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-rw-r--r--sound/soc/pxa/Kconfig27
-rw-r--r--sound/soc/pxa/Makefile6
-rw-r--r--sound/soc/pxa/corgi.c58
-rw-r--r--sound/soc/pxa/e740_wm9705.c211
-rw-r--r--sound/soc/pxa/e750_wm9705.c187
-rw-r--r--sound/soc/pxa/e800_wm9712.c115
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c250
-rw-r--r--sound/soc/pxa/palm27x.c15
-rw-r--r--sound/soc/pxa/poodle.c56
-rw-r--r--sound/soc/pxa/pxa-ssp.c150
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c59
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c54
-rw-r--r--sound/soc/pxa/spitz.c14
-rw-r--r--sound/soc/pxa/tosa.c14
-rw-r--r--sound/soc/pxa/zylonite.c132
15 files changed, 1126 insertions, 222 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f82e1069947..5998ab366e8 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -61,6 +61,24 @@ 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_E740
+ tristate "SoC AC97 Audio support for e740"
+ depends on SND_PXA2XX_SOC && MACH_E740
+ select SND_SOC_WM9705
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ toshiba e740 PDA
+
+config SND_PXA2XX_SOC_E750
+ tristate "SoC AC97 Audio support for e750"
+ depends on SND_PXA2XX_SOC && MACH_E750
+ select SND_SOC_WM9705
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ toshiba e750 PDA
+
config SND_PXA2XX_SOC_E800
tristate "SoC AC97 Audio support for e800"
depends on SND_PXA2XX_SOC && MACH_E800
@@ -97,3 +115,12 @@ config SND_SOC_ZYLONITE
help
Say Y if you want to add support for SoC audio on the
Marvell Zylonite reference platform.
+
+config SND_PXA2XX_SOC_MIOA701
+ tristate "SoC Audio support for MIO A701"
+ depends on SND_PXA2XX_SOC && MACH_MIOA701
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9713
+ help
+ Say Y if you want to add support for SoC audio on the
+ MIO A701.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 08a9f279772..8ed881c5e5c 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,17 +13,23 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
snd-soc-corgi-objs := corgi.o
snd-soc-poodle-objs := poodle.o
snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
snd-soc-zylonite-objs := zylonite.o
+snd-soc-mioa701-objs := mioa701_wm9713.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 1ba25a55952..02263e5d8f0 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -16,6 +16,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
+#include <linux/i2c.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/gpio.h>
@@ -100,7 +101,7 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
static int corgi_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->codec;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
/* check the jack status at stream startup */
corgi_ext_control(codec);
@@ -275,18 +276,16 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
*/
static int corgi_wm8731_init(struct snd_soc_codec *codec)
{
- int i, err;
+ int err;
snd_soc_dapm_nc_pin(codec, "LLINEIN");
snd_soc_dapm_nc_pin(codec, "RLINEIN");
/* Add corgi specific controls */
- for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
+ err = snd_soc_add_controls(codec, wm8731_corgi_controls,
+ ARRAY_SIZE(wm8731_corgi_controls));
+ if (err < 0)
+ return err;
/* Add corgi specific widgets */
snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
@@ -317,19 +316,44 @@ static struct snd_soc_card snd_soc_corgi = {
.num_links = 1,
};
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
- .i2c_bus = 0,
- .i2c_address = 0x1b,
-};
-
/* corgi audio subsystem */
static struct snd_soc_device corgi_snd_devdata = {
.card = &snd_soc_corgi,
.codec_dev = &soc_codec_dev_wm8731,
- .codec_data = &corgi_wm8731_setup,
};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = 0x1b;
+ strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(0);
+ if (!adapter) {
+ printk(KERN_ERR "can't get i2c adapter 0\n");
+ return -ENODEV;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ printk(KERN_ERR "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
static struct platform_device *corgi_snd_device;
static int __init corgi_init(void)
@@ -340,6 +364,10 @@ static int __init corgi_init(void)
machine_is_husky()))
return -ENODEV;
+ ret = wm8731_i2c_register();
+ if (ret != 0)
+ return ret;
+
corgi_snd_device = platform_device_alloc("soc-audio", -1);
if (!corgi_snd_device)
return -ENOMEM;
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
new file mode 100644
index 00000000000..7cd2f89d7b1
--- /dev/null
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -0,0 +1,211 @@
+/*
+ * e740-wm9705.c -- SoC audio for e740
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.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; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+
+#define E740_AUDIO_OUT 1
+#define E740_AUDIO_IN 2
+
+static int e740_audio_power;
+
+static void e740_sync_audio_power(int status)
+{
+ gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status);
+ gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0);
+ gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0);
+}
+
+static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ e740_audio_power |= E740_AUDIO_IN;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ e740_audio_power &= ~E740_AUDIO_IN;
+
+ e740_sync_audio_power(e740_audio_power);
+
+ return 0;
+}
+
+static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ e740_audio_power |= E740_AUDIO_OUT;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ e740_audio_power &= ~E740_AUDIO_OUT;
+
+ e740_sync_audio_power(e740_audio_power);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Output Amp", NULL, "LOUT"},
+ {"Output Amp", NULL, "ROUT"},
+ {"Output Amp", NULL, "MONOOUT"},
+
+ {"Speaker", NULL, "Output Amp"},
+ {"Headphone Jack", NULL, "Output Amp"},
+
+ {"MIC1", NULL, "Mic Amp"},
+ {"Mic Amp", NULL, "Mic (Internal)"},
+};
+
+static int e740_ac97_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_nc_pin(codec, "HPOUTL");
+ snd_soc_dapm_nc_pin(codec, "HPOUTR");
+ snd_soc_dapm_nc_pin(codec, "PHONE");
+ snd_soc_dapm_nc_pin(codec, "LINEINL");
+ snd_soc_dapm_nc_pin(codec, "LINEINR");
+ snd_soc_dapm_nc_pin(codec, "CDINL");
+ snd_soc_dapm_nc_pin(codec, "CDINR");
+ snd_soc_dapm_nc_pin(codec, "PCBEEP");
+ snd_soc_dapm_nc_pin(codec, "MIC2");
+
+ snd_soc_dapm_new_controls(codec, e740_dapm_widgets,
+ ARRAY_SIZE(e740_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link e740_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+ .init = e740_ac97_init,
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+ },
+};
+
+static struct snd_soc_card e740 = {
+ .name = "Toshiba e740",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = e740_dai,
+ .num_links = ARRAY_SIZE(e740_dai),
+};
+
+static struct snd_soc_device e740_snd_devdata = {
+ .card = &e740,
+ .codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e740_snd_device;
+
+static int __init e740_init(void)
+{
+ int ret;
+
+ if (!machine_is_e740())
+ return -ENODEV;
+
+ ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp");
+ if (ret)
+ return ret;
+
+ ret = gpio_request(GPIO_E740_AMP_ON, "Output amp");
+ if (ret)
+ goto free_mic_amp_gpio;
+
+ ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power");
+ if (ret)
+ goto free_op_amp_gpio;
+
+ /* Disable audio */
+ ret = gpio_direction_output(GPIO_E740_MIC_ON, 0);
+ if (ret)
+ goto free_apwr_gpio;
+ ret = gpio_direction_output(GPIO_E740_AMP_ON, 0);
+ if (ret)
+ goto free_apwr_gpio;
+ ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1);
+ if (ret)
+ goto free_apwr_gpio;
+
+ e740_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!e740_snd_device) {
+ ret = -ENOMEM;
+ goto free_apwr_gpio;
+ }
+
+ platform_set_drvdata(e740_snd_device, &e740_snd_devdata);
+ e740_snd_devdata.dev = &e740_snd_device->dev;
+ ret = platform_device_add(e740_snd_device);
+
+ if (!ret)
+ return 0;
+
+/* Fail gracefully */
+ platform_device_put(e740_snd_device);
+free_apwr_gpio:
+ gpio_free(GPIO_E740_WM9705_nAVDD2);
+free_op_amp_gpio:
+ gpio_free(GPIO_E740_AMP_ON);
+free_mic_amp_gpio:
+ gpio_free(GPIO_E740_MIC_ON);
+
+ return ret;
+}
+
+static void __exit e740_exit(void)
+{
+ platform_device_unregister(e740_snd_device);
+}
+
+module_init(e740_init);
+module_exit(e740_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e740");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
new file mode 100644
index 00000000000..8dceccc5e05
--- /dev/null
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -0,0 +1,187 @@
+/*
+ * e750-wm9705.c -- SoC audio for e750
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.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; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1);
+
+ return 0;
+}
+
+static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E750_HP_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E750_HP_AMP_OFF, 1);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Amp", NULL, "HPOUTL"},
+ {"Headphone Amp", NULL, "HPOUTR"},
+ {"Headphone Jack", NULL, "Headphone Amp"},
+
+ {"Speaker Amp", NULL, "MONOOUT"},
+ {"Speaker", NULL, "Speaker Amp"},
+
+ {"MIC1", NULL, "Mic (Internal)"},
+};
+
+static int e750_ac97_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_nc_pin(codec, "LOUT");
+ snd_soc_dapm_nc_pin(codec, "ROUT");
+ snd_soc_dapm_nc_pin(codec, "PHONE");
+ snd_soc_dapm_nc_pin(codec, "LINEINL");
+ snd_soc_dapm_nc_pin(codec, "LINEINR");
+ snd_soc_dapm_nc_pin(codec, "CDINL");
+ snd_soc_dapm_nc_pin(codec, "CDINR");
+ snd_soc_dapm_nc_pin(codec, "PCBEEP");
+ snd_soc_dapm_nc_pin(codec, "MIC2");
+
+ snd_soc_dapm_new_controls(codec, e750_dapm_widgets,
+ ARRAY_SIZE(e750_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link e750_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+ .init = e750_ac97_init,
+ /* use ops to check startup state */
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+ },
+};
+
+static struct snd_soc_card e750 = {
+ .name = "Toshiba e750",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = e750_dai,
+ .num_links = ARRAY_SIZE(e750_dai),
+};
+
+static struct snd_soc_device e750_snd_devdata = {
+ .card = &e750,
+ .codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e750_snd_device;
+
+static int __init e750_init(void)
+{
+ int ret;
+
+ if (!machine_is_e750())
+ return -ENODEV;
+
+ ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp");
+ if (ret)
+ return ret;
+
+ ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp");
+ if (ret)
+ goto free_hp_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
+ e750_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!e750_snd_device) {
+ ret = -ENOMEM;
+ goto free_spk_amp_gpio;
+ }
+
+ platform_set_drvdata(e750_snd_device, &e750_snd_devdata);
+ e750_snd_devdata.dev = &e750_snd_device->dev;
+ ret = platform_device_add(e750_snd_device);
+
+ if (!ret)
+ return 0;
+
+/* Fail gracefully */
+ platform_device_put(e750_snd_device);
+free_spk_amp_gpio:
+ gpio_free(GPIO_E750_SPK_AMP_OFF);
+free_hp_amp_gpio:
+ gpio_free(GPIO_E750_HP_AMP_OFF);
+
+ return ret;
+}
+
+static void __exit e750_exit(void)
+{
+ platform_device_unregister(e750_snd_device);
+ gpio_free(GPIO_E750_SPK_AMP_OFF);
+ gpio_free(GPIO_E750_HP_AMP_OFF);
+}
+
+module_init(e750_init);
+module_exit(e750_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e750");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 2e3386dfa0f..bc019cdce42 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -1,8 +1,6 @@
/*
* e800-wm9712.c -- SoC audio for e800
*
- * Based on tosa.c
- *
* Copyright 2007 (c) Ian Molton <spyro@f2s.com>
*
* This program is free software; you can redistribute it and/or modify it
@@ -13,7 +11,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/device.h>
+#include <linux/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -21,23 +19,85 @@
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
#include "../codecs/wm9712.h"
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
-static struct snd_soc_card e800;
+static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
-static struct snd_soc_dai_link e800_dai[] = {
+ return 0;
+}
+
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (event & SND_SOC_DAPM_PRE_PMU)
+ gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
+ SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+ {"Headphone Jack", NULL, "Headphone Amp"},
+
+ {"Speaker Amp", NULL, "MONOOUT"},
+ {"Speaker", NULL, "Speaker Amp"},
+
+ {"MIC1", NULL, "Mic (Internal1)"},
+ {"MIC2", NULL, "Mic (Internal2)"},
+};
+
+static int e800_ac97_init(struct snd_soc_codec *codec)
{
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
- .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
-},
+ snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
+ ARRAY_SIZE(e800_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link e800_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .init = e800_ac97_init,
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ },
};
static struct snd_soc_card e800 = {
@@ -61,6 +121,22 @@ static int __init e800_init(void)
if (!machine_is_e800())
return -ENODEV;
+ ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp");
+ if (ret)
+ return ret;
+
+ ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp");
+ if (ret)
+ goto free_hp_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
+ ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1);
+ if (ret)
+ goto free_spk_amp_gpio;
+
e800_snd_device = platform_device_alloc("soc-audio", -1);
if (!e800_snd_device)
return -ENOMEM;
@@ -69,8 +145,15 @@ static int __init e800_init(void)
e800_snd_devdata.dev = &e800_snd_device->dev;
ret = platform_device_add(e800_snd_device);
- if (ret)
- platform_device_put(e800_snd_device);
+ if (!ret)
+ return 0;
+
+/* Fail gracefully */
+ platform_device_put(e800_snd_device);
+free_spk_amp_gpio:
+ gpio_free(GPIO_E800_SPK_AMP_ON);
+free_hp_amp_gpio:
+ gpio_free(GPIO_E800_HP_AMP_OFF);
return ret;
}
@@ -78,6 +161,8 @@ static int __init e800_init(void)
static void __exit e800_exit(void)
{
platform_device_unregister(e800_snd_device);
+ gpio_free(GPIO_E800_SPK_AMP_ON);
+ gpio_free(GPIO_E800_HP_AMP_OFF);
}
module_init(e800_init);
@@ -86,4 +171,4 @@ module_exit(e800_exit);
/* Module information */
MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
MODULE_DESCRIPTION("ALSA SoC driver for e800");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
new file mode 100644
index 00000000000..19eda8bbfda
--- /dev/null
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -0,0 +1,250 @@
+/*
+ * Handles the Mitac mioa701 SoC system
+ *
+ * Copyright (C) 2008 Robert Jarzmik
+ *
+ * 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 in version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * This is a little schema of the sound interconnections :
+ *
+ * Sagem X200 Wolfson WM9713
+ * +--------+ +-------------------+ Rear Speaker
+ * | | | | /-+
+ * | +--->----->---+MONOIN SPKL+--->----+-+ |
+ * | GSM | | | | | |
+ * | +--->----->---+PCBEEP SPKR+--->----+-+ |
+ * | CHIP | | | \-+
+ * | +---<-----<---+MONO |
+ * | | | | Front Speaker
+ * +--------+ | | /-+
+ * | HPL+--->----+-+ |
+ * | | | | |
+ * | OUT3+--->----+-+ |
+ * | | \-+
+ * | |
+ * | | Front Micro
+ * | | +
+ * | MIC1+-----<--+o+
+ * | | +
+ * +-------------------+ ---
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "../codecs/wm9713.h"
+
+#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x)
+
+#define AC97_GPIO_PULL 0x58
+
+/* Use GPIO8 for rear speaker amplifier */
+static int rear_amp_power(struct snd_soc_codec *codec, int power)
+{
+ unsigned short reg;
+
+ if (power) {
+ reg = snd_soc_read(codec, AC97_GPIO_CFG);
+ snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100);
+ reg = snd_soc_read(codec, AC97_GPIO_PULL);
+ snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15));
+ } else {
+ reg = snd_soc_read(codec, AC97_GPIO_CFG);
+ snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100);
+ reg = snd_soc_read(codec, AC97_GPIO_PULL);
+ snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15));
+ }
+
+ return 0;
+}
+
+static int rear_amp_event(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kctl, int event)
+{
+ struct snd_soc_codec *codec = widget->codec;
+
+ return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event));
+}
+
+/* mioa701 machine dapm widgets */
+static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Front Speaker", NULL),
+ SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event),
+ SND_SOC_DAPM_MIC("Headset", NULL),
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Front Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Call Mic */
+ {"Mic Bias", NULL, "Front Mic"},
+ {"MIC1", NULL, "Mic Bias"},
+
+ /* Headset Mic */
+ {"LINEL", NULL, "Headset Mic"},
+ {"LINER", NULL, "Headset Mic"},
+
+ /* GSM Module */
+ {"MONOIN", NULL, "GSM Line Out"},
+ {"PCBEEP", NULL, "GSM Line Out"},
+ {"GSM Line In", NULL, "MONO"},
+
+ /* headphone connected to HPL, HPR */
+ {"Headset", NULL, "HPL"},
+ {"Headset", NULL, "HPR"},
+
+ /* front speaker connected to HPL, OUT3 */
+ {"Front Speaker", NULL, "HPL"},
+ {"Front Speaker", NULL, "OUT3"},
+
+ /* rear speaker connected to SPKL, SPKR */
+ {"Rear Speaker", NULL, "SPKL"},
+ {"Rear Speaker", NULL, "SPKR"},
+};
+
+static int mioa701_wm9713_init(struct snd_soc_codec *codec)
+{
+ unsigned short reg;
+
+ /* Add mioa701 specific widgets */
+ snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+
+ /* Set up mioa701 specific audio path audio_mapnects */
+ snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map));
+
+ /* Prepare GPIO8 for rear speaker amplifier */
+ reg = codec->read(codec, AC97_GPIO_CFG);
+ codec->write(codec, AC97_GPIO_CFG, reg | 0x0100);
+
+ /* Prepare MIC input */
+ reg = codec->read(codec, AC97_3D_CONTROL);
+ codec->write(codec, AC97_3D_CONTROL, reg | 0xc000);
+
+ snd_soc_dapm_enable_pin(codec, "Front Speaker");
+ snd_soc_dapm_enable_pin(codec, "Rear Speaker");
+ snd_soc_dapm_enable_pin(codec, "Front Mic");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_ops mioa701_ops;
+
+static struct snd_soc_dai_link mioa701_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+ .init = mioa701_wm9713_init,
+ .ops = &mioa701_ops,
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+ .ops = &mioa701_ops,
+ },
+};
+
+static struct snd_soc_card mioa701 = {
+ .name = "MioA701",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = mioa701_dai,
+ .num_links = ARRAY_SIZE(mioa701_dai),
+};
+
+static struct snd_soc_device mioa701_snd_devdata = {
+ .card = &mioa701,
+ .codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *mioa701_snd_device;
+
+static int mioa701_wm9713_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ if (!machine_is_mioa701())
+ return -ENODEV;
+
+ dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+ "lead to overheating and possible destruction of your device."
+ "Do not use without a good knowledge of mio's board design!\n");
+
+ mioa701_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!mioa701_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata);
+ mioa701_snd_devdata.dev = &mioa701_snd_device->dev;
+
+ ret = platform_device_add(mioa701_snd_device);
+ if (!ret)
+ return 0;
+
+ platform_device_put(mioa701_snd_device);
+ return ret;
+}
+
+static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
+{
+ platform_device_unregister(mioa701_snd_device);
+ return 0;
+}
+
+static struct platform_driver mioa701_wm9713_driver = {
+ .probe = mioa701_wm9713_probe,
+ .remove = __devexit_p(mioa701_wm9713_remove),
+ .driver = {
+ .name = "mioa701-wm9713",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init mioa701_asoc_init(void)
+{
+ return platform_driver_register(&mioa701_wm9713_driver);
+}
+
+static void __exit mioa701_asoc_exit(void)
+{
+ platform_driver_unregister(&mioa701_wm9713_driver);
+}
+
+module_init(mioa701_asoc_init);
+module_exit(mioa701_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
+MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 4a9cf3083af..48a73f64500 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -55,7 +55,7 @@ static void palm27x_ext_control(struct snd_soc_codec *codec)
static int palm27x_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->codec;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
/* check the jack status at stream startup */
palm27x_ext_control(codec);
@@ -146,19 +146,16 @@ static const struct snd_kcontrol_new palm27x_controls[] = {
static int palm27x_ac97_init(struct snd_soc_codec *codec)
{
- int i, err;
+ int err;
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "MONOOUT");
/* add palm27x specific controls */
- for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&palm27x_controls[i],
- codec, NULL));
- if (err < 0)
- return err;
- }
+ err = snd_soc_add_controls(codec, palm27x_controls,
+ ARRAY_SIZE(palm27x_controls));
+ if (err < 0)
+ return err;
/* add palm27x specific widgets */
snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6e9827189ff..ef7c6c8dc8f 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -17,6 +17,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
+#include <linux/i2c.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
@@ -77,7 +78,7 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
static int poodle_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->codec;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
/* check the jack status at stream startup */
poodle_ext_control(codec);
@@ -240,19 +241,17 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
*/
static int poodle_wm8731_init(struct snd_soc_codec *codec)
{
- int i, err;
+ int err;
snd_soc_dapm_nc_pin(codec, "LLINEIN");
snd_soc_dapm_nc_pin(codec, "RLINEIN");
snd_soc_dapm_enable_pin(codec, "MICIN");
/* Add poodle specific controls */
- for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
+ err = snd_soc_add_controls(codec, wm8731_poodle_controls,
+ ARRAY_SIZE(wm8731_poodle_controls));
+ if (err < 0)
+ return err;
/* Add poodle specific widgets */
snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
@@ -283,17 +282,42 @@ static struct snd_soc_card snd_soc_poodle = {
.num_links = 1,
};
-/* poodle audio private data */
-static struct wm8731_setup_data poodle_wm8731_setup = {
- .i2c_bus = 0,
- .i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = 0x1b;
+ strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(0);
+ if (!adapter) {
+ printk(KERN_ERR "can't get i2c adapter 0\n");
+ return -ENODEV;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ printk(KERN_ERR "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ return -ENODEV;
+ }
+
+ return 0;
+}
/* poodle audio subsystem */
static struct snd_soc_device poodle_snd_devdata = {
.card = &snd_soc_poodle,
.codec_dev = &soc_codec_dev_wm8731,
- .codec_data = &poodle_wm8731_setup,
};
static struct platform_device *poodle_snd_device;
@@ -305,6 +329,10 @@ static int __init poodle_init(void)
if (!machine_is_poodle())
return -ENODEV;
+ ret = wm8731_i2c_register();
+ if (ret != 0)
+ return ret;
+
locomo_gpio_set_dir(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_AMP_ON, 0);
/* should we mute HP at startup - burning power ?*/
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 73cb6b4c2f2..b0bf40973d5 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -1,4 +1,3 @@
-#define DEBUG
/*
* pxa-ssp.c -- ALSA Soc Audio Layer
*
@@ -21,6 +20,8 @@
#include <linux/clk.h>
#include <linux/io.h>
+#include <asm/irq.h>
+
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
@@ -221,9 +222,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
int ret = 0;
if (!cpu_dai->active) {
- ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
- if (ret < 0)
- return ret;
+ priv->dev.port = cpu_dai->id + 1;
+ priv->dev.irq = NO_IRQ;
+ clk_enable(priv->dev.ssp->clk);
ssp_disable(&priv->dev);
}
return ret;
@@ -238,7 +239,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
if (!cpu_dai->active) {
ssp_disable(&priv->dev);
- ssp_exit(&priv->dev);
+ clk_disable(priv->dev.ssp->clk);
}
}
@@ -298,7 +299,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int val;
u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
- ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
dev_dbg(&ssp->pdev->dev,
"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
@@ -326,7 +327,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case PXA_SSP_CLK_AUDIO:
priv->sysclk = 0;
ssp_set_scr(&priv->dev, 1);
- sscr0 |= SSCR0_ADC;
+ sscr0 |= SSCR0_ACS;
break;
default:
return -ENODEV;
@@ -520,9 +521,20 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
u32 sscr1;
u32 sspsp;
+ /* check if we need to change anything at all */
+ if (priv->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_ADC);
+ (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
sspsp = 0;
@@ -545,18 +557,18 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- sscr0 |= SSCR0_MOD | SSCR0_PSP;
+ sscr0 |= SSCR0_PSP;
sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+ /* See hw_params() */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
- sspsp |= SSPSP_FSRT;
+ sspsp |= SSPSP_SFRMP;
break;
case SND_SOC_DAIFMT_NB_IF:
- sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
break;
case SND_SOC_DAIFMT_IB_IF:
- sspsp |= SSPSP_SFRMP;
+ sspsp |= SSPSP_SCMODE(3);
break;
default:
return -EINVAL;
@@ -642,34 +654,65 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
sscr0 |= SSCR0_FPCKE;
#endif
sscr0 |= SSCR0_DataSize(16);
- if (params_channels(params) > 1)
- sscr0 |= SSCR0_EDSS;
break;
case SNDRV_PCM_FORMAT_S24_LE:
sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
- /* we must be in network mode (2 slots) for 24 bit stereo */
break;
case SNDRV_PCM_FORMAT_S32_LE:
sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
- /* we must be in network mode (2 slots) for 32 bit stereo */
break;
}
ssp_write_reg(ssp, SSCR0, sscr0);
switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- /* Cleared when the DAI format is set */
- sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+ sspsp = ssp_read_reg(ssp, SSPSP);
+
+ if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(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.
+ */
+
+#ifdef CONFIG_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
+ return -EINVAL;
+#endif
+ } 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;
}
- /* We always use a network mode so we always require TDM slots
+ /* When we use a network mode, we always require TDM slots
* - complain loudly and fail if they've not been set up yet.
*/
- if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+ if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
return -EINVAL;
}
@@ -751,7 +794,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
if (!priv)
return -ENOMEM;
- priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+ priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio");
if (priv->dev.ssp == NULL) {
ret = -ENODEV;
goto err_priv;
@@ -782,6 +825,19 @@ static void pxa_ssp_remove(struct platform_device *pdev,
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
+static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+ .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_clkdiv = pxa_ssp_set_dai_clkdiv,
+ .set_pll = pxa_ssp_set_dai_pll,
+ .set_fmt = pxa_ssp_set_dai_fmt,
+ .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+ .set_tristate = pxa_ssp_set_dai_tristate,
+};
+
struct snd_soc_dai pxa_ssp_dai[] = {
{
.name = "pxa2xx-ssp1",
@@ -802,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
- .ops = {
- .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_clkdiv = pxa_ssp_set_dai_clkdiv,
- .set_pll = pxa_ssp_set_dai_pll,
- .set_fmt = pxa_ssp_set_dai_fmt,
- .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
- .set_tristate = pxa_ssp_set_dai_tristate,
- },
+ .ops = &pxa_ssp_dai_ops,
},
{ .name = "pxa2xx-ssp2",
.id = 1,
@@ -833,18 +878,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
- .ops = {
- .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_clkdiv = pxa_ssp_set_dai_clkdiv,
- .set_pll = pxa_ssp_set_dai_pll,
- .set_fmt = pxa_ssp_set_dai_fmt,
- .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
- .set_tristate = pxa_ssp_set_dai_tristate,
- },
+ .ops = &pxa_ssp_dai_ops,
},
{
.name = "pxa2xx-ssp3",
@@ -865,18 +899,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
- .ops = {
- .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_clkdiv = pxa_ssp_set_dai_clkdiv,
- .set_pll = pxa_ssp_set_dai_pll,
- .set_fmt = pxa_ssp_set_dai_fmt,
- .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
- .set_tristate = pxa_ssp_set_dai_tristate,
- },
+ .ops = &pxa_ssp_dai_ops,
},
{
.name = "pxa2xx-ssp4",
@@ -897,18 +920,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
.rates = PXA_SSP_RATES,
.formats = PXA_SSP_FORMATS,
},
- .ops = {
- .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_clkdiv = pxa_ssp_set_dai_clkdiv,
- .set_pll = pxa_ssp_set_dai_pll,
- .set_fmt = pxa_ssp_set_dai_fmt,
- .set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
- .set_tristate = pxa_ssp_set_dai_tristate,
- },
+ .ops = &pxa_ssp_dai_ops,
},
};
EXPORT_SYMBOL_GPL(pxa_ssp_dai);
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 812c2b4d3e0..01c21c6cdbb 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -106,13 +106,13 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
static int pxa2xx_ac97_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- return pxa2xx_ac97_hw_probe(pdev);
+ return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
}
static void pxa2xx_ac97_remove(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- pxa2xx_ac97_hw_remove(pdev);
+ pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
}
static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
@@ -164,6 +164,18 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)
+static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
+ .hw_params = pxa2xx_ac97_hw_params,
+};
+
+static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
+ .hw_params = pxa2xx_ac97_hw_aux_params,
+};
+
+static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
+ .hw_params = pxa2xx_ac97_hw_mic_params,
+};
+
/*
* There is only 1 physical AC97 interface for pxa2xx, but it
* has extra fifo's that can be used for aux DACs and ADCs.
@@ -189,8 +201,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
.channels_max = 2,
.rates = PXA2XX_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .ops = {
- .hw_params = pxa2xx_ac97_hw_params,},
+ .ops = &pxa_ac97_hifi_dai_ops,
},
{
.name = "pxa2xx-ac97-aux",
@@ -208,8 +219,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
.channels_max = 1,
.rates = PXA2XX_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .ops = {
- .hw_params = pxa2xx_ac97_hw_aux_params,},
+ .ops = &pxa_ac97_aux_dai_ops,
},
{
.name = "pxa2xx-ac97-mic",
@@ -221,23 +231,52 @@ struct snd_soc_dai pxa_ac97_dai[] = {
.channels_max = 1,
.rates = PXA2XX_AC97_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .ops = {
- .hw_params = pxa2xx_ac97_hw_mic_params,},
+ .ops = &pxa_ac97_mic_dai_ops,
},
};
EXPORT_SYMBOL_GPL(pxa_ac97_dai);
EXPORT_SYMBOL_GPL(soc_ac97_ops);
-static int __init pxa_ac97_init(void)
+static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev)
{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++)
+ pxa_ac97_dai[i].dev = &pdev->dev;
+
+ /* 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 snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
}
+
+static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+
+ return 0;
+}
+
+static struct platform_driver pxa2xx_ac97_driver = {
+ .probe = pxa2xx_ac97_dev_probe,
+ .remove = __devexit_p(pxa2xx_ac97_dev_remove),
+ .driver = {
+ .name = "pxa2xx-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init pxa_ac97_init(void)
+{
+ return platform_driver_register(&pxa2xx_ac97_driver);
+}
module_init(pxa_ac97_init);
static void __exit pxa_ac97_exit(void)
{
- snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+ platform_driver_unregister(&pxa2xx_ac97_driver);
}
module_exit(pxa_ac97_exit);
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 517991fb109..e6c24408c5f 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -25,20 +25,11 @@
#include <mach/hardware.h>
#include <mach/pxa-regs.h>
-#include <mach/pxa2xx-gpio.h>
#include <mach/audio.h>
#include "pxa2xx-pcm.h"
#include "pxa2xx-i2s.h"
-struct pxa2xx_gpio {
- u32 sys;
- u32 rx;
- u32 tx;
- u32 clk;
- u32 frm;
-};
-
/*
* I2S Controller Register and Bit Definitions
*/
@@ -106,21 +97,6 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
DCMD_BURST32 | DCMD_WIDTH4,
};
-static struct pxa2xx_gpio gpio_bus[] = {
- { /* I2S SoC Slave */
- .rx = GPIO29_SDATA_IN_I2S_MD,
- .tx = GPIO30_SDATA_OUT_I2S_MD,
- .clk = GPIO28_BITCLK_IN_I2S_MD,
- .frm = GPIO31_SYNC_I2S_MD,
- },
- { /* I2S SoC Master */
- .rx = GPIO29_SDATA_IN_I2S_MD,
- .tx = GPIO30_SDATA_OUT_I2S_MD,
- .clk = GPIO28_BITCLK_OUT_I2S_MD,
- .frm = GPIO31_SYNC_I2S_MD,
- },
-};
-
static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -181,9 +157,6 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
if (clk_id != PXA2XX_I2S_SYSCLK)
return -ENODEV;
- if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
- pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
-
return 0;
}
@@ -194,10 +167,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
- pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
- pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
- pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
- pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
BUG_ON(IS_ERR(clk_i2s));
clk_enable(clk_i2s);
pxa_i2s_wait();
@@ -335,6 +304,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+static struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+ .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,
+};
+
struct snd_soc_dai pxa_i2s_dai = {
.name = "pxa2xx-i2s",
.id = 0,
@@ -350,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = {
.channels_max = 2,
.rates = PXA2XX_I2S_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .ops = {
- .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,
- },
+ .ops = &pxa_i2s_dai_ops,
};
EXPORT_SYMBOL_GPL(pxa_i2s_dai);
@@ -398,11 +369,6 @@ static struct platform_driver pxa2xx_i2s_driver = {
static int __init pxa2xx_i2s_init(void)
{
- if (cpu_is_pxa27x())
- gpio_bus[1].sys = GPIO113_I2S_SYSCLK_MD;
- else
- gpio_bus[1].sys = GPIO32_SYSCLK_I2S_MD;
-
clk_i2s = ERR_PTR(-ENOENT);
return platform_driver_register(&pxa2xx_i2s_driver);
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index a3b9e6bdf97..6ca9f53080c 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -109,7 +109,7 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
static int spitz_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->codec;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
/* check the jack status at stream startup */
spitz_ext_control(codec);
@@ -278,7 +278,7 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
*/
static int spitz_wm8750_init(struct snd_soc_codec *codec)
{
- int i, err;
+ int err;
/* NC codec pins */
snd_soc_dapm_nc_pin(codec, "RINPUT1");
@@ -290,12 +290,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
snd_soc_dapm_nc_pin(codec, "MONO1");
/* Add spitz specific controls */
- for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
- if (err < 0)
- return err;
- }
+ err = snd_soc_add_controls(codec, wm8750_spitz_controls,
+ ARRAY_SIZE(wm8750_spitz_controls));
+ if (err < 0)
+ return err;
/* Add spitz specific widgets */
snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index c77194f74c9..fc781374b1b 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -82,7 +82,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
static int tosa_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->codec;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
/* check the jack status at stream startup */
tosa_ext_control(codec);
@@ -188,18 +188,16 @@ static const struct snd_kcontrol_new tosa_controls[] = {
static int tosa_ac97_init(struct snd_soc_codec *codec)
{
- int i, err;
+ int err;
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "MONOOUT");
/* add tosa specific controls */
- for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
- err = snd_ctl_add(codec->card,
- snd_soc_cnew(&tosa_controls[i],codec, NULL));
- if (err < 0)
- return err;
- }
+ err = snd_soc_add_controls(codec, tosa_controls,
+ ARRAY_SIZE(tosa_controls));
+ if (err < 0)
+ return err;
/* add tosa specific widgets */
snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index f8e9ecd589d..9a386b4c4ed 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -14,6 +14,7 @@
#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>
@@ -26,6 +27,17 @@
#include "pxa2xx-ac97.h"
#include "pxa-ssp.h"
+/*
+ * There is a physical switch SW15 on the board which changes the MCLK
+ * for the WM9713 between the standard AC97 master clock and the
+ * output of the CLK_POUT signal from the PXA.
+ */
+static int clk_pout;
+module_param(clk_pout, int, 0);
+MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
+
+static struct clk *pout;
+
static struct snd_soc_card zylonite;
static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
@@ -61,10 +73,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int zylonite_wm9713_init(struct snd_soc_codec *codec)
{
- /* Currently we only support use of the AC97 clock here. If
- * CLK_POUT is selected by SW15 then the clock API will need
- * to be used to request and enable it here.
- */
+ if (clk_pout)
+ snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
ARRAY_SIZE(zylonite_dapm_widgets));
@@ -86,40 +96,35 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int pll_out = 0;
- unsigned int acds = 0;
unsigned int wm9713_div = 0;
int ret = 0;
+ int rate = params_rate(params);
+ int width = snd_pcm_format_physical_width(params_format(params));
- switch (params_rate(params)) {
+ /* Only support ratios that we can generate neatly from the AC97
+ * based master clock - in particular, this excludes 44.1kHz.
+ * In most applications the voice DAC will be used for telephony
+ * data so multiples of 8kHz will be the common case.
+ */
+ switch (rate) {
case 8000:
wm9713_div = 12;
- pll_out = 2048000;
break;
case 16000:
wm9713_div = 6;
- pll_out = 4096000;
break;
case 48000:
- default:
wm9713_div = 2;
- pll_out = 12288000;
- acds = 1;
break;
+ default:
+ /* Don't support OSS emulation */
+ return -EINVAL;
}
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- 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_CBS_CFS);
- if (ret < 0)
- return ret;
+ /* Add 1 to the width for the leading clock cycle */
+ pll_out = rate * (width + 1) * 8;
- ret = snd_soc_dai_set_tdm_slot(cpu_dai,
- params_channels(params),
- params_channels(params));
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
if (ret < 0)
return ret;
@@ -127,19 +132,22 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+ if (clk_pout)
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
+ WM9713_PCMDIV(wm9713_div));
+ else
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+ WM9713_PCMDIV(wm9713_div));
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
- /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
- * to be set instead.
- */
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
- WM9713_PCMDIV(wm9713_div));
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
@@ -173,8 +181,72 @@ static struct snd_soc_dai_link zylonite_dai[] = {
},
};
+static int zylonite_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ if (clk_pout) {
+ pout = clk_get(NULL, "CLK_POUT");
+ if (IS_ERR(pout)) {
+ dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n",
+ PTR_ERR(pout));
+ return PTR_ERR(pout);
+ }
+
+ ret = clk_enable(pout);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+ ret);
+ clk_put(pout);
+ return ret;
+ }
+
+ dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n",
+ clk_get_rate(pout));
+ }
+
+ return 0;
+}
+
+static int zylonite_remove(struct platform_device *pdev)
+{
+ if (clk_pout) {
+ clk_disable(pout);
+ clk_put(pout);
+ }
+
+ return 0;
+}
+
+static int zylonite_suspend_post(struct platform_device *pdev,
+ pm_message_t state)
+{
+ if (clk_pout)
+ clk_disable(pout);
+
+ return 0;
+}
+
+static int zylonite_resume_pre(struct platform_device *pdev)
+{
+ int ret = 0;
+
+ if (clk_pout) {
+ ret = clk_enable(pout);
+ if (ret != 0)
+ dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+
static struct snd_soc_card zylonite = {
.name = "Zylonite",
+ .probe = &zylonite_probe,
+ .remove = &zylonite_remove,
+ .suspend_post = &zylonite_suspend_post,
+ .resume_pre = &zylonite_resume_pre,
.platform = &pxa2xx_soc_platform,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-19 22:07:48 +0000