[PATCH 2/3] ASoC: add tavorevb3 machine driver for 88pm860x
Haojian Zhuang
haojian.zhuang at marvell.com
Tue Aug 17 07:24:35 CEST 2010
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 at 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 at opensource.wolfsonmicro.com>
+ *
+ * Copyright 2009-2010 Marvell International Ltd.
+ * Author: Haojian Zhuang <haojian.zhuang at 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 at 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 at 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 at 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 at 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 at marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
+MODULE_LICENSE("GPL");
+
--
1.5.6.5
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