1 */ -/* - Unfortunately the S3C24XX in master mode has a limited capacity of - generating the clock for the codec. If you define this only rates - that are really available will be enforced. But be careful, most - user level application just want the usual sampling frequencies (8, - 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly - operation for embedded systems. So if you aren't very lucky or your - hardware engineer wasn't very forward-looking it's better to leave - this undefined. If you do so an approximate value for the requested - sampling rate in the range -/+ 5% will be chosen. If this in not - possible an error will be returned. -*/ - -static unsigned int rates[33 * 2]; -#ifdef ENFORCE_RATES -static const struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; -#endif - -static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int ret = 0; - - mutex_lock(&priv->clk_lock); - - if (priv->clk_users == 0) { - priv->xtal = clk_get(rtd->dev, "xtal"); - if (IS_ERR(priv->xtal)) { - dev_err(rtd->dev, "%s cannot get xtal\n", __func__); - ret = PTR_ERR(priv->xtal); - } else { - priv->pclk = clk_get(cpu_dai->dev, "iis"); - if (IS_ERR(priv->pclk)) { - dev_err(rtd->dev, "%s cannot get pclk\n", - __func__); - clk_put(priv->xtal); - ret = PTR_ERR(priv->pclk); - } - } - if (!ret) { - int i, j; - - for (i = 0; i < 2; i++) { - int fs = i ? 256 : 384; - - rates[i*33] = clk_get_rate(priv->xtal) / fs; - for (j = 1; j < 33; j++) - rates[i*33 + j] = clk_get_rate(priv->pclk) / - (j * fs); - } - } - } - priv->clk_users += 1; - mutex_unlock(&priv->clk_lock); - - if (!ret) { -#ifdef ENFORCE_RATES - ret = snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates); - if (ret < 0) - dev_err(rtd->dev, "%s cannot set constraints\n", - __func__); -#endif - } - return ret; -} - -static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card); - - mutex_lock(&priv->clk_lock); - priv->clk_users -= 1; - if (priv->clk_users == 0) { - clk_put(priv->xtal); - priv->xtal = NULL; - clk_put(priv->pclk); - priv->pclk = NULL; - } - mutex_unlock(&priv->clk_lock); -} - -static int s3c24xx_uda134x_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; - int clk_source, fs_mode; - unsigned long rate = params_rate(params); - long err, cerr; - unsigned int div; - int i, bi; - - err = 999999; - bi = 0; - for (i = 0; i < 2*33; i++) { - cerr = rates[i] - rate; - if (cerr < 0) - cerr = -cerr; - if (cerr < err) { - err = cerr; - bi = i; - } - } - if (bi / 33 == 1) - fs_mode = S3C2410_IISMOD_256FS; - else - fs_mode = S3C2410_IISMOD_384FS; - if (bi % 33 == 0) { - clk_source = S3C24XX_CLKSRC_MPLL; - div = 1; - } else { - clk_source = S3C24XX_CLKSRC_PCLK; - div = bi % 33; - } - - dev_dbg(rtd->dev, "%s desired rate %lu, %d\n", __func__, rate, bi); - - clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; - - dev_dbg(rtd->dev, "%s will use: %s %s %d sysclk %d err %ld\n", __func__, - fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", - clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", - div, clk, err); - - if ((err * 100 / rate) > 5) { - dev_err(rtd->dev, "effective frequency too different " - "from desired (%ld%%)\n", err * 100 / rate); - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(div, div)); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops s3c24xx_uda134x_ops = { - .startup = s3c24xx_uda134x_startup, - .shutdown = s3c24xx_uda134x_shutdown, - .hw_params = s3c24xx_uda134x_hw_params, -}; - -SND_SOC_DAILINK_DEFS(uda134x, - DAILINK_COMP_ARRAY(COMP_CPU("s3c24xx-iis")), - DAILINK_COMP_ARRAY(COMP_CODEC("uda134x-codec", "uda134x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c24xx-iis"))); - -static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { - .name = "UDA134X", - .stream_name = "UDA134X", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &s3c24xx_uda134x_ops, - SND_SOC_DAILINK_REG(uda134x), -}; - -static struct snd_soc_card snd_soc_s3c24xx_uda134x = { - .name = "S3C24XX_UDA134X", - .owner = THIS_MODULE, - .dai_link = &s3c24xx_uda134x_dai_link, - .num_links = 1, -}; - -static int s3c24xx_uda134x_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_s3c24xx_uda134x; - struct s3c24xx_uda134x *priv; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - mutex_init(&priv->clk_lock); - - card->dev = &pdev->dev; - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "failed to register card: %d\n", ret); - - return ret; -} - -static struct platform_driver s3c24xx_uda134x_driver = { - .probe = s3c24xx_uda134x_probe, - .driver = { - .name = "s3c24xx_uda134x", - }, -}; -module_platform_driver(s3c24xx_uda134x_driver); - -MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>"); -MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c deleted file mode 100644 index 29bf917242fe..000000000000 --- a/sound/soc/samsung/smartq_wm8987.c +++ /dev/null @@ -1,224 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// Copyright 2010 Maurus Cuelenaere <mcuelenaere@gmail.com> -// -// Based on smdk6410_wm8987.c -// Copyright 2007 Wolfson Microelectronics PLC. - linux@wolfsonmicro.com -// Graeme Gregory - graeme.gregory@wolfsonmicro.com - -#include <linux/gpio/consumer.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/jack.h> - -#include "i2s.h" -#include "../codecs/wm8750.h" - -/* - * WM8987 is register compatible with WM8750, so using that as base driver. - */ - -static struct snd_soc_card snd_soc_smartq; - -static int smartq_hifi_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 32000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - case 88200: - clk = 11289600; - break; - } - - /* Use PCLK for I2S signal generation */ - ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Gate the RCLK output on PAD */ - ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* 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; - - return 0; -} - -/* - * SmartQ WM8987 HiFi DAI operations. - */ -static const struct snd_soc_ops smartq_hifi_ops = { - .hw_params = smartq_hifi_hw_params, -}; - -static struct snd_soc_jack smartq_jack; - -static struct snd_soc_jack_pin smartq_jack_pins[] = { - /* Disable speaker when headphone is plugged in */ - { - .pin = "Internal Speaker", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static struct snd_soc_jack_gpio smartq_jack_gpios[] = { - { - .gpio = -1, - .name = "headphone detect", - .report = SND_JACK_HEADPHONE, - .debounce_time = 200, - }, -}; - -static const struct snd_kcontrol_new wm8987_smartq_controls[] = { - SOC_DAPM_PIN_SWITCH("Internal Speaker"), - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Internal Mic"), -}; - -static int smartq_speaker_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, - int event) -{ - struct gpio_desc *gpio = snd_soc_card_get_drvdata(&snd_soc_smartq); - - gpiod_set_value(gpio, SND_SOC_DAPM_EVENT_OFF(event)); - - return 0; -} - -static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Internal Mic", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"Headphone Jack", NULL, "LOUT2"}, - {"Headphone Jack", NULL, "ROUT2"}, - - {"Internal Speaker", NULL, "LOUT2"}, - {"Internal Speaker", NULL, "ROUT2"}, - - {"Mic Bias", NULL, "Internal Mic"}, - {"LINPUT2", NULL, "Mic Bias"}, -}; - -static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm = &rtd->card->dapm; - int err = 0; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "LINPUT1"); - snd_soc_dapm_nc_pin(dapm, "RINPUT1"); - snd_soc_dapm_nc_pin(dapm, "OUT3"); - snd_soc_dapm_nc_pin(dapm, "ROUT1"); - - /* Headphone jack detection */ - err = snd_soc_card_jack_new_pins(rtd->card, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack, - smartq_jack_pins, - ARRAY_SIZE(smartq_jack_pins)); - if (err) - return err; - - err = snd_soc_jack_add_gpios(&smartq_jack, - ARRAY_SIZE(smartq_jack_gpios), - smartq_jack_gpios); - - return err; -} - -SND_SOC_DAILINK_DEFS(wm8987, - DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8750.0-0x1a", "wm8750-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0"))); - -static struct snd_soc_dai_link smartq_dai[] = { - { - .name = "wm8987", - .stream_name = "SmartQ Hi-Fi", - .init = smartq_wm8987_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &smartq_hifi_ops, - SND_SOC_DAILINK_REG(wm8987), - }, -}; - -static struct snd_soc_card snd_soc_smartq = { - .name = "SmartQ", - .owner = THIS_MODULE, - .dai_link = smartq_dai, - .num_links = ARRAY_SIZE(smartq_dai), - - .dapm_widgets = wm8987_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8987_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .controls = wm8987_smartq_controls, - .num_controls = ARRAY_SIZE(wm8987_smartq_controls), -}; - -static int smartq_probe(struct platform_device *pdev) -{ - struct gpio_desc *gpio; - int ret; - - platform_set_drvdata(pdev, &snd_soc_smartq); - - /* Initialise GPIOs used by amplifiers */ - gpio = devm_gpiod_get(&pdev->dev, "amplifiers shutdown", - GPIOD_OUT_HIGH); - if (IS_ERR(gpio)) { - dev_err(&pdev->dev, "Failed to register GPK12\n"); - ret = PTR_ERR(gpio); - goto out; - } - snd_soc_card_set_drvdata(&snd_soc_smartq, gpio); - - ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_smartq); - if (ret) - dev_err(&pdev->dev, "Failed to register card\n"); - -out: - return ret; -} - -static struct platform_driver smartq_driver = { - .driver = { - .name = "smartq-audio", - }, - .probe = smartq_probe, -}; - -module_platform_driver(smartq_driver); - -/* Module information */ -MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere@gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c deleted file mode 100644 index 78703d095a6f..000000000000 --- a/sound/soc/samsung/smdk_wm8580.c +++ /dev/null @@ -1,211 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// Copyright (c) 2009 Samsung Electronics Co. Ltd -// Author: Jaswinder Singh <jassisinghbrar@gmail.com> - -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/wm8580.h" -#include "i2s.h" - -/* - * Default CFG switch settings to use this driver: - * - * SMDK6410: Set CFG1 1-3 Off, CFG2 1-4 On - */ - -/* SMDK has a 12MHZ crystal attached to WM8580 */ -#define SMDK_WM8580_FREQ 12000000 - -static int smdk_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); - unsigned int pll_out; - int rfs, ret; - - switch (params_width(params)) { - case 8: - case 16: - break; - default: - return -EINVAL; - } - - /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. - * This criterion can't be met if we request PLL output - * as {8000x256, 64000x256, 11025x256}Hz. - * As a wayout, we rather change rfs to a minimum value that - * results in (params_rate(params) * rfs), and itself, acceptable - * to both - the CODEC and the CPU. - */ - switch (params_rate(params)) { - case 16000: - case 22050: - case 32000: - case 44100: - case 48000: - case 88200: - case 96000: - rfs = 256; - break; - case 64000: - rfs = 384; - break; - case 8000: - case 11025: - rfs = 512; - break; - default: - return -EINVAL; - } - pll_out = params_rate(params) * rfs; - - /* Set WM8580 to drive MCLK from its PLLA */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - SMDK_WM8580_FREQ, pll_out); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_PLLA, - pll_out, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -/* - * SMDK WM8580 DAI operations. - */ -static const struct snd_soc_ops smdk_ops = { - .hw_params = smdk_hw_params, -}; - -/* SMDK Playback widgets */ -static const struct snd_soc_dapm_widget smdk_wm8580_dapm_widgets[] = { - SND_SOC_DAPM_HP("Front", NULL), - SND_SOC_DAPM_HP("Center+Sub", NULL), - SND_SOC_DAPM_HP("Rear", NULL), - - SND_SOC_DAPM_MIC("MicIn", NULL), - SND_SOC_DAPM_LINE("LineIn", NULL), -}; - -/* SMDK-PAIFTX connections */ -static const struct snd_soc_dapm_route smdk_wm8580_audio_map[] = { - /* MicIn feeds AINL */ - {"AINL", NULL, "MicIn"}, - - /* LineIn feeds AINL/R */ - {"AINL", NULL, "LineIn"}, - {"AINR", NULL, "LineIn"}, - - /* Front Left/Right are fed VOUT1L/R */ - {"Front", NULL, "VOUT1L"}, - {"Front", NULL, "VOUT1R"}, - - /* Center/Sub are fed VOUT2L/R */ - {"Center+Sub", NULL, "VOUT2L"}, - {"Center+Sub", NULL, "VOUT2R"}, - - /* Rear Left/Right are fed VOUT3L/R */ - {"Rear", NULL, "VOUT3L"}, - {"Rear", NULL, "VOUT3R"}, -}; - -static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) -{ - /* Enabling the microphone requires the fitting of a 0R - * resistor to connect the line from the microphone jack. - */ - snd_soc_dapm_disable_pin(&rtd->card->dapm, "MicIn"); - - return 0; -} - -enum { - PRI_PLAYBACK = 0, - PRI_CAPTURE, -}; - -#define SMDK_DAI_FMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBM_CFM) - -SND_SOC_DAILINK_DEFS(paif_rx, - DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.2")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8580.0-001b", "wm8580-hifi-playback")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0"))); - -SND_SOC_DAILINK_DEFS(paif_tx, - DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.2")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8580.0-001b", "wm8580-hifi-capture")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0"))); - -static struct snd_soc_dai_link smdk_dai[] = { - [PRI_PLAYBACK] = { /* Primary Playback i/f */ - .name = "WM8580 PAIF RX", - .stream_name = "Playback", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_ops, - SND_SOC_DAILINK_REG(paif_rx), - }, - [PRI_CAPTURE] = { /* Primary Capture i/f */ - .name = "WM8580 PAIF TX", - .stream_name = "Capture", - .dai_fmt = SMDK_DAI_FMT, - .init = smdk_wm8580_init_paiftx, - .ops = &smdk_ops, - SND_SOC_DAILINK_REG(paif_tx), - }, -}; - -static struct snd_soc_card smdk = { - .name = "SMDK-I2S", - .owner = THIS_MODULE, - .dai_link = smdk_dai, - .num_links = ARRAY_SIZE(smdk_dai), - - .dapm_widgets = smdk_wm8580_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(smdk_wm8580_dapm_widgets), - .dapm_routes = smdk_wm8580_audio_map, - .num_dapm_routes = ARRAY_SIZE(smdk_wm8580_audio_map), -}; - -static struct platform_device *smdk_snd_device; - -static int __init smdk_audio_init(void) -{ - int ret; - - smdk_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_device) - return -ENOMEM; - - platform_set_drvdata(smdk_snd_device, &smdk); - ret = platform_device_add(smdk_snd_device); - - if (ret) - platform_device_put(smdk_snd_device); - - return ret; -} -module_init(smdk_audio_init); - -static void __exit smdk_audio_exit(void) -{ - platform_device_unregister(smdk_snd_device); -} -module_exit(smdk_audio_exit); - -MODULE_AUTHOR("Jaswinder Singh, jassisinghbrar@gmail.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK WM8580"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 40110e9a9e8a..593be22503b5 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -40,13 +40,6 @@ config SND_SOC_DAVINCI_MCASP - Keystone devices - K3 devices (am654, j721e) -config SND_SOC_DAVINCI_VCIF - tristate "daVinci Voice Interface (VCIF) support" - depends on ARCH_DAVINCI || COMPILE_TEST - select SND_SOC_TI_EDMA_PCM - help - Say Y or M here if you want audio support via daVinci VCIF. - config SND_SOC_OMAP_DMIC tristate "Digital Microphone Module (DMIC) support" depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST && COMMON_CLK @@ -177,14 +170,6 @@ config SND_SOC_OMAP_OSK5912 config SND_SOC_DAVINCI_EVM tristate "SoC Audio support for DaVinci EVMs" depends on ARCH_DAVINCI && I2C - select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_DM355_EVM - select SND_SOC_DAVINCI_ASP if SND_SOC_DM365_AIC3X_CODEC - select SND_SOC_DAVINCI_VCIF if SND_SOC_DM365_VOICE_CODEC - select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_EVM # DM6446 - select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DM6467_EVM - select SND_SOC_SPDIF if MACH_DAVINCI_DM6467_EVM - select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA830_EVM - select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA850_EVM select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on the following TI @@ -196,31 +181,6 @@ config SND_SOC_DAVINCI_EVM - DM830 - DM850 -choice - prompt "DM365 codec select" - depends on SND_SOC_DAVINCI_EVM - depends on MACH_DAVINCI_DM365_EVM - -config SND_SOC_DM365_AIC3X_CODEC - bool "Audio Codec - AIC3101" - help - Say Y if you want to add support for AIC3101 audio codec - -config SND_SOC_DM365_VOICE_CODEC - bool "Voice Codec - CQ93VC" - help - Say Y if you want to add support for SoC On-chip voice codec -endchoice - -config SND_SOC_DM365_SELECT_VOICE_CODECS - def_tristate y - depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC - select MFD_DAVINCI_VOICECODEC - select SND_SOC_CQ0093VC - help - The is an internal symbol needed to ensure that the codec - and MFD driver can be built as loadable modules if necessary. - config SND_SOC_J721E_EVM tristate "SoC Audio support for j721e EVM" depends on ARCH_K3 || COMPILE_TEST && COMMON_CLK diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile index a21e5b0061de..41cdcaec770d 100644 --- a/sound/soc/ti/Makefile +++ b/sound/soc/ti/Makefile @@ -12,14 +12,12 @@ obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o # CPU DAI drivers snd-soc-davinci-asp-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs := davinci-mcasp.o -snd-soc-davinci-vcif-objs := davinci-vcif.o snd-soc-omap-dmic-objs := omap-dmic.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o obj-$(CONFIG_SND_SOC_DAVINCI_ASP) += snd-soc-davinci-asp.o obj-$(CONFIG_SND_SOC_DAVINCI_MCASP) += snd-soc-davinci-mcasp.o -obj-$(CONFIG_SND_SOC_DAVINCI_VCIF) += snd-soc-davinci-vcif.o obj-$(CONFIG_SND_SOC_OMAP_DMIC) += snd-soc-omap-dmic.o obj-$(CONFIG_SND_SOC_OMAP_MCBSP) += snd-soc-omap-mcbsp.o obj-$(CONFIG_SND_SOC_OMAP_MCPDM) += snd-soc-omap-mcpdm.o diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 68d69e32681a..983d69b951b0 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -138,214 +138,6 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* davinci-evm digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEFS(dm6446, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-001b", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp"))); - -static struct snd_soc_dai_link dm6446_evm_dai = { - .name = "TLV320AIC3X", - .stream_name = "AIC3X", - .init = evm_aic3x_init, - .ops = &evm_ops, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(dm6446), -}; - -SND_SOC_DAILINK_DEFS(dm355, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp.1")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-001b", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp.1"))); - -static struct snd_soc_dai_link dm355_evm_dai = { - .name = "TLV320AIC3X", - .stream_name = "AIC3X", - .init = evm_aic3x_init, - .ops = &evm_ops, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(dm355), -}; - -#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC -SND_SOC_DAILINK_DEFS(dm365, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp"))); -#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC) -SND_SOC_DAILINK_DEFS(dm365, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-vcif")), - DAILINK_COMP_ARRAY(COMP_CODEC("cq93vc-codec", "cq93vc-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-vcif"))); -#endif - -static struct snd_soc_dai_link dm365_evm_dai = { -#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC - .name = "TLV320AIC3X", - .stream_name = "AIC3X", - .init = evm_aic3x_init, - .ops = &evm_ops, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(dm365), -#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC) - .name = "Voice Codec - CQ93VC", - .stream_name = "CQ93", - SND_SOC_DAILINK_REG(dm365), -#endif -}; - -SND_SOC_DAILINK_DEFS(dm6467_aic3x, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.0")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.0-001a", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.0"))); - -SND_SOC_DAILINK_DEFS(dm6467_spdif, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.1")), - DAILINK_COMP_ARRAY(COMP_CODEC("spdif_dit", "dit-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.1"))); - -static struct snd_soc_dai_link dm6467_evm_dai[] = { - { - .name = "TLV320AIC3X", - .stream_name = "AIC3X", - .init = evm_aic3x_init, - .ops = &evm_ops, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(dm6467_aic3x), - }, - { - .name = "McASP", - .stream_name = "spdif", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(dm6467_spdif), - }, -}; - -SND_SOC_DAILINK_DEFS(da830, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.1")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.1"))); - -static struct snd_soc_dai_link da830_evm_dai = { - .name = "TLV320AIC3X", - .stream_name = "AIC3X", - .init = evm_aic3x_init, - .ops = &evm_ops, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(da830), -}; - -SND_SOC_DAILINK_DEFS(da850, - DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.0")), - DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018", - "tlv320aic3x-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.0"))); - -static struct snd_soc_dai_link da850_evm_dai = { - .name = "TLV320AIC3X", - .stream_name = "AIC3X", - .init = evm_aic3x_init, - .ops = &evm_ops, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, - SND_SOC_DAILINK_REG(da850), -}; - -/* davinci dm6446 evm audio machine driver */ -/* - * ASP0 in DM6446 EVM is clocked by U55, as configured by - * board-dm644x-evm.c using GPIOs from U18. There are six - * options; here we "know" we use a 48 KHz sample rate. - */ -static struct snd_soc_card_drvdata_davinci dm6446_snd_soc_card_drvdata = { - .sysclk = 12288000, -}; - -static struct snd_soc_card dm6446_snd_soc_card_evm = { - .name = "DaVinci DM6446 EVM", - .owner = THIS_MODULE, - .dai_link = &dm6446_evm_dai, - .num_links = 1, - .drvdata = &dm6446_snd_soc_card_drvdata, -}; - -/* davinci dm355 evm audio machine driver */ -/* ASP1 on DM355 EVM is clocked by an external oscillator */ -static struct snd_soc_card_drvdata_davinci dm355_snd_soc_card_drvdata = { - .sysclk = 27000000, -}; - -static struct snd_soc_card dm355_snd_soc_card_evm = { - .name = "DaVinci DM355 EVM", - .owner = THIS_MODULE, - .dai_link = &dm355_evm_dai, - .num_links = 1, - .drvdata = &dm355_snd_soc_card_drvdata, -}; - -/* davinci dm365 evm audio machine driver */ -static struct snd_soc_card_drvdata_davinci dm365_snd_soc_card_drvdata = { - .sysclk = 27000000, -}; - -static struct snd_soc_card dm365_snd_soc_card_evm = { - .name = "DaVinci DM365 EVM", - .owner = THIS_MODULE, - .dai_link = &dm365_evm_dai, - .num_links = 1, - .drvdata = &dm365_snd_soc_card_drvdata, -}; - -/* davinci dm6467 evm audio machine driver */ -static struct snd_soc_card_drvdata_davinci dm6467_snd_soc_card_drvdata = { - .sysclk = 27000000, -}; - -static struct snd_soc_card dm6467_snd_soc_card_evm = { - .name = "DaVinci DM6467 EVM", - .owner = THIS_MODULE, - .dai_link = dm6467_evm_dai, - .num_links = ARRAY_SIZE(dm6467_evm_dai), - .drvdata = &dm6467_snd_soc_card_drvdata, -}; - -static struct snd_soc_card_drvdata_davinci da830_snd_soc_card_drvdata = { - .sysclk = 24576000, -}; - -static struct snd_soc_card da830_snd_soc_card = { - .name = "DA830/OMAP-L137 EVM", - .owner = THIS_MODULE, - .dai_link = &da830_evm_dai, - .num_links = 1, - .drvdata = &da830_snd_soc_card_drvdata, -}; - -static struct snd_soc_card_drvdata_davinci da850_snd_soc_card_drvdata = { - .sysclk = 24576000, -}; - -static struct snd_soc_card da850_snd_soc_card = { - .name = "DA850/OMAP-L138 EVM", - .owner = THIS_MODULE, - .dai_link = &da850_evm_dai, - .num_links = 1, - .drvdata = &da850_snd_soc_card_drvdata, -}; - -#if defined(CONFIG_OF) - /* * The struct is used as place holder. It will be completely * filled with data from dt node. @@ -461,71 +253,18 @@ static struct platform_driver davinci_evm_driver = { .driver = { .name = "davinci_evm", .pm = &snd_soc_pm_ops, - .of_match_table = of_match_ptr(davinci_evm_dt_ids), + .of_match_table = davinci_evm_dt_ids, }, }; -#endif - -static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct snd_soc_card *evm_snd_dev_data; - int index; - int ret; - - /* - * If dtb is there, the devices will be created dynamically. - * Only register platfrom driver structure. - */ -#if defined(CONFIG_OF) - if (of_have_populated_dt()) - return platform_driver_register(&davinci_evm_driver); -#endif - - if (machine_is_davinci_evm()) { - evm_snd_dev_data = &dm6446_snd_soc_card_evm; - index = 0; - } else if (machine_is_davinci_dm355_evm()) { - evm_snd_dev_data = &dm355_snd_soc_card_evm; - index = 1; - } else if (machine_is_davinci_dm365_evm()) { - evm_snd_dev_data = &dm365_snd_soc_card_evm; - index = 0; - } else if (machine_is_davinci_dm6467_evm()) { - evm_snd_dev_data = &dm6467_snd_soc_card_evm; - index = 0; - } else if (machine_is_davinci_da830_evm()) { - evm_snd_dev_data = &da830_snd_soc_card; - index = 1; - } else if (machine_is_davinci_da850_evm()) { - evm_snd_dev_data = &da850_snd_soc_card; - index = 0; - } else - return -EINVAL; - - evm_snd_device = platform_device_alloc("soc-audio", index); - if (!evm_snd_device) - return -ENOMEM; - - platform_set_drvdata(evm_snd_device, evm_snd_dev_data); - ret = platform_device_add(evm_snd_device); - if (ret) - platform_device_put(evm_snd_device); - - return ret; + return platform_driver_register(&davinci_evm_driver); } static void __exit evm_exit(void) { -#if defined(CONFIG_OF) - if (of_have_populated_dt()) { - platform_driver_unregister(&davinci_evm_driver); - return; - } -#endif - - platform_device_unregister(evm_snd_device); + platform_driver_unregister(&davinci_evm_driver); } module_init(evm_init); diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c deleted file mode 100644 index 36fa97e2b9e2..000000000000 --- a/sound/soc/ti/davinci-vcif.c +++ /dev/null @@ -1,247 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * ALSA SoC Voice Codec Interface for TI DAVINCI processor - * - * Copyright (C) 2010 Texas Instruments. - * - * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com> - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <linux/mfd/davinci_voicecodec.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <sound/dmaengine_pcm.h> - -#include "edma-pcm.h" -#include "davinci-i2s.h" - -#define MOD_REG_BIT(val, mask, set) do { \ - if (set) { \ - val |= mask; \ - } else { \ - val &= ~mask; \ - } \ -} while (0) - -struct davinci_vcif_dev { - struct davinci_vc *davinci_vc; - struct snd_dmaengine_dai_dma_data dma_data[2]; - int dma_request[2]; -}; - -static void davinci_vcif_start(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct davinci_vcif_dev *davinci_vcif_dev = - snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); - struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - u32 w; - - /* Start the sample generator and enable transmitter/receiver */ - w = readl(davinci_vc->base + DAVINCI_VC_CTRL); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); - else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); - - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); -} - -static void davinci_vcif_stop(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct davinci_vcif_dev *davinci_vcif_dev = - snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); - struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - u32 w; - - /* Reset transmitter/receiver and sample rate/frame sync generators */ - w = readl(davinci_vc->base + DAVINCI_VC_CTRL); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); - else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); - - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); -} - -static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); - struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - u32 w; - - /* Restart the codec before setup */ - davinci_vcif_stop(substream); - davinci_vcif_start(substream); - - /* General line settings */ - writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); - - writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); - - writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); - - w = readl(davinci_vc->base + DAVINCI_VC_CTRL); - - /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | - DAVINCI_VC_CTRL_RD_UNSIGNED | - DAVINCI_VC_CTRL_WD_BITS_8 | - DAVINCI_VC_CTRL_WD_UNSIGNED, 1); - break; - case SNDRV_PCM_FORMAT_S8: - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | - DAVINCI_VC_CTRL_WD_BITS_8, 1); - - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | - DAVINCI_VC_CTRL_WD_UNSIGNED, 0); - break; - case SNDRV_PCM_FORMAT_S16_LE: - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | - DAVINCI_VC_CTRL_RD_UNSIGNED | - DAVINCI_VC_CTRL_WD_BITS_8 | - DAVINCI_VC_CTRL_WD_UNSIGNED, 0); - break; - default: - printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); - return -EINVAL; - } - - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); - - return 0; -} - -static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - davinci_vcif_start(substream); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_vcif_stop(substream); - break; - default: - ret = -EINVAL; - } - - return ret; -} - -#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 - -static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { - .trigger = davinci_vcif_trigger, - .hw_params = davinci_vcif_hw_params, -}; - -static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); - - dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - - return 0; -} - -static struct snd_soc_dai_driver davinci_vcif_dai = { - .probe = davinci_vcif_dai_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = DAVINCI_VCIF_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = DAVINCI_VCIF_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_vcif_dai_ops, - -}; - -static const struct snd_soc_component_driver davinci_vcif_component = { - .name = "davinci-vcif", - .legacy_dai_naming = 1, -}; - -static int davinci_vcif_probe(struct platform_device *pdev) -{ - struct davinci_vc *davinci_vc = pdev->dev.platform_data; - struct davinci_vcif_dev *davinci_vcif_dev; - int ret; - - davinci_vcif_dev = devm_kzalloc(&pdev->dev, - sizeof(struct davinci_vcif_dev), - GFP_KERNEL); - if (!davinci_vcif_dev) - return -ENOMEM; - - /* DMA tx params */ - davinci_vcif_dev->davinci_vc = davinci_vc; - davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = - &davinci_vc->davinci_vcif.dma_tx_channel; - davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = - davinci_vc->davinci_vcif.dma_tx_addr; - - /* DMA rx params */ - davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = - &davinci_vc->davinci_vcif.dma_rx_channel; - davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = - davinci_vc->davinci_vcif.dma_rx_addr; - - dev_set_drvdata(&pdev->dev, davinci_vcif_dev); - - ret = devm_snd_soc_register_component(&pdev->dev, - &davinci_vcif_component, - &davinci_vcif_dai, 1); - if (ret != 0) { - dev_err(&pdev->dev, "could not register dai\n"); - return ret; - } - - ret = edma_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - return ret; - } - - return 0; -} - -static struct platform_driver davinci_vcif_driver = { - .probe = davinci_vcif_probe, - .driver = { - .name = "davinci-vcif", - }, -}; - -module_platform_driver(davinci_vcif_driver); - -MODULE_AUTHOR("Miguel Aguilar"); -MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); -MODULE_LICENSE("GPL"); |