[alsa-devel] [PATCH] Add ASoC Magician machine support.

Mark Brown broonie at opensource.wolfsonmicro.com
Tue Nov 20 10:25:43 CET 2007


From: Liam Girdwood <liam at localhost.localdomain>

Signed-off-by: Philipp Zabel <philipp.zabel at gmail.com>
Signed-off-by: Liam Girdwood <lg at opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig    |   11 +
 sound/soc/pxa/Makefile   |    2 +
 sound/soc/pxa/magician.c |  539 ++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 552 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/pxa/magician.c

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index bcb3aa0..3682f38 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -58,6 +58,17 @@ config SND_PXA2XX_SOC_TOSA
 	  Say Y if you want to add support for SoC audio on Sharp
 	  Zaurus SL-C6000x models (Tosa).
 
+config SND_PXA2XX_SOC_MAGICIAN
+	tristate "SoC Audio support for HTC Magician"
+	depends on SND_PXA2XX_SOC
+	select SND_PXA2XX_SOC_I2S
+	select SND_PXA2XX_SOC_SSP
+	select SND_SOC_UDA1380
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  HTC Magician.
+
+
 config SND_PXA2XX_SOC_AMESOM_TLV320
 	tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k"
 	depends on SND_PXA2XX_SOC && MACH_AMESOM
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 931bdc7..1faa751 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -15,9 +15,11 @@ snd-soc-poodle-objs := poodle.o
 snd-soc-tosa-objs := tosa.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-amesom-tlv320-objs := amesom_tlv320.o
+snd-soc-magician-objs := magician.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
 obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
\ No newline at end of file
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644
index 0000000..7eb671c
--- /dev/null
+++ b/sound/soc/pxa/magician.c
@@ -0,0 +1,539 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel at gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
+ *          Richard Purdie <richard at openedhand.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/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/hardware/scoop.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/magician.h>
+#include <asm/arch/magician_cpld.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+#include "pxa2xx-ssp.h"
+
+#define MAGICIAN_HP_ON     0
+#define MAGICIAN_HP_OFF    1
+
+#define MAGICIAN_SPK_ON    0
+#define MAGICIAN_SPK_OFF   1
+
+#define MAGICIAN_MIC       0
+#define MAGICIAN_MIC_EXT   1
+
+/*
+ * SSP GPIO's
+ */
+#define GPIO23_SSPSCLK_MD	(23 | GPIO_ALT_FN_2_OUT)
+#define GPIO24_SSPSFRM_MD	(24 | GPIO_ALT_FN_2_OUT)
+#define GPIO25_SSPTXD_MD	(25 | GPIO_ALT_FN_2_OUT)
+
+static int magician_hp_func = MAGICIAN_HP_OFF;
+static int magician_spk_func = MAGICIAN_SPK_ON;
+static int magician_in_sel = MAGICIAN_MIC;
+
+extern struct platform_device magician_cpld;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_set_endpoint(codec, "Speaker",
+			(magician_spk_func == MAGICIAN_SPK_ON));
+
+	snd_soc_dapm_set_endpoint(codec, "Headphone Jack",
+			(magician_hp_func == MAGICIAN_HP_ON));
+
+	switch (magician_in_sel) {
+	case MAGICIAN_MIC:
+		snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+		snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+		break;
+	case MAGICIAN_MIC_EXT:
+		snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+		snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
+		break;
+	}
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	magician_ext_control(codec);
+
+	return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
+				       struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int acps, acds, div4;
+	int ret = 0;
+
+	/*
+	 * Rate = SSPSCLK / (word size(16))
+	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+	 */
+	switch (params_rate(params)) {
+	case 8000:
+		acps = 32842000;
+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_32;	/* wrong - 32 bits/sample */
+		div4 = PXA2XX_SSP_CLK_SCDB_4;
+		break;
+	case 11025:
+		acps = 5622000;
+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_8;	/* 16 bits/sample, 1 slot */
+		div4 = PXA2XX_SSP_CLK_SCDB_4;
+		break;
+	case 22050:
+		acps = 5622000;
+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
+		div4 = PXA2XX_SSP_CLK_SCDB_4;
+		break;
+	case 44100:
+		acps = 11345000;
+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
+		div4 = PXA2XX_SSP_CLK_SCDB_4;
+		break;
+	case 48000:
+		acps = 12235000;
+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
+		div4 = PXA2XX_SSP_CLK_SCDB_4;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set audio clock as clock source */
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	/* set the SSP audio system clock ACDS divider */
+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+			PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	/* set the SSP audio system clock SCDB divider4 */
+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+			PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
+	if (ret < 0)
+		return ret;
+
+	/* set SSP audio pll clock */
+	ret = cpu_dai->dai_ops.set_pll(cpu_dai, 0, 0, acps);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * We have to enable the SSP port early so the UDA1380 can flush
+ * it's register cache. The UDA1380 can only write it's interpolator and
+ * decimator registers when the link is running.
+ */
+static int magician_playback_prepare(struct snd_pcm_substream *substream)
+{
+	/* enable SSP clock - is this needed ? */
+	SSCR0_P(1) |= SSCR0_SSE;
+
+	/* FIXME: ENABLE I2S */
+	SACR0 |= SACR0_BCKD;
+	SACR0 |= SACR0_ENB;
+	pxa_set_cken(CKEN8_I2S, 1);
+
+	return 0;
+}
+
+static int magician_playback_hw_free(struct snd_pcm_substream *substream)
+{
+	/* FIXME: DISABLE I2S */
+	SACR0 &= ~SACR0_ENB;
+	SACR0 &= ~SACR0_BCKD;
+	pxa_set_cken(CKEN8_I2S, 0);
+	return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
+				      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret = 0;
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as output */
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * We have to enable the I2S port early so the UDA1380 can flush
+ * it's register cache. The UDA1380 can only write it's interpolator and
+ * decimator registers when the link is running.
+ */
+static int magician_capture_prepare(struct snd_pcm_substream *substream)
+{
+	SACR0 |= SACR0_ENB;
+	return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+	.startup = magician_startup,
+	.hw_params = magician_capture_hw_params,
+	.prepare = magician_capture_prepare,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+	.startup = magician_startup,
+	.hw_params = magician_playback_hw_params,
+	.prepare = magician_playback_prepare,
+	.hw_free = magician_playback_hw_free,
+};
+
+static int magician_get_jack(struct snd_kcontrol * kcontrol,
+			     struct snd_ctl_elem_value * ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_hp_func;
+	return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol * kcontrol,
+			     struct snd_ctl_elem_value * ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (magician_hp_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_hp_func = ucontrol->value.integer.value[0];
+	magician_ext_control(codec);
+	return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol * kcontrol,
+			    struct snd_ctl_elem_value * ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_spk_func;
+	return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol * kcontrol,
+			    struct snd_ctl_elem_value * ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (magician_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_spk_func = ucontrol->value.integer.value[0];
+	magician_ext_control(codec);
+	return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol * kcontrol,
+			      struct snd_ctl_elem_value * ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_in_sel;
+	return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol * kcontrol,
+			      struct snd_ctl_elem_value * ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (magician_in_sel == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_in_sel = ucontrol->value.integer.value[0];
+
+	switch (magician_in_sel) {
+	case MAGICIAN_MIC:
+		magician_egpio_disable(&magician_cpld,
+				       EGPIO_NR_MAGICIAN_IN_SEL0);
+		magician_egpio_enable(&magician_cpld,
+				      EGPIO_NR_MAGICIAN_IN_SEL1);
+		break;
+	case MAGICIAN_MIC_EXT:
+		magician_egpio_disable(&magician_cpld,
+				       EGPIO_NR_MAGICIAN_IN_SEL0);
+		magician_egpio_disable(&magician_cpld,
+				       EGPIO_NR_MAGICIAN_IN_SEL1);
+	}
+
+	return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		magician_egpio_enable(&magician_cpld,
+				      EGPIO_NR_MAGICIAN_SPK_POWER);
+	else
+		magician_egpio_disable(&magician_cpld,
+				       EGPIO_NR_MAGICIAN_SPK_POWER);
+	return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		magician_egpio_enable(&magician_cpld,
+				      EGPIO_NR_MAGICIAN_EP_POWER);
+	else
+		magician_egpio_disable(&magician_cpld,
+				       EGPIO_NR_MAGICIAN_EP_POWER);
+	return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w, int event)
+{
+//	if (SND_SOC_DAPM_EVENT_ON(event))
+//		magician_egpio_enable(&magician_cpld,
+//			EGPIO_NR_MAGICIAN_MIC_POWER);
+//	else
+//		magician_egpio_disable(&magician_cpld,
+//			EGPIO_NR_MAGICIAN_MIC_POWER);
+	return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+	SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+	SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const char *audio_map[][3] = {
+
+	/* Headphone connected to VOUTL, VOUTR */
+	{"Headphone Jack", NULL, "VOUTL"},
+	{"Headphone Jack", NULL, "VOUTR"},
+
+	/* Speaker connected to VOUTL, VOUTR */
+	{"Speaker", NULL, "VOUTL"},
+	{"Speaker", NULL, "VOUTR"},
+
+	/* Mics are connected to VINM */
+	{"VINM", NULL, "Headset Mic"},
+	{"VINM", NULL, "Call Mic"},
+
+	{NULL, NULL, NULL},
+};
+
+static const char *hp_function[] = { "On", "Off" };
+static const char *spk_function[] = { "On", "Off" };
+static const char *input_select[] = { "Call Mic", "Headset Mic" };
+static const struct soc_enum magician_enum[] = {
+	SOC_ENUM_SINGLE_EXT(4, hp_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+	SOC_ENUM_SINGLE_EXT(2, input_select),
+};
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+	SOC_ENUM_EXT("Headphone Switch", magician_enum[0], magician_get_jack,
+			magician_set_hp),
+	SOC_ENUM_EXT("Speaker Switch", magician_enum[1], magician_get_spk,
+			magician_set_spk),
+	SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input,
+			magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	/* NC codec pins */
+	snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
+	snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
+
+	/* FIXME: is anything connected here? */
+	snd_soc_dapm_set_endpoint(codec, "VINL", 0);
+	snd_soc_dapm_set_endpoint(codec, "VINR", 0);
+
+	/* Add magician specific controls */
+	for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {
+		if ((err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&uda1380_magician_controls[i],
+				codec, NULL))) < 0)
+			return err;
+	}
+
+	/* Add magician specific widgets */
+	for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
+	}
+
+	/* Set up magician specific audio path interconnects */
+	for (i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+				audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+	.name = "uda1380",
+	.stream_name = "UDA1380 Playback",
+	.cpu_dai = &pxa_ssp_dai[0],
+	.codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+	.init = magician_uda1380_init,
+	.ops = &magician_playback_ops,
+},
+{
+	.name = "uda1380",
+	.stream_name = "UDA1380 Capture",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+	.ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_machine snd_soc_machine_magician = {
+	.name = "Magician",
+	.dai_link = magician_dai,
+	.num_links = ARRAY_SIZE(magician_dai),
+};
+
+/* magician audio private data */
+static struct uda1380_setup_data magician_uda1380_setup = {
+	.i2c_address = 0x18,
+	.dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+/* magician audio subsystem */
+static struct snd_soc_device magician_snd_devdata = {
+	.machine = &snd_soc_machine_magician,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_uda1380,
+	.codec_data = &magician_uda1380_setup,
+};
+
+static struct platform_device *magician_snd_device;
+
+static int __init magician_init(void)
+{
+	int ret;
+
+	if (!machine_is_magician())
+		return -ENODEV;
+
+	magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
+
+	/* we may need to have the clock running here - pH5 */
+	magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
+	udelay(5);
+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
+
+	/* correct place? we'll need it to talk to the uda1380 */
+	request_module("i2c-pxa");
+
+	magician_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!magician_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
+	magician_snd_devdata.dev = &magician_snd_device->dev;
+	ret = platform_device_add(magician_snd_device);
+
+	if (ret)
+		platform_device_put(magician_snd_device);
+
+	pxa_gpio_mode(GPIO23_SSPSCLK_MD);
+	pxa_gpio_mode(GPIO24_SSPSFRM_MD);
+	pxa_gpio_mode(GPIO25_SSPTXD_MD);
+
+	return ret;
+}
+
+static void __exit magician_exit(void)
+{
+	platform_device_unregister(magician_snd_device);
+
+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
-- 
1.5.3.5



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