88PM860x codec is used in Marvell tavorevb3 development board. 88PM860x codec is used as master mode of SSP communication. Only I2S format is supported.

Signed-off-by: Haojian Zhuang haojian.zhuang@marvell.com --- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa2xx-ssp.c | 532 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa2xx-ssp.h | 59 +++++ sound/soc/pxa/ssp.c | 298 +++++++++++++++++++++++++ sound/soc/pxa/ssp.h | 42 ++++ sound/soc/pxa/tavorevb3.c | 193 ++++++++++++++++ 7 files changed, 1135 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/pxa2xx-ssp.c create mode 100644 sound/soc/pxa/pxa2xx-ssp.h create mode 100644 sound/soc/pxa/ssp.c create mode 100644 sound/soc/pxa/ssp.h create mode 100644 sound/soc/pxa/tavorevb3.c

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c832..04ddc7b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer.

+config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8..315941f 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,7 @@ 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-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(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-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c new file mode 100644 index 0000000..5eab055 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -0,0 +1,532 @@ +/* + * pxa2xx-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown broonie@opensource.wolfsonmicro.com + * + * Copyright 2009-2010 Marvell International Ltd. + * Author: Haojian Zhuang haojian.zhuang@marvell.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.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 <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-ssp.h> +#include <mach/audio.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ssp.h" +#include "ssp.h" + +static void dump_registers(struct ssp_device *ssp) +{ + dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", + ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), + ssp_read_reg(ssp, SSTO)); + + dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", + ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), + ssp_read_reg(ssp, SSACD)); +} + +/** + * ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int val; + + u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA2XX_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + info->sysclk = 1843200; + else + info->sysclk = 13000000; + break; + case PXA2XX_SSP_CLK_EXT: + info->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA2XX_SSP_CLK_NET: + info->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_AUDIO: + info->sysclk = 0; + 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 (!cpu_is_pxa3xx()) + clk_disable(info->dev.ssp->clk); + val = ssp_read_reg(ssp, SSCR0) | sscr0; + ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(info->dev.ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int val; + + switch (div_id) { + case PXA2XX_SSP_AUDIO_DIV_ACDS: + val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_AUDIO_DIV_SCDB: + val = ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; + switch (div) { + case PXA2XX_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA2XX_SSP_CLK_SCDB_4: + break; + case PXA2XX_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_DIV_SCR: + ssp_set_scr(ssp, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa2xx_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + + if (cpu_is_pxa3xx()) + ssp_write_reg(ssp, SSACDD, 0); + + switch (freq_out) { + 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 (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64; + ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq_out); + break; + } + + return -EINVAL; + } + + ssp_write_reg(ssp, SSACD, ssacd); + + 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 pxa2xx_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* check if we need to change anything at all */ + if (info->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + dev_err(&ssp->pdev->dev, + "can't change hardware dai format: stream is in use"); + return -EINVAL; + } + + /* reset port settings */ + sscr0 = ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (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 (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; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + + 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. + */ + info->dai_fmt = fmt; + + return 0; +} + +/* + * Set the SSP audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int chn = params_channels(params); + u32 sscr0; + u32 sspsp; + int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); + + /* 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. + */ + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + return 0; + + /* clear selected SSP bits */ + sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + ssp_write_reg(ssp, SSCR0, sscr0); + + /* bit size */ + sscr0 = ssp_read_reg(ssp, SSCR0); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (cpu_is_pxa3xx()) + 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; + } + ssp_write_reg(ssp, SSCR0, sscr0); + + switch (info->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspsp = ssp_read_reg(ssp, SSPSP); + + if ((ssp_get_scr(ssp) == 4) && (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 (!cpu_is_pxa3xx()) + 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); + } + + 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->pdev->dev, "No TDM timeslot configured\n"); + return -EINVAL; + } + + dump_registers(ssp); + + return 0; +} + +static struct snd_soc_dai_ops pxa2xx_ssp_dai_ops = { + .hw_params = pxa2xx_ssp_hw_params, + .set_sysclk = pxa2xx_ssp_set_dai_sysclk, + .set_clkdiv = pxa2xx_ssp_set_dai_clkdiv, + .set_pll = pxa2xx_ssp_set_dai_pll, + .set_fmt = pxa2xx_ssp_set_dai_fmt, +}; + +struct snd_soc_dai pxa2xx_ssp_dai[PXA2XX_DAI_SSP_MAX]; +EXPORT_SYMBOL(pxa2xx_ssp_dai); + +static int __devinit pxa2xx_ssp_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_dai *dai; + int ret; + + if (pdev->id >= PXA2XX_DAI_SSP_MAX) { + dev_err(&pdev->dev, "id %d is out of range\n", pdev->id); + return -EINVAL; + } + + dai = &pxa2xx_ssp_dai[pdev->id]; + dai->dev = &pdev->dev; + dai->name = "pxa2xx-ssp"; + dai->id = pdev->id; + dai->playback.channels_min = 1; + dai->playback.channels_max = 8; + dai->playback.rates = PXA2XX_SSP_RATES; + dai->playback.formats = PXA2XX_SSP_FORMATS; + dai->capture.channels_min = 1; + dai->capture.channels_max = 8; + dai->capture.rates = PXA2XX_SSP_RATES; + dai->capture.formats = PXA2XX_SSP_FORMATS; + dai->ops = &pxa2xx_ssp_dai_ops; + + ret = pxa_ssp_register_dai(dai); + return ret; +} + +static int __devexit pxa2xx_ssp_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_dai *dai; + + dai = &pxa2xx_ssp_dai[pdev->id]; + snd_soc_unregister_dai(dai); + return 0; +} + +static struct platform_driver pxa2xx_ssp_driver = { + .probe = pxa2xx_ssp_dev_probe, + .remove = __devexit_p(pxa2xx_ssp_dev_remove), + .driver = { + .name = "pxa2xx-ssp", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa2xx_ssp_init(void) +{ + return platform_driver_register(&pxa2xx_ssp_driver); +} +module_init(pxa2xx_ssp_init); + +static void __exit pxa2xx_ssp_exit(void) +{ + platform_driver_unregister(&pxa2xx_ssp_driver); +} +module_exit(pxa2xx_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown broonie@opensource.wolfsonmicro.com"); +MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-ssp.h b/sound/soc/pxa/pxa2xx-ssp.h new file mode 100644 index 0000000..2455bf4 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.h @@ -0,0 +1,59 @@ +/* + * ASoC PXA SSP port support + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __PXA2XX_SOC_SSP_H +#define __PXA2XX_SOC_SSP_H + +#define PXA2XX_SSP_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_88200 | SNDRV_PCM_RATE_96000) + +#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* pxa DAI SSP IDs */ +enum { + PXA2XX_DAI_SSP1, + PXA2XX_DAI_SSP2, + PXA2XX_DAI_SSP3, + PXA2XX_DAI_SSP4, + PXA2XX_DAI_SSP_MAX, +}; + +/* SSP clock sources */ +#define PXA2XX_SSP_CLK_PLL 0 +#define PXA2XX_SSP_CLK_EXT 1 +#define PXA2XX_SSP_CLK_NET 2 +#define PXA2XX_SSP_CLK_AUDIO 3 +#define PXA2XX_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA2XX_SSP_AUDIO_DIV_ACDS 0 +#define PXA2XX_SSP_AUDIO_DIV_SCDB 1 +#define PXA2XX_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA2XX_SSP_CLK_AUDIO_DIV_1 0 +#define PXA2XX_SSP_CLK_AUDIO_DIV_2 1 +#define PXA2XX_SSP_CLK_AUDIO_DIV_4 2 +#define PXA2XX_SSP_CLK_AUDIO_DIV_8 3 +#define PXA2XX_SSP_CLK_AUDIO_DIV_16 4 +#define PXA2XX_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA2XX_SSP_CLK_SCDB_4 0 +#define PXA2XX_SSP_CLK_SCDB_1 1 +#define PXA2XX_SSP_CLK_SCDB_8 2 + +#define PXA2XX_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa2xx_ssp_dai[PXA2XX_DAI_SSP_MAX]; + +#endif /* __PXA2XX_SOC_SSP_H */ diff --git a/sound/soc/pxa/ssp.c b/sound/soc/pxa/ssp.c new file mode 100644 index 0000000..444b643 --- /dev/null +++ b/sound/soc/pxa/ssp.c @@ -0,0 +1,298 @@ +/* + * ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2009-2010 Marvell International Ltd. + * Author: + * Haojian Zhuang haojian.zhuang@marvell.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-ssp.h> + +#include <plat/ssp.h> + +#include "ssp.h" + +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; +}; + +struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + width4 ? "32-bit" : "16-bit", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} +EXPORT_SYMBOL(pxa_ssp_get_dma_params); + +static int pxa_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_info *info = cpu_dai->private_data; + int ret = 0; + + if (!cpu_dai->active) { + info->dev.port = cpu_dai->id + 1; + info->dev.irq = NO_IRQ; + clk_enable(info->dev.ssp->clk); + ssp_disable(&info->dev); + } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } + return ret; +} + +static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_info *info = cpu_dai->private_data; + + if (!cpu_dai->active) { + ssp_disable(&info->dev); + clk_disable(info->dev.ssp->clk); + } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } +} + +#ifdef CONFIG_PM + +static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) +{ + struct ssp_info *info = cpu_dai->private_data; + + if (!cpu_dai->active) + clk_enable(info->dev.ssp->clk); + + ssp_save_state(&info->dev, &info->state); + clk_disable(info->dev.ssp->clk); + + return 0; +} + +static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) +{ + struct ssp_info *info = cpu_dai->private_data; + + clk_enable(info->dev.ssp->clk); + ssp_restore_state(&info->dev, &info->state); + + if (cpu_dai->active) + ssp_enable(&info->dev); + else + clk_disable(info->dev.ssp->clk); + + return 0; +} + +#else +#define pxa_ssp_suspend NULL +#define pxa_ssp_resume NULL +#endif + +/* + * 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_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 sscr0; + + sscr0 = 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 */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } + 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_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + u32 sscr1; + + sscr1 = ssp_read_reg(ssp, SSCR1); + if (tristate) + sscr1 &= ~SSCR1_TTE; + else + sscr1 |= SSCR1_TTE; + ssp_write_reg(ssp, SSCR1, sscr1); + + return 0; +} + +static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + struct ssp_info *info = cpu_dai->private_data; + struct ssp_device *ssp = info->dev.ssp; + int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + ssp_enable(&info->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + val = ssp_read_reg(ssp, SSSR); + ssp_write_reg(ssp, SSSR, val); + break; + case SNDRV_PCM_TRIGGER_START: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + ssp_enable(&info->dev); + break; + case SNDRV_PCM_TRIGGER_STOP: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + ssp_disable(&info->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int pxa_ssp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_info *info; + int ret; + + info = kzalloc(sizeof(struct ssp_info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + info->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); + if (info->dev.ssp == NULL) { + ret = -ENODEV; + goto err; + } + + info->dai_fmt = (unsigned int) -1; + dai->private_data = info; + + return 0; + +err: + kfree(info); + return ret; +} + +static void pxa_ssp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_info *info = dai->private_data; + ssp_free(info->dev.ssp); +} + +int pxa_ssp_register_dai(struct snd_soc_dai *dai) +{ + struct snd_soc_dai_ops *ops = dai->ops; + + ops->startup = pxa_ssp_startup; + ops->shutdown = pxa_ssp_shutdown; + ops->trigger = pxa_ssp_trigger; + ops->set_tdm_slot = pxa_ssp_set_dai_tdm_slot; + ops->set_tristate = pxa_ssp_set_dai_tristate; + + dai->probe = pxa_ssp_probe; + dai->remove = pxa_ssp_remove; + dai->suspend = pxa_ssp_suspend; + dai->resume = pxa_ssp_resume; + + return snd_soc_register_dai(dai); +} +EXPORT_SYMBOL(pxa_ssp_register_dai); diff --git a/sound/soc/pxa/ssp.h b/sound/soc/pxa/ssp.h new file mode 100644 index 0000000..314c06d --- /dev/null +++ b/sound/soc/pxa/ssp.h @@ -0,0 +1,42 @@ +/* + * ssp.h -- ALSA Soc Audio Layer Head file + * + * Copyright 2009-2010 Marvell International Ltd. + * Author: + * Haojian Zhuang haojian.zhuang@marvell.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __PXA_SSP_H +#define __PXA_SSP_H + +/* + * SSP audio data + */ +struct ssp_info { + struct ssp_dev dev; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + struct ssp_state state; +#endif +}; + +struct dai_ssp { + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + struct ssp_state state; +#endif +}; + +extern struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out); +extern int pxa_ssp_register_dai(struct snd_soc_dai *dai); + +#endif /* __PXA_SSP_H */ diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 0000000..3ee39d4 --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,193 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang haojian.zhuang@marvell.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#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 <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; + +static struct snd_soc_jack hs_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_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), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route 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 evb3_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET, + SND_JACK_BTN_0); + pm860x_hook_detect(codec, &hs_jack, SND_JACK_BTN_1, SND_JACK_BTN_2); + return 0; +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang haojian.zhuang@marvell.com"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); +MODULE_LICENSE("GPL"); +