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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-11 08:27:49 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-11 08:27:49 +0000 |
commit | ace9429bb58fd418f0c81d4c2835699bddf6bde6 (patch) | |
tree | b2d64bc10158fdd5497876388cd68142ca374ed3 /sound/soc/pxa | |
parent | Initial commit. (diff) | |
download | linux-ace9429bb58fd418f0c81d4c2835699bddf6bde6.tar.xz linux-ace9429bb58fd418f0c81d4c2835699bddf6bde6.zip |
Adding upstream version 6.6.15.upstream/6.6.15
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'sound/soc/pxa')
-rw-r--r-- | sound/soc/pxa/Kconfig | 58 | ||||
-rw-r--r-- | sound/soc/pxa/Makefile | 17 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-sspa.c | 585 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-sspa.h | 70 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 887 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.h | 36 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-ac97.c | 306 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-i2s.c | 412 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-i2s.h | 12 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-pcm.c | 49 | ||||
-rw-r--r-- | sound/soc/pxa/spitz.c | 322 |
11 files changed, 2754 insertions, 0 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig new file mode 100644 index 0000000000..e6bca90709 --- /dev/null +++ b/sound/soc/pxa/Kconfig @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0-only +config SND_PXA2XX_SOC + tristate "SoC Audio for the Intel PXA2xx chip" + depends on ARCH_PXA || COMPILE_TEST + select SND_PXA2XX_LIB + help + Say Y or M if you want to add support for codecs attached to + the PXA2xx AC97, I2S or SSP interface. You will also need + to select the audio interfaces to support below. + +config SND_PXA2XX_AC97 + tristate + +config SND_PXA2XX_SOC_AC97 + tristate "SoC AC97 support for PXA2xx" + depends on SND_PXA2XX_SOC + depends on AC97_BUS=n + default y + select AC97_BUS_NEW + select SND_PXA2XX_LIB + select SND_PXA2XX_LIB_AC97 + select SND_SOC_AC97_BUS_NEW + +config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB + tristate + +config SND_PXA_SOC_SSP + tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" + depends on PLAT_PXA + select PXA_SSP + select SND_PXA2XX_LIB + +config SND_MMP_SOC_SSPA + tristate "SoC Audio via MMP SSPA ports" + depends on ARCH_MMP + select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_ARM + help + Say Y if you want to add support for codecs attached to + the MMP SSPA interface. + +config SND_PXA2XX_SOC_SPITZ + tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" + depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C + select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8750 + help + Say Y if you want to add support for SoC audio on Sharp + Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita). + +config SND_PXA910_SOC + tristate "SoC Audio for Marvell PXA910 chip" + depends on ARCH_MMP && SND + select SND_PCM + help + Say Y if you want to add support for SoC audio on the + Marvell PXA910 reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile new file mode 100644 index 0000000000..406605fc74 --- /dev/null +++ b/sound/soc/pxa/Makefile @@ -0,0 +1,17 @@ +# SPDX-License-Identifier: GPL-2.0 +# PXA Platform Support +snd-soc-pxa2xx-objs := pxa2xx-pcm.o +snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o +snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o +snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-mmp-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-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o + +# PXA Machine Support +snd-soc-spitz-objs := spitz.o +obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c new file mode 100644 index 0000000000..abfaf3cdf5 --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.c @@ -0,0 +1,585 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * linux/sound/soc/pxa/mmp-sspa.c + * Base on pxa2xx-ssp.c + * + * Copyright (C) 2011 Marvell International Ltd. + */ +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/dmaengine.h> +#include <linux/pm_runtime.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> +#include "mmp-sspa.h" + +/* + * SSPA audio private data + */ +struct sspa_priv { + void __iomem *tx_base; + void __iomem *rx_base; + + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct clk *clk; + struct clk *audio_clk; + struct clk *sysclk; + + int running_cnt; + u32 sp; + u32 ctrl; +}; + +static void mmp_sspa_tx_enable(struct sspa_priv *sspa) +{ + unsigned int sspa_sp = sspa->sp; + + sspa_sp &= ~SSPA_SP_MSL; + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + __raw_writel(sspa_sp, sspa->tx_base + SSPA_SP); +} + +static void mmp_sspa_tx_disable(struct sspa_priv *sspa) +{ + unsigned int sspa_sp = sspa->sp; + + sspa_sp &= ~SSPA_SP_MSL; + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + __raw_writel(sspa_sp, sspa->tx_base + SSPA_SP); +} + +static void mmp_sspa_rx_enable(struct sspa_priv *sspa) +{ + unsigned int sspa_sp = sspa->sp; + + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + __raw_writel(sspa_sp, sspa->rx_base + SSPA_SP); +} + +static void mmp_sspa_rx_disable(struct sspa_priv *sspa) +{ + unsigned int sspa_sp = sspa->sp; + + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + __raw_writel(sspa_sp, sspa->rx_base + SSPA_SP); +} + +static int mmp_sspa_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(dai); + + clk_prepare_enable(sspa->sysclk); + clk_prepare_enable(sspa->clk); + + return 0; +} + +static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(sspa->clk); + clk_disable_unprepare(sspa->sysclk); +} + +/* + * Set the SSP ports SYSCLK. + */ +static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = cpu_dai->component->dev; + int ret = 0; + + if (dev->of_node) + return -ENOTSUPP; + + switch (clk_id) { + case MMP_SSPA_CLK_AUDIO: + ret = clk_set_rate(sspa->audio_clk, freq); + if (ret) + return ret; + break; + case MMP_SSPA_CLK_PLL: + case MMP_SSPA_CLK_VCXO: + /* not support yet */ + return -EINVAL; + default: + return -EINVAL; + } + + return 0; +} + +static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = cpu_dai->component->dev; + int ret = 0; + + if (dev->of_node) + return -ENOTSUPP; + + switch (pll_id) { + case MMP_SYSCLK: + ret = clk_set_rate(sspa->sysclk, freq_out); + if (ret) + return ret; + break; + case MMP_SSPA_CLK: + ret = clk_set_rate(sspa->clk, freq_out); + if (ret) + return ret; + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Set up the sspa dai format. + */ +static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(cpu_dai); + + /* reset port settings */ + sspa->sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; + sspa->ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: + sspa->sp |= SSPA_SP_MSL; + break; + case SND_SOC_DAIFMT_BC_FC: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspa->sp |= SSPA_SP_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspa->ctrl |= SSPA_CTL_XDATDLY(1); + break; + default: + return -EINVAL; + } + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + return 0; +} + +/* + * Set the SSPA audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(dai); + struct device *dev = dai->component->dev; + u32 sspa_ctrl = sspa->ctrl; + int bits; + int bitval; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + bitval = SSPA_CTL_8_BITS; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + bitval = SSPA_CTL_16_BITS; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + bits = 24; + bitval = SSPA_CTL_24_BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + bitval = SSPA_CTL_32_BITS; + break; + default: + return -EINVAL; + } + + sspa_ctrl &= ~SSPA_CTL_XPH; + if (dev->of_node || params_channels(params) == 2) + sspa_ctrl |= SSPA_CTL_XPH; + + sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN1(bitval); + + sspa_ctrl &= ~SSPA_CTL_XWDLEN2_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN2(bitval); + + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; + sspa_ctrl |= SSPA_CTL_XSSZ1(bitval); + + sspa_ctrl &= ~SSPA_CTL_XSSZ2_MASK; + sspa_ctrl |= SSPA_CTL_XSSZ2(bitval); + + sspa->sp &= ~SSPA_SP_FWID_MASK; + sspa->sp |= SSPA_SP_FWID(bits - 1); + + sspa->sp &= ~SSPA_TXSP_FPER_MASK; + sspa->sp |= SSPA_TXSP_FPER(bits * 2 - 1); + + if (dev->of_node) { + clk_set_rate(sspa->clk, params_rate(params) * + params_channels(params) * bits); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + __raw_writel(sspa_ctrl, sspa->tx_base + SSPA_CTL); + __raw_writel(0x1, sspa->tx_base + SSPA_FIFO_UL); + } else { + __raw_writel(sspa_ctrl, sspa->rx_base + SSPA_CTL); + __raw_writel(0x0, sspa->rx_base + SSPA_FIFO_UL); + } + + return 0; +} + +static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * whatever playback or capture, must enable rx. + * this is a hw issue, so need check if rx has been + * enabled or not; if has been enabled by another + * stream, do not enable again. + */ + if (!sspa->running_cnt) + mmp_sspa_rx_enable(sspa); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_enable(sspa); + + sspa->running_cnt++; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sspa->running_cnt--; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_disable(sspa); + + /* have no capture stream, disable rx port */ + if (!sspa->running_cnt) + mmp_sspa_rx_disable(sspa); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int mmp_sspa_probe(struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa = dev_get_drvdata(dai->dev); + + snd_soc_dai_init_dma_data(dai, + &sspa->playback_dma_data, + &sspa->capture_dma_data); + + return 0; +} + +#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000 +#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { + .probe = mmp_sspa_probe, + .startup = mmp_sspa_startup, + .shutdown = mmp_sspa_shutdown, + .trigger = mmp_sspa_trigger, + .hw_params = mmp_sspa_hw_params, + .set_sysclk = mmp_sspa_set_dai_sysclk, + .set_pll = mmp_sspa_set_dai_pll, + .set_fmt = mmp_sspa_set_dai_fmt, +}; + +static struct snd_soc_dai_driver mmp_sspa_dai = { + .playback = { + .channels_min = 1, + .channels_max = 128, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .ops = &mmp_sspa_dai_ops, +}; + +#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 const 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 const struct snd_dmaengine_pcm_config mmp_pcm_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .pcm_hardware = mmp_pcm_hardware, + .prealloc_buffer_size = 4096, +}; + +static int mmp_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + vm_flags_set(vma, VM_DONTEXPAND | VM_DONTDUMP); + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int mmp_sspa_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct sspa_priv *sspa = snd_soc_component_get_drvdata(component); + + pm_runtime_get_sync(component->dev); + + /* we can only change the settings if the port is not in use */ + if ((__raw_readl(sspa->tx_base + SSPA_SP) & SSPA_SP_S_EN) || + (__raw_readl(sspa->rx_base + SSPA_SP) & SSPA_SP_S_EN)) { + dev_err(component->dev, + "can't change hardware dai format: stream is in use\n"); + return -EBUSY; + } + + __raw_writel(sspa->sp, sspa->tx_base + SSPA_SP); + __raw_writel(sspa->sp, sspa->rx_base + SSPA_SP); + + sspa->sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH); + __raw_writel(sspa->sp, sspa->tx_base + SSPA_SP); + __raw_writel(sspa->sp, sspa->rx_base + SSPA_SP); + + /* + * FIXME: hw issue, for the tx serial port, + * can not config the master/slave mode; + * so must clean this bit. + * The master/slave mode has been set in the + * rx port. + */ + __raw_writel(sspa->sp & ~SSPA_SP_MSL, sspa->tx_base + SSPA_SP); + + __raw_writel(sspa->ctrl, sspa->tx_base + SSPA_CTL); + __raw_writel(sspa->ctrl, sspa->rx_base + SSPA_CTL); + + return 0; +} + +static int mmp_sspa_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + pm_runtime_put_sync(component->dev); + return 0; +} + +static const struct snd_soc_component_driver mmp_sspa_component = { + .name = "mmp-sspa", + .mmap = mmp_pcm_mmap, + .open = mmp_sspa_open, + .close = mmp_sspa_close, + .legacy_dai_naming = 1, +}; + +static int asoc_mmp_sspa_probe(struct platform_device *pdev) +{ + struct sspa_priv *sspa; + int ret; + + sspa = devm_kzalloc(&pdev->dev, + sizeof(struct sspa_priv), GFP_KERNEL); + if (!sspa) + return -ENOMEM; + + if (pdev->dev.of_node) { + sspa->rx_base = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(sspa->rx_base)) + return PTR_ERR(sspa->rx_base); + + sspa->tx_base = devm_platform_ioremap_resource(pdev, 1); + if (IS_ERR(sspa->tx_base)) + return PTR_ERR(sspa->tx_base); + + sspa->clk = devm_clk_get(&pdev->dev, "bitclk"); + if (IS_ERR(sspa->clk)) + return PTR_ERR(sspa->clk); + + sspa->audio_clk = devm_clk_get(&pdev->dev, "audio"); + if (IS_ERR(sspa->audio_clk)) + return PTR_ERR(sspa->audio_clk); + } else { + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_IO, 0); + if (res == NULL) + return -ENODEV; + + sspa->rx_base = devm_ioremap(&pdev->dev, res->start, 0x30); + if (!sspa->rx_base) + return -ENOMEM; + + sspa->tx_base = devm_ioremap(&pdev->dev, + res->start + 0x80, 0x30); + if (!sspa->tx_base) + return -ENOMEM; + + sspa->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(sspa->clk)) + return PTR_ERR(sspa->clk); + + sspa->audio_clk = clk_get(NULL, "mmp-audio"); + if (IS_ERR(sspa->audio_clk)) + return PTR_ERR(sspa->audio_clk); + + sspa->sysclk = clk_get(NULL, "mmp-sysclk"); + if (IS_ERR(sspa->sysclk)) { + clk_put(sspa->audio_clk); + return PTR_ERR(sspa->sysclk); + } + } + platform_set_drvdata(pdev, sspa); + + sspa->playback_dma_data.maxburst = 4; + sspa->capture_dma_data.maxburst = 4; + /* You know, these addresses are actually ignored. */ + sspa->capture_dma_data.addr = SSPA_D; + sspa->playback_dma_data.addr = 0x80 + SSPA_D; + + if (pdev->dev.of_node) { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, + &mmp_pcm_config, 0); + if (ret) + return ret; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); + if (ret) + return ret; + + pm_runtime_enable(&pdev->dev); + clk_prepare_enable(sspa->audio_clk); + + return 0; +} + +static void asoc_mmp_sspa_remove(struct platform_device *pdev) +{ + struct sspa_priv *sspa = platform_get_drvdata(pdev); + + clk_disable_unprepare(sspa->audio_clk); + pm_runtime_disable(&pdev->dev); + + if (pdev->dev.of_node) + return; + + clk_put(sspa->audio_clk); + clk_put(sspa->sysclk); +} + +#ifdef CONFIG_OF +static const struct of_device_id mmp_sspa_of_match[] = { + { .compatible = "marvell,mmp-sspa" }, + {}, +}; + +MODULE_DEVICE_TABLE(of, mmp_sspa_of_match); +#endif + +static struct platform_driver asoc_mmp_sspa_driver = { + .driver = { + .name = "mmp-sspa-dai", + .of_match_table = of_match_ptr(mmp_sspa_of_match), + }, + .probe = asoc_mmp_sspa_probe, + .remove_new = asoc_mmp_sspa_remove, +}; + +module_platform_driver(asoc_mmp_sspa_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("MMP SSPA SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-sspa-dai"); diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h new file mode 100644 index 0000000000..938ef2f667 --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.h @@ -0,0 +1,70 @@ +/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* + * linux/sound/soc/pxa/mmp-sspa.h + * + * Copyright (C) 2011 Marvell International Ltd. + */ +#ifndef _MMP_SSPA_H +#define _MMP_SSPA_H + +/* + * SSPA Registers + */ +#define SSPA_D (0x00) +#define SSPA_ID (0x04) +#define SSPA_CTL (0x08) +#define SSPA_SP (0x0c) +#define SSPA_FIFO_UL (0x10) +#define SSPA_INT_MASK (0x14) +#define SSPA_C (0x18) +#define SSPA_FIFO_NOFS (0x1c) +#define SSPA_FIFO_SIZE (0x20) + +/* SSPA Control Register */ +#define SSPA_CTL_XPH (1 << 31) /* Read Phase */ +#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */ +#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */ +#define SSPA_CTL_XFRLEN2_MASK (7 << 24) +#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */ +#define SSPA_CTL_XWDLEN2_MASK (7 << 21) +#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */ +#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Transmit Data Delay */ +#define SSPA_CTL_XSSZ2_MASK (7 << 16) +#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */ +#define SSPA_CTL_XFRLEN1_MASK (7 << 8) +#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */ +#define SSPA_CTL_XWDLEN1_MASK (7 << 5) +#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */ +#define SSPA_CTL_XSSZ1_MASK (7 << 0) +#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */ + +#define SSPA_CTL_8_BITS (0x0) /* Sample Size */ +#define SSPA_CTL_12_BITS (0x1) +#define SSPA_CTL_16_BITS (0x2) +#define SSPA_CTL_20_BITS (0x3) +#define SSPA_CTL_24_BITS (0x4) +#define SSPA_CTL_32_BITS (0x5) + +/* SSPA Serial Port Register */ +#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */ +#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */ +#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */ +#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */ +#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */ +#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */ +#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */ +#define SSPA_SP_FWID_MASK (0x3f << 20) +#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */ +#define SSPA_TXSP_FPER_MASK (0x3f << 4) +#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */ + +/* sspa clock sources */ +#define MMP_SSPA_CLK_PLL 0 +#define MMP_SSPA_CLK_VCXO 1 +#define MMP_SSPA_CLK_AUDIO 3 + +/* sspa pll id */ +#define MMP_SYSCLK 0 +#define MMP_SSPA_CLK 1 + +#endif /* _MMP_SSPA_H */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c new file mode 100644 index 0000000000..b8a3cb8b75 --- /dev/null +++ b/sound/soc/pxa/pxa-ssp.c @@ -0,0 +1,887 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * pxa-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/pxa2xx_ssp.h> +#include <linux/of.h> +#include <linux/dmaengine.h> + +#include <asm/irq.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> + +#include "pxa-ssp.h" + +/* + * SSP audio private data + */ +struct ssp_priv { + struct ssp_device *ssp; + struct clk *extclk; + unsigned long ssp_clk; + unsigned int sysclk; + unsigned int dai_fmt; + unsigned int configured_dai_fmt; +#ifdef CONFIG_PM + uint32_t cr0; + uint32_t cr1; + uint32_t to; + uint32_t psp; +#endif +}; + +static void dump_registers(struct ssp_device *ssp) +{ + dev_dbg(ssp->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", + pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1), + pxa_ssp_read_reg(ssp, SSTO)); + + dev_dbg(ssp->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", + pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR), + pxa_ssp_read_reg(ssp, SSACD)); +} + +static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, + int out, struct snd_dmaengine_dai_dma_data *dma) +{ + dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES : + DMA_SLAVE_BUSWIDTH_2_BYTES; + dma->maxburst = 16; + dma->addr = ssp->phys_base + SSDR; +} + +static int pxa_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + struct snd_dmaengine_dai_dma_data *dma; + int ret = 0; + + if (!snd_soc_dai_active(cpu_dai)) { + clk_prepare_enable(ssp->clk); + pxa_ssp_disable(ssp); + } + + clk_prepare_enable(priv->extclk); + + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); + if (!dma) + return -ENOMEM; + dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "tx" : "rx"; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma); + + return ret; +} + +static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + + if (!snd_soc_dai_active(cpu_dai)) { + pxa_ssp_disable(ssp); + clk_disable_unprepare(ssp->clk); + } + + clk_disable_unprepare(priv->extclk); + + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); +} + +#ifdef CONFIG_PM + +static int pxa_ssp_suspend(struct snd_soc_component *component) +{ + struct ssp_priv *priv = snd_soc_component_get_drvdata(component); + struct ssp_device *ssp = priv->ssp; + + if (!snd_soc_component_active(component)) + clk_prepare_enable(ssp->clk); + + priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); + priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); + priv->to = __raw_readl(ssp->mmio_base + SSTO); + priv->psp = __raw_readl(ssp->mmio_base + SSPSP); + + pxa_ssp_disable(ssp); + clk_disable_unprepare(ssp->clk); + return 0; +} + +static int pxa_ssp_resume(struct snd_soc_component *component) +{ + struct ssp_priv *priv = snd_soc_component_get_drvdata(component); + struct ssp_device *ssp = priv->ssp; + uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; + + clk_prepare_enable(ssp->clk); + + __raw_writel(sssr, ssp->mmio_base + SSSR); + __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); + __raw_writel(priv->cr1, ssp->mmio_base + SSCR1); + __raw_writel(priv->to, ssp->mmio_base + SSTO); + __raw_writel(priv->psp, ssp->mmio_base + SSPSP); + + if (snd_soc_component_active(component)) + pxa_ssp_enable(ssp); + else + clk_disable_unprepare(ssp->clk); + + return 0; +} + +#else +#define pxa_ssp_suspend NULL +#define pxa_ssp_resume NULL +#endif + +/* + * ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + + if (ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + + dev_dbg(ssp->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (ssp->type == PXA25x_SSP) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA_SSP_CLK_AUDIO: + priv->sysclk = 0; + pxa_ssp_set_scr(ssp, 1); + sscr0 |= SSCR0_ACS; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (ssp->type != PXA3xx_SSP) + clk_disable_unprepare(ssp->clk); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + if (ssp->type != PXA3xx_SSP) + clk_prepare_enable(ssp->clk); + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq) +{ + struct ssp_device *ssp = priv->ssp; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; + + if (ssp->type == PXA3xx_SSP) + pxa_ssp_write_reg(ssp, SSACDD, 0); + + switch (freq) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (ssp->type == PXA3xx_SSP) { + u32 val; + u64 tmp = 19968; + + tmp *= 1000000; + do_div(tmp, freq); + val = tmp; + + val = (val << 16) | 64; + pxa_ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(ssp->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq); + break; + } + + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +/* + * Set the active slots in TDM/Network mode + */ +static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + u32 sscr0; + + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); + + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + pxa_ssp_write_reg(ssp, SSTSA, tx_mask); + pxa_ssp_write_reg(ssp, SSRSA, rx_mask); + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + + return 0; +} + +/* + * Tristate the SSP DAI lines + */ +static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, + int tristate) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + u32 sscr1; + + sscr1 = pxa_ssp_read_reg(ssp, SSCR1); + if (tristate) + sscr1 &= ~SSCR1_TTE; + else + sscr1 |= SSCR1_TTE; + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + + return 0; +} + +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BP_FP: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + break; + + default: + return -EINVAL; + } + + /* Settings will be applied in hw_params() */ + priv->dai_fmt = fmt; + + return 0; +} + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) +{ + struct ssp_device *ssp = priv->ssp; + u32 sscr0, sscr1, sspsp, scfr; + + /* check if we need to change anything at all */ + if (priv->configured_dai_fmt == priv->dai_fmt) + return 0; + + /* reset port settings */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_PSP | SSCR0_MOD); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1) & + ~(SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR | + SSCR1_RWOT | SSCR1_TRAIL | SSCR1_TFT | SSCR1_RFT); + sspsp = pxa_ssp_read_reg(ssp, SSPSP) & + ~(SSPSP_SFRMP | SSPSP_SCMODE(3)); + + sscr1 |= SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR; + break; + case SND_SOC_DAIFMT_BC_FP: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR; + break; + case SND_SOC_DAIFMT_BP_FP: + break; + default: + return -EINVAL; + } + + switch (priv->dai_fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + fallthrough; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: + scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR; + pxa_ssp_write_reg(ssp, SSCR1, scfr); + + while (pxa_ssp_read_reg(ssp, SSSR) & SSSR_BSY) + cpu_relax(); + break; + } + + dump_registers(ssp); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + priv->configured_dai_fmt = priv->dai_fmt; + + return 0; +} + +struct pxa_ssp_clock_mode { + int rate; + int pll; + u8 acds; + u8 scdb; +}; + +static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = { + { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X }, + { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X }, + { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X }, + { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X }, + {} +}; + +/* + * Set the SSP audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + int chn = params_channels(params); + u32 sscr0, sspsp; + int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; + struct snd_dmaengine_dai_dma_data *dma_data; + int rate = params_rate(params); + int bclk = rate * chn * (width / 8); + int ret; + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + + /* Network mode with one active slot (ttsa == 1) can be used + * to force 16-bit frame width on the wire (for S16_LE), even + * with two channels. Use 16-bit DMA transfers for this case. + */ + pxa_ssp_set_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data); + + /* we can only change the settings if the port is not in use */ + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + return 0; + + ret = pxa_ssp_configure_dai_fmt(priv); + if (ret < 0) + return ret; + + /* clear selected SSP bits */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (ssp->type == PXA3xx_SSP) + sscr0 |= SSCR0_FPCKE; + sscr0 |= SSCR0_DataSize(16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); + break; + case SNDRV_PCM_FORMAT_S32_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); + break; + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + + if (sscr0 & SSCR0_ACS) { + ret = pxa_ssp_set_pll(priv, bclk); + + /* + * If we were able to generate the bclk directly, + * all is fine. Otherwise, look up the closest rate + * from the table and also set the dividers. + */ + + if (ret < 0) { + const struct pxa_ssp_clock_mode *m; + int ssacd; + + for (m = pxa_ssp_clock_modes; m->rate; m++) { + if (m->rate == rate) + break; + } + + if (!m->rate) + return -EINVAL; + + ret = pxa_ssp_set_pll(priv, bclk); + if (ret < 0) + return ret; + + ssacd = pxa_ssp_read_reg(ssp, SSACD); + ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); + ssacd |= SSACD_ACDS(m->acds); + ssacd |= m->scdb; + pxa_ssp_write_reg(ssp, SSACD, ssacd); + } + } else if (sscr0 & SSCR0_ECS) { + /* + * For setups with external clocking, the PLL and its diviers + * are not active. Instead, the SCR bits in SSCR0 can be used + * to divide the clock. + */ + pxa_ssp_set_scr(ssp, bclk / rate); + } + + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspsp = pxa_ssp_read_reg(ssp, SSPSP); + + if (((priv->sysclk / bclk) == 64) && (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + if (ssp->type != PXA3xx_SSP) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); + } else { + /* The frame width is the width the LRCLK is + * asserted for; the delay is expressed in + * half cycle units. We need the extra cycle + * because the data starts clocking out one BCLK + * after LRCLK changes polarity. + */ + sspsp |= SSPSP_SFRMWDTH(width + 1); + sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_DMYSTRT(1); + } + + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + break; + default: + break; + } + + /* When we use a network mode, we always require TDM slots + * - complain loudly and fail if they've not been set up yet. + */ + if ((sscr0 & SSCR0_MOD) && !ttsa) { + dev_err(ssp->dev, "No TDM timeslot configured\n"); + return -EINVAL; + } + + dump_registers(ssp); + + return 0; +} + +static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream, + struct ssp_device *ssp, int value) +{ + uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1); + uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP); + uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR); + + if (value && (sscr0 & SSCR0_SSE)) + pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (value) + sscr1 |= SSCR1_TSRE; + else + sscr1 &= ~SSCR1_TSRE; + } else { + if (value) + sscr1 |= SSCR1_RSRE; + else + sscr1 &= ~SSCR1_RSRE; + } + + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + + if (value) { + pxa_ssp_write_reg(ssp, SSSR, sssr); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE); + } +} + +static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *ssp = priv->ssp; + int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + pxa_ssp_enable(ssp); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pxa_ssp_set_running_bit(substream, ssp, 1); + val = pxa_ssp_read_reg(ssp, SSSR); + pxa_ssp_write_reg(ssp, SSSR, val); + break; + case SNDRV_PCM_TRIGGER_START: + pxa_ssp_set_running_bit(substream, ssp, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + pxa_ssp_set_running_bit(substream, ssp, 0); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + pxa_ssp_disable(ssp); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pxa_ssp_set_running_bit(substream, ssp, 0); + break; + + default: + ret = -EINVAL; + } + + dump_registers(ssp); + + return ret; +} + +static int pxa_ssp_probe(struct snd_soc_dai *dai) +{ + struct device *dev = dai->dev; + struct ssp_priv *priv; + int ret; + + priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + if (dev->of_node) { + struct device_node *ssp_handle; + + ssp_handle = of_parse_phandle(dev->of_node, "port", 0); + if (!ssp_handle) { + dev_err(dev, "unable to get 'port' phandle\n"); + ret = -ENODEV; + goto err_priv; + } + + priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + goto err_priv; + + priv->extclk = NULL; + } + } else { + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + } + + priv->dai_fmt = (unsigned int) -1; + snd_soc_dai_set_drvdata(dai, priv); + + return 0; + +err_priv: + kfree(priv); + return ret; +} + +static int pxa_ssp_remove(struct snd_soc_dai *dai) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(dai); + + pxa_ssp_free(priv->ssp); + kfree(priv); + return 0; +} + +#define PXA_SSP_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_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, +}; + +static struct snd_soc_dai_driver pxa_ssp_dai = { + .playback = { + .channels_min = 1, + .channels_max = 8, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = &pxa_ssp_dai_ops, +}; + +static const struct snd_soc_component_driver pxa_ssp_component = { + .name = "pxa-ssp", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .legacy_dai_naming = 1, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pxa_ssp_of_ids[] = { + { .compatible = "mrvl,pxa-ssp-dai" }, + {} +}; +MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids); +#endif + +static int asoc_ssp_probe(struct platform_device *pdev) +{ + return devm_snd_soc_register_component(&pdev->dev, &pxa_ssp_component, + &pxa_ssp_dai, 1); +} + +static struct platform_driver asoc_ssp_driver = { + .driver = { + .name = "pxa-ssp-dai", + .of_match_table = of_match_ptr(pxa_ssp_of_ids), + }, + + .probe = asoc_ssp_probe, +}; + +module_platform_driver(asoc_ssp_driver); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-ssp-dai"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h new file mode 100644 index 0000000000..d3b05109df --- /dev/null +++ b/sound/soc/pxa/pxa-ssp.h @@ -0,0 +1,36 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * ASoC PXA SSP port support + */ + +#ifndef _PXA_SSP_H +#define _PXA_SSP_H + +/* SSP clock sources */ +#define PXA_SSP_CLK_PLL 0 +#define PXA_SSP_CLK_EXT 1 +#define PXA_SSP_CLK_NET 2 +#define PXA_SSP_CLK_AUDIO 3 +#define PXA_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA_SSP_AUDIO_DIV_ACDS 0 +#define PXA_SSP_AUDIO_DIV_SCDB 1 +#define PXA_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA_SSP_CLK_AUDIO_DIV_1 0 +#define PXA_SSP_CLK_AUDIO_DIV_2 1 +#define PXA_SSP_CLK_AUDIO_DIV_4 2 +#define PXA_SSP_CLK_AUDIO_DIV_8 3 +#define PXA_SSP_CLK_AUDIO_DIV_16 4 +#define PXA_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA_SSP_CLK_SCDB_4 0 +#define PXA_SSP_CLK_SCDB_1 1 +#define PXA_SSP_CLK_SCDB_8 2 + +#define PXA_SSP_PLL_OUT 0 + +#endif diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c new file mode 100644 index 0000000000..e73bd62c03 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -0,0 +1,306 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip. + * + * Author: Nicolas Pitre + * Created: Dec 02, 2004 + * Copyright: MontaVista Software Inc. + */ + +#include <linux/init.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> + +#include <sound/ac97/controller.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> + +#include <linux/platform_data/asoc-pxa.h> + +#define PCDR 0x0040 /* PCM FIFO Data Register */ +#define MODR 0x0140 /* Modem FIFO Data Register */ +#define MCDR 0x0060 /* Mic-in FIFO Data Register */ + +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) +{ + pxa2xx_ac97_try_warm_reset(); + + pxa2xx_ac97_finish_reset(); +} + +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) +{ + pxa2xx_ac97_try_cold_reset(); + + pxa2xx_ac97_finish_reset(); +} + +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) +{ + return pxa2xx_ac97_read(slot, reg); +} + +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) +{ + return pxa2xx_ac97_write(slot, reg, val); +} + +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, + .warm_reset = pxa2xx_ac97_warm_reset, + .reset = pxa2xx_ac97_cold_reset, +}; + +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", + .maxburst = 32, +}; + +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", + .maxburst = 32, +}; + +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", + .maxburst = 16, +}; + +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", + .maxburst = 16, +}; + +static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { + .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", + .maxburst = 16, +}; + +static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_dmaengine_dai_dma_data *dma_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &pxa2xx_ac97_pcm_stereo_out; + else + dma_data = &pxa2xx_ac97_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + +static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_dmaengine_dai_dma_data *dma_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &pxa2xx_ac97_pcm_aux_mono_out; + else + dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + +static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); + + return 0; +} + +#define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { + .startup = pxa2xx_ac97_hifi_startup, +}; + +static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { + .startup = pxa2xx_ac97_aux_startup, +}; + +static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { + .startup = pxa2xx_ac97_mic_startup, +}; + +/* + * There is only 1 physical AC97 interface for pxa2xx, but it + * has extra fifo's that can be used for aux DACs and ADCs. + */ +static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { +{ + .name = "pxa2xx-ac97", + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PXA2XX_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = PXA2XX_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &pxa_ac97_hifi_dai_ops, +}, +{ + .name = "pxa2xx-ac97-aux", + .playback = { + .stream_name = "AC97 Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = PXA2XX_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Aux Capture", + .channels_min = 1, + .channels_max = 1, + .rates = PXA2XX_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &pxa_ac97_aux_dai_ops, +}, +{ + .name = "pxa2xx-ac97-mic", + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = PXA2XX_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &pxa_ac97_mic_dai_ops, +}, +}; + +static const struct snd_soc_component_driver pxa_ac97_component = { + .name = "pxa-ac97", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pxa2xx_ac97_dt_ids[] = { + { .compatible = "marvell,pxa250-ac97", }, + { .compatible = "marvell,pxa270-ac97", }, + { .compatible = "marvell,pxa300-ac97", }, + { } +}; +MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); + +#endif + +static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) +{ + int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + struct resource *regs; + void **codecs_pdata; + + if (pdev->id != -1) { + dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); + return -ENXIO; + } + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) + return -ENXIO; + + pxa2xx_ac97_pcm_stereo_in.addr = regs->start + PCDR; + pxa2xx_ac97_pcm_stereo_out.addr = regs->start + PCDR; + pxa2xx_ac97_pcm_aux_mono_out.addr = regs->start + MODR; + pxa2xx_ac97_pcm_aux_mono_in.addr = regs->start + MODR; + pxa2xx_ac97_pcm_mic_mono_in.addr = regs->start + MCDR; + + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret) { + dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret); + return ret; + } + + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + + platform_set_drvdata(pdev, ctrl); + /* Punt most of the init to the SoC probe; we may need the machine + * driver to do interesting things with the clocking to get us up + * and running. + */ + return devm_snd_soc_register_component(&pdev->dev, &pxa_ac97_component, + pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver)); +} + +static void pxa2xx_ac97_dev_remove(struct platform_device *pdev) +{ + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + + snd_ac97_controller_unregister(ctrl); + pxa2xx_ac97_hw_remove(pdev); +} + +#ifdef CONFIG_PM_SLEEP +static int pxa2xx_ac97_dev_suspend(struct device *dev) +{ + return pxa2xx_ac97_hw_suspend(); +} + +static int pxa2xx_ac97_dev_resume(struct device *dev) +{ + return pxa2xx_ac97_hw_resume(); +} + +static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, + pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume); +#endif + +static struct platform_driver pxa2xx_ac97_driver = { + .probe = pxa2xx_ac97_dev_probe, + .remove_new = pxa2xx_ac97_dev_remove, + .driver = { + .name = "pxa2xx-ac97", +#ifdef CONFIG_PM_SLEEP + .pm = &pxa2xx_ac97_pm_ops, +#endif + .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), + }, +}; + +module_platform_driver(pxa2xx_ac97_driver); + +MODULE_AUTHOR("Nicolas Pitre"); +MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c new file mode 100644 index 0000000000..437bfccd04 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -0,0 +1,412 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * pxa2xx-i2s.c -- ALSA Soc Audio Layer + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * lrg@slimlogic.co.uk + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> + +#include <linux/platform_data/asoc-pxa.h> + +#include "pxa2xx-i2s.h" + +/* + * I2S Controller Register and Bit Definitions + */ +#define SACR0 (0x0000) /* Global Control Register */ +#define SACR1 (0x0004) /* Serial Audio I 2 S/MSB-Justified Control Register */ +#define SASR0 (0x000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */ +#define SAIMR (0x0014) /* Serial Audio Interrupt Mask Register */ +#define SAICR (0x0018) /* Serial Audio Interrupt Clear Register */ +#define SADIV (0x0060) /* Audio Clock Divider Register. */ +#define SADR (0x0080) /* Serial Audio Data Register (TX and RX FIFO access Register). */ + +#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */ +#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */ +#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */ +#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ +#define SACR0_ENB (1 << 0) /* Enable I2S Link */ +#define SACR1_ENLBF (1 << 5) /* Enable Loopback */ +#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ +#define SACR1_DREC (1 << 3) /* Disable Recording Function */ +#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */ + +#define SASR0_I2SOFF (1 << 7) /* Controller Status */ +#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */ +#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */ +#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */ +#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */ +#define SASR0_BSY (1 << 2) /* I2S Busy */ +#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */ +#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ + +#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */ +#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */ + +#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */ +#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */ +#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */ +#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */ + +struct pxa_i2s_port { + u32 sadiv; + u32 sacr0; + u32 sacr1; + u32 saimr; + int master; + u32 fmt; +}; +static struct pxa_i2s_port pxa_i2s; +static struct clk *clk_i2s; +static int clk_ena = 0; +static void __iomem *i2s_reg_base; + +static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", + .maxburst = 32, +}; + +static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", + .maxburst = 32, +}; + +static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + + if (IS_ERR(clk_i2s)) + return PTR_ERR(clk_i2s); + + if (!snd_soc_dai_active(cpu_dai)) + writel(0, i2s_reg_base + SACR0); + + return 0; +} + +/* wait for I2S controller to be ready */ +static int pxa_i2s_wait(void) +{ + int i; + + /* flush the Rx FIFO */ + for (i = 0; i < 16; i++) + readl(i2s_reg_base + SADR); + return 0; +} + +static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + pxa_i2s.fmt = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + pxa_i2s.fmt = SACR1_AMSL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: + pxa_i2s.master = 1; + break; + case SND_SOC_DAIFMT_BC_FP: + pxa_i2s.master = 0; + break; + default: + break; + } + return 0; +} + +static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + if (clk_id != PXA2XX_I2S_SYSCLK) + return -ENODEV; + + return 0; +} + +static int pxa2xx_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; + + if (WARN_ON(IS_ERR(clk_i2s))) + return -EINVAL; + clk_prepare_enable(clk_i2s); + clk_ena = 1; + pxa_i2s_wait(); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &pxa2xx_i2s_pcm_stereo_out; + else + dma_data = &pxa2xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(dai, substream, dma_data); + + /* is port used by another stream */ + if (!(SACR0 & SACR0_ENB)) { + writel(0, i2s_reg_base + SACR0); + if (pxa_i2s.master) + writel(readl(i2s_reg_base + SACR0) | (SACR0_BCKD), i2s_reg_base + SACR0); + + writel(readl(i2s_reg_base + SACR0) | (SACR0_RFTH(14) | SACR0_TFTH(1)), i2s_reg_base + SACR0); + writel(readl(i2s_reg_base + SACR1) | (pxa_i2s.fmt), i2s_reg_base + SACR1); + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + writel(readl(i2s_reg_base + SAIMR) | (SAIMR_TFS), i2s_reg_base + SAIMR); + else + writel(readl(i2s_reg_base + SAIMR) | (SAIMR_RFS), i2s_reg_base + SAIMR); + + switch (params_rate(params)) { + case 8000: + writel(0x48, i2s_reg_base + SADIV); + break; + case 11025: + writel(0x34, i2s_reg_base + SADIV); + break; + case 16000: + writel(0x24, i2s_reg_base + SADIV); + break; + case 22050: + writel(0x1a, i2s_reg_base + SADIV); + break; + case 44100: + writel(0xd, i2s_reg_base + SADIV); + break; + case 48000: + writel(0xc, i2s_reg_base + SADIV); + break; + case 96000: /* not in manual and possibly slightly inaccurate */ + writel(0x6, i2s_reg_base + SADIV); + break; + } + + return 0; +} + +static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + writel(readl(i2s_reg_base + SACR1) & (~SACR1_DRPL), i2s_reg_base + SACR1); + else + writel(readl(i2s_reg_base + SACR1) & (~SACR1_DREC), i2s_reg_base + SACR1); + writel(readl(i2s_reg_base + SACR0) | (SACR0_ENB), i2s_reg_base + SACR0); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + writel(readl(i2s_reg_base + SACR1) | (SACR1_DRPL), i2s_reg_base + SACR1); + writel(readl(i2s_reg_base + SAIMR) & (~SAIMR_TFS), i2s_reg_base + SAIMR); + } else { + writel(readl(i2s_reg_base + SACR1) | (SACR1_DREC), i2s_reg_base + SACR1); + writel(readl(i2s_reg_base + SAIMR) & (~SAIMR_RFS), i2s_reg_base + SAIMR); + } + + if ((readl(i2s_reg_base + SACR1) & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { + writel(readl(i2s_reg_base + SACR0) & (~SACR0_ENB), i2s_reg_base + SACR0); + pxa_i2s_wait(); + if (clk_ena) { + clk_disable_unprepare(clk_i2s); + clk_ena = 0; + } + } +} + +#ifdef CONFIG_PM +static int pxa2xx_soc_pcm_suspend(struct snd_soc_component *component) +{ + /* store registers */ + pxa_i2s.sacr0 = readl(i2s_reg_base + SACR0); + pxa_i2s.sacr1 = readl(i2s_reg_base + SACR1); + pxa_i2s.saimr = readl(i2s_reg_base + SAIMR); + pxa_i2s.sadiv = readl(i2s_reg_base + SADIV); + + /* deactivate link */ + writel(readl(i2s_reg_base + SACR0) & (~SACR0_ENB), i2s_reg_base + SACR0); + pxa_i2s_wait(); + return 0; +} + +static int pxa2xx_soc_pcm_resume(struct snd_soc_component *component) +{ + pxa_i2s_wait(); + + writel(pxa_i2s.sacr0 & ~SACR0_ENB, i2s_reg_base + SACR0); + writel(pxa_i2s.sacr1, i2s_reg_base + SACR1); + writel(pxa_i2s.saimr, i2s_reg_base + SAIMR); + writel(pxa_i2s.sadiv, i2s_reg_base + SADIV); + + writel(pxa_i2s.sacr0, i2s_reg_base + SACR0); + + return 0; +} + +#else +#define pxa2xx_soc_pcm_suspend NULL +#define pxa2xx_soc_pcm_resume NULL +#endif + +static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) +{ + clk_i2s = clk_get(dai->dev, "I2SCLK"); + if (IS_ERR(clk_i2s)) + return PTR_ERR(clk_i2s); + + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + writel(SACR0_RST, i2s_reg_base + SACR0); + writel(0, i2s_reg_base + SACR0); + /* Make sure RPL and REC are disabled */ + writel(SACR1_DRPL | SACR1_DREC, i2s_reg_base + SACR1); + /* Along with FIFO servicing */ + writel(readl(i2s_reg_base + SAIMR) & (~(SAIMR_RFS | SAIMR_TFS)), i2s_reg_base + SAIMR); + + snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out, + &pxa2xx_i2s_pcm_stereo_in); + + return 0; +} + +static int pxa2xx_i2s_remove(struct snd_soc_dai *dai) +{ + clk_put(clk_i2s); + clk_i2s = ERR_PTR(-ENOENT); + return 0; +} + +#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .probe = pxa2xx_i2s_probe, + .remove = pxa2xx_i2s_remove, + .startup = pxa2xx_i2s_startup, + .shutdown = pxa2xx_i2s_shutdown, + .trigger = pxa2xx_i2s_trigger, + .hw_params = pxa2xx_i2s_hw_params, + .set_fmt = pxa2xx_i2s_set_dai_fmt, + .set_sysclk = pxa2xx_i2s_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver pxa_i2s_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = PXA2XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = PXA2XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &pxa_i2s_dai_ops, + .symmetric_rate = 1, +}; + +static const struct snd_soc_component_driver pxa_i2s_component = { + .name = "pxa-i2s", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, + .suspend = pxa2xx_soc_pcm_suspend, + .resume = pxa2xx_soc_pcm_resume, + .legacy_dai_naming = 1, +}; + +static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) +{ + struct resource *res; + + i2s_reg_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); + if (IS_ERR(i2s_reg_base)) + return PTR_ERR(i2s_reg_base); + + pxa2xx_i2s_pcm_stereo_out.addr = res->start + SADR; + pxa2xx_i2s_pcm_stereo_in.addr = res->start + SADR; + + return devm_snd_soc_register_component(&pdev->dev, &pxa_i2s_component, + &pxa_i2s_dai, 1); +} + +static struct platform_driver pxa2xx_i2s_driver = { + .probe = pxa2xx_i2s_drv_probe, + + .driver = { + .name = "pxa2xx-i2s", + }, +}; + +static int __init pxa2xx_i2s_init(void) +{ + clk_i2s = ERR_PTR(-ENOENT); + return platform_driver_register(&pxa2xx_i2s_driver); +} + +static void __exit pxa2xx_i2s_exit(void) +{ + platform_driver_unregister(&pxa2xx_i2s_driver); +} + +module_init(pxa2xx_i2s_init); +module_exit(pxa2xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); +MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-i2s"); diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h new file mode 100644 index 0000000000..263568d445 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-i2s.h @@ -0,0 +1,12 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/soc/pxa/pxa2xx-i2s.h + */ + +#ifndef _PXA2XX_I2S_H +#define _PXA2XX_I2S_H + +/* I2S clock */ +#define PXA2XX_I2S_SYSCLK 0 + +#endif diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c new file mode 100644 index 0000000000..9d6c41f775 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -0,0 +1,49 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + */ + +#include <linux/dma-mapping.h> +#include <linux/module.h> +#include <linux/dmaengine.h> +#include <linux/of.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include <sound/dmaengine_pcm.h> + +static const struct snd_soc_component_driver pxa2xx_soc_platform = { + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, +}; + +static int pxa2xx_soc_platform_probe(struct platform_device *pdev) +{ + return devm_snd_soc_register_component(&pdev->dev, &pxa2xx_soc_platform, + NULL, 0); +} + +static struct platform_driver pxa_pcm_driver = { + .driver = { + .name = "pxa-pcm-audio", + }, + + .probe = pxa2xx_soc_platform_probe, +}; + +module_platform_driver(pxa_pcm_driver); + +MODULE_AUTHOR("Nicolas Pitre"); +MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-pcm-audio"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c new file mode 100644 index 0000000000..70442315f5 --- /dev/null +++ b/sound/soc/pxa/spitz.c @@ -0,0 +1,322 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita + * + * 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/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 <asm/mach-types.h> +#include "../codecs/wm8750.h" +#include "pxa2xx-i2s.h" + +#define SPITZ_HP 0 +#define SPITZ_MIC 1 +#define SPITZ_LINE 2 +#define SPITZ_HEADSET 3 +#define SPITZ_HP_OFF 4 +#define SPITZ_SPK_ON 0 +#define SPITZ_SPK_OFF 1 + + /* audio clock in Hz - rounded from 12.235MHz */ +#define SPITZ_AUDIO_CLOCK 12288000 + +static int spitz_jack_func; +static int spitz_spk_func; +static struct gpio_desc *gpiod_mic, *gpiod_mute_l, *gpiod_mute_r; + +static void spitz_ext_control(struct snd_soc_dapm_context *dapm) +{ + snd_soc_dapm_mutex_lock(dapm); + + if (spitz_spk_func == SPITZ_SPK_ON) + snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk"); + else + snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk"); + + /* set up jack connection */ + switch (spitz_jack_func) { + case SPITZ_HP: + /* enable and unmute hp jack, disable mic bias */ + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); + gpiod_set_value(gpiod_mute_l, 1); + gpiod_set_value(gpiod_mute_r, 1); + break; + case SPITZ_MIC: + /* enable mic jack and bias, mute hp */ + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); + break; + case SPITZ_LINE: + /* enable line jack, disable mic bias and mute hp */ + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); + break; + case SPITZ_HEADSET: + /* enable and unmute headset jack enable mic bias, mute L hp */ + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 1); + break; + case SPITZ_HP_OFF: + + /* jack removed, everything off */ + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); + break; + } + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static int spitz_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + /* check the jack status at stream startup */ + spitz_ext_control(&rtd->card->dapm); + + return 0; +} + +static int spitz_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + 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 const struct snd_soc_ops spitz_ops = { + .startup = spitz_startup, + .hw_params = spitz_hw_params, +}; + +static int spitz_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = spitz_jack_func; + return 0; +} + +static int spitz_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + + if (spitz_jack_func == ucontrol->value.enumerated.item[0]) + return 0; + + spitz_jack_func = ucontrol->value.enumerated.item[0]; + spitz_ext_control(&card->dapm); + return 1; +} + +static int spitz_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = spitz_spk_func; + return 0; +} + +static int spitz_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + + if (spitz_spk_func == ucontrol->value.enumerated.item[0]) + return 0; + + spitz_spk_func = ucontrol->value.enumerated.item[0]; + spitz_ext_control(&card->dapm); + return 1; +} + +static int spitz_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(gpiod_mic, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* spitz 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", spitz_mic_bias), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), + + /* headset is a mic and mono headphone */ + SND_SOC_DAPM_HP("Headset Jack", NULL), +}; + +/* Spitz machine audio_map */ +static const struct snd_soc_dapm_route spitz_audio_map[] = { + + /* headphone connected to LOUT1, ROUT1 */ + {"Headphone Jack", NULL, "LOUT1"}, + {"Headphone Jack", NULL, "ROUT1"}, + + /* headset connected to ROUT1 and LINPUT1 with bias (def below) */ + {"Headset Jack", NULL, "ROUT1"}, + + /* ext speaker connected to LOUT2, ROUT2 */ + {"Ext Spk", NULL, "ROUT2"}, + {"Ext Spk", NULL, "LOUT2"}, + + /* mic is connected to input 1 - with bias */ + {"LINPUT1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Mic Jack"}, + + /* line is connected to input 1 - no bias */ + {"LINPUT1", 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 spitz_enum[] = { + SOC_ENUM_SINGLE_EXT(5, jack_function), + SOC_ENUM_SINGLE_EXT(2, spk_function), +}; + +static const struct snd_kcontrol_new wm8750_spitz_controls[] = { + SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack, + spitz_set_jack), + SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk, + spitz_set_spk), +}; + +/* spitz 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 spitz_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &spitz_ops, + SND_SOC_DAILINK_REG(wm8750), +}; + +/* spitz audio machine driver */ +static struct snd_soc_card snd_soc_spitz = { + .name = "Spitz", + .owner = THIS_MODULE, + .dai_link = &spitz_dai, + .num_links = 1, + + .controls = wm8750_spitz_controls, + .num_controls = ARRAY_SIZE(wm8750_spitz_controls), + .dapm_widgets = wm8750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), + .dapm_routes = spitz_audio_map, + .num_dapm_routes = ARRAY_SIZE(spitz_audio_map), + .fully_routed = true, +}; + +static int spitz_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_spitz; + int ret; + + gpiod_mic = devm_gpiod_get(&pdev->dev, "mic", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mic)) + return PTR_ERR(gpiod_mic); + gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_l)) + return PTR_ERR(gpiod_mute_l); + gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_r)) + return PTR_ERR(gpiod_mute_r); + + card->dev = &pdev->dev; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + + return ret; +} + +static struct platform_driver spitz_driver = { + .driver = { + .name = "spitz-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = spitz_probe, +}; + +module_platform_driver(spitz_driver); + +MODULE_AUTHOR("Richard Purdie"); +MODULE_DESCRIPTION("ALSA SoC Spitz"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spitz-audio"); |