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-rw-r--r--sound/soc/pxa/Kconfig11
-rw-r--r--sound/soc/pxa/Makefile3
-rw-r--r--sound/soc/pxa/corgi.c70
-rw-r--r--sound/soc/pxa/em-x270.c102
-rw-r--r--sound/soc/pxa/poodle.c50
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c18
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.h2
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c29
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.h2
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c2
-rw-r--r--sound/soc/pxa/spitz.c91
-rw-r--r--sound/soc/pxa/tosa.c47
12 files changed, 261 insertions, 166 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 484f883459e0..12f6ac99b04c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,6 +1,6 @@
config SND_PXA2XX_SOC
tristate "SoC Audio for the Intel PXA2xx chip"
- depends on ARCH_PXA && SND_SOC
+ depends on ARCH_PXA
help
Say Y or M if you want to add support for codecs attached to
the PXA2xx AC97, I2S or SSP interface. You will also need
@@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800
help
Say Y if you want to add support for SoC audio on the
Toshiba e800 PDA
+
+config SND_PXA2XX_SOC_EM_X270
+ tristate "SoC Audio support for CompuLab EM-x270"
+ depends on SND_PXA2XX_SOC && MACH_EM_X270
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on
+ CompuLab EM-x270.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 04e5646f75ba..5bc8edf9dca9 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o
snd-soc-tosa-objs := tosa.o
snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
+snd-soc-em-x270-objs := em-x270.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_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
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 7f32a1167572..c0294464a23a 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -11,10 +11,6 @@
* 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.
- *
- * Revision history
- * 30th Nov 2005 Initial version.
- *
*/
#include <linux/module.h>
@@ -54,47 +50,51 @@ static int corgi_spk_func;
static void corgi_ext_control(struct snd_soc_codec *codec)
{
- int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
-
/* set up jack connection */
switch (corgi_jack_func) {
case CORGI_HP:
- hp = 1;
/* set = unmute headphone */
set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_MIC:
- mic = 1;
/* reset = mute headphone */
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_LINE:
- line = 1;
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_HEADSET:
- hs = 1;
- mic = 1;
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
break;
}
if (corgi_spk_func == CORGI_SPK_ON)
- spk = 1;
-
- /* set the enpoints to their new connetion states */
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", line);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int corgi_startup(struct snd_pcm_substream *substream)
@@ -123,8 +123,8 @@ static int corgi_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;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int clk = 0;
int ret = 0;
@@ -143,25 +143,25 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
}
/* set codec DAI configuration */
- ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ 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;
/* set cpu DAI configuration */
- ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ 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;
/* set the codec system clock for DAC and ADC */
- ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set the I2S system clock as input (unused) */
- ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
@@ -247,7 +247,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL),
};
/* Corgi machine audio map (connections to the codec pins) */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* headset Jack - in = micin, out = LHPOUT*/
{"Headset Jack", NULL, "LHPOUT"},
@@ -265,8 +265,6 @@ static const char *audio_map[][3] = {
/* Same as the above but no mic bias for line signals */
{"MICIN", NULL, "Line Jack"},
-
- {NULL, NULL, NULL},
};
static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -291,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
{
int i, err;
- snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+ snd_soc_dapm_disable_pin(codec, "LLINEIN");
+ snd_soc_dapm_disable_pin(codec, "RLINEIN");
/* Add corgi specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
@@ -303,15 +301,13 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
}
/* Add corgi specific widgets */
- for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
/* Set up corgi specific audio path audio_map */
- 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_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
new file mode 100644
index 000000000000..02dcac39cdf6
--- /dev/null
+++ b/sound/soc/pxa/em-x270.c
@@ -0,0 +1,102 @@
+/*
+ * em-x270.c -- SoC audio for EM-X270
+ *
+ * Copyright 2007 CompuLab, Ltd.
+ *
+ * Author: Mike Rapoport <mike@compulab.co.il>
+ *
+ * Copied from tosa.c:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Richard Purdie <richard@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/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_dai_link em_x270_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ },
+ {
+ .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_machine em_x270 = {
+ .name = "EM-X270",
+ .dai_link = em_x270_dai,
+ .num_links = ARRAY_SIZE(em_x270_dai),
+};
+
+static struct snd_soc_device em_x270_snd_devdata = {
+ .machine = &em_x270,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *em_x270_snd_device;
+
+static int __init em_x270_init(void)
+{
+ int ret;
+
+ if (!machine_is_em_x270())
+ return -ENODEV;
+
+ em_x270_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!em_x270_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata);
+ em_x270_snd_devdata.dev = &em_x270_snd_device->dev;
+ ret = platform_device_add(em_x270_snd_device);
+
+ if (ret)
+ platform_device_put(em_x270_snd_device);
+
+ return ret;
+}
+
+static void __exit em_x270_exit(void)
+{
+ platform_device_unregister(em_x270_snd_device);
+}
+
+module_init(em_x270_init);
+module_exit(em_x270_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mike Rapoport");
+MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 7e830b218943..65a4e9a8c39e 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -48,8 +48,6 @@ static int poodle_spk_func;
static void poodle_ext_control(struct snd_soc_codec *codec)
{
- int spk = 0;
-
/* set up jack connection */
if (poodle_jack_func == POODLE_HP) {
/* set = unmute headphone */
@@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
POODLE_LOCOMO_GPIO_MUTE_L, 1);
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_R, 1);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
} else {
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_L, 0);
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_R, 0);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
}
- if (poodle_spk_func == POODLE_SPK_ON)
- spk = 1;
-
/* set the enpoints to their new connetion states */
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+ if (poodle_spk_func == POODLE_SPK_ON)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int poodle_startup(struct snd_pcm_substream *substream)
@@ -104,8 +102,8 @@ static int poodle_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;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int clk = 0;
int ret = 0;
@@ -124,25 +122,25 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
}
/* set codec DAI configuration */
- ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ 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;
/* set cpu DAI configuration */
- ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ 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;
/* set the codec system clock for DAC and ADC */
- ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set the I2S system clock as input (unused) */
- ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
@@ -215,8 +213,8 @@ SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
};
-/* Corgi machine audio_mapnections to the codec pins */
-static const char *audio_map[][3] = {
+/* Corgi machine connections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to LHPOUT1, RHPOUT1 */
{"Headphone Jack", NULL, "LHPOUT"},
@@ -225,8 +223,6 @@ static const char *audio_map[][3] = {
/* speaker connected to LOUT, ROUT */
{"Ext Spk", NULL, "ROUT"},
{"Ext Spk", NULL, "LOUT"},
-
- {NULL, NULL, NULL},
};
static const char *jack_function[] = {"Off", "Headphone"};
@@ -250,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
{
int i, err;
- snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "MICIN", 1);
+ snd_soc_dapm_disable_pin(codec, "LLINEIN");
+ snd_soc_dapm_disable_pin(codec, "RLINEIN");
+ snd_soc_dapm_enable_pin(codec, "MICIN");
/* Add poodle specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
@@ -263,15 +259,13 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
}
/* Add poodle specific widgets */
- for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
/* Set up poodle specific audio path audio_map */
- 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_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 97ec2d90547c..059af815ea0c 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -283,7 +283,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
#ifdef CONFIG_PM
static int pxa2xx_ac97_suspend(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
GCR |= GCR_ACLINK_OFF;
clk_disable(ac97_clk);
@@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev,
}
static int pxa2xx_ac97_resume(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
@@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev,
#define pxa2xx_ac97_resume NULL
#endif
-static int pxa2xx_ac97_probe(struct platform_device *pdev)
+static int pxa2xx_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
{
int ret;
@@ -355,7 +356,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev)
return ret;
}
-static void pxa2xx_ac97_remove(struct platform_device *pdev)
+static void pxa2xx_ac97_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
{
GCR |= GCR_ACLINK_OFF;
free_irq(IRQ_AC97, NULL);
@@ -372,7 +374,7 @@ static int pxa2xx_ac97_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
@@ -386,7 +388,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
@@ -400,7 +402,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
@@ -418,7 +420,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
* There is only 1 physical AC97 interface for pxa2xx, but it
* has extra fifo's that can be used for aux DACs and ADCs.
*/
-struct snd_soc_cpu_dai pxa_ac97_dai[] = {
+struct snd_soc_dai pxa_ac97_dai[] = {
{
.name = "pxa2xx-ac97",
.id = 0,
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
index b8ccfee095c4..e390de8edcd4 100644
--- a/sound/soc/pxa/pxa2xx-ac97.h
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -14,7 +14,7 @@
#define PXA2XX_DAI_AC97_AUX 1
#define PXA2XX_DAI_AC97_MIC 2
-extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
+extern struct snd_soc_dai pxa_ac97_dai[3];
/* platform data */
extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 425071030970..8f96d87f7b4b 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -9,15 +9,13 @@
* 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.
- *
- * Revision history
- * 12th Aug 2005 Initial version.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/delay.h>
+#include <linux/clk.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/initval.h>
@@ -40,6 +38,7 @@ struct pxa_i2s_port {
u32 fmt;
};
static struct pxa_i2s_port pxa_i2s;
+static struct clk *clk_i2s;
static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
.name = "I2S PCM Stereo out",
@@ -80,7 +79,11 @@ static struct pxa2xx_gpio gpio_bus[] = {
static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ clk_i2s = clk_get(NULL, "I2SCLK");
+ if (IS_ERR(clk_i2s))
+ return PTR_ERR(clk_i2s);
if (!cpu_dai->active) {
SACR0 |= SACR0_RST;
@@ -101,7 +104,7 @@ static int pxa_i2s_wait(void)
return 0;
}
-static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
/* interface format */
@@ -127,7 +130,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
return 0;
}
-static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
if (clk_id != PXA2XX_I2S_SYSCLK)
@@ -143,13 +146,13 @@ static int pxa2xx_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ 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);
- pxa_set_cken(CKEN_I2S, 1);
+ clk_enable(clk_i2s);
pxa_i2s_wait();
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -234,13 +237,15 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
SACR0 &= ~SACR0_ENB;
pxa_i2s_wait();
- pxa_set_cken(CKEN_I2S, 0);
+ clk_disable(clk_i2s);
}
+
+ clk_put(clk_i2s);
}
#ifdef CONFIG_PM
static int pxa2xx_i2s_suspend(struct platform_device *dev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
if (!dai->active)
return 0;
@@ -258,7 +263,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
}
static int pxa2xx_i2s_resume(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
if (!dai->active)
return 0;
@@ -283,7 +288,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
-struct snd_soc_cpu_dai pxa_i2s_dai = {
+struct snd_soc_dai pxa_i2s_dai = {
.name = "pxa2xx-i2s",
.id = 0,
.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
index 4435bd9f884f..e2def441153e 100644
--- a/sound/soc/pxa/pxa2xx-i2s.h
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -15,6 +15,6 @@
/* I2S clock */
#define PXA2XX_I2S_SYSCLK 0
-extern struct snd_soc_cpu_dai pxa_i2s_dai;
+extern struct snd_soc_dai pxa_i2s_dai;
#endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 01ad7bf716b7..2df03ee5819e 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
-int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d8b8372db00e..64385797da5d 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -12,9 +12,6 @@
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
- * Revision history
- * 30th Nov 2005 Initial version.
- *
*/
#include <linux/module.h>
@@ -54,60 +51,60 @@ static int spitz_spk_func;
static void spitz_ext_control(struct snd_soc_codec *codec)
{
if (spitz_spk_func == SPITZ_SPK_ON)
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
else
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0);
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
}
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int spitz_startup(struct snd_pcm_substream *substream)
@@ -124,8 +121,8 @@ static int spitz_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;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int clk = 0;
int ret = 0;
@@ -144,25 +141,25 @@ static int spitz_hw_params(struct snd_pcm_substream *substream,
}
/* set codec DAI configuration */
- ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ 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;
/* set cpu DAI configuration */
- ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ 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;
/* set the codec system clock for DAC and ADC */
- ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set the I2S system clock as input (unused) */
- ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
@@ -250,7 +247,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
};
/* Spitz machine audio_map */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to LOUT1, ROUT1 */
{"Headphone Jack", NULL, "LOUT1"},
@@ -269,8 +266,6 @@ static const char *audio_map[][3] = {
/* line is connected to input 1 - no bias */
{"LINPUT1", NULL, "Line Jack"},
-
- {NULL, NULL, NULL},
};
static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -296,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
int i, err;
/* NC codec pins */
- snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0);
- snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0);
- snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0);
- snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "MONO", 0);
+ snd_soc_dapm_disable_pin(codec, "RINPUT1");
+ snd_soc_dapm_disable_pin(codec, "LINPUT2");
+ snd_soc_dapm_disable_pin(codec, "RINPUT2");
+ snd_soc_dapm_disable_pin(codec, "LINPUT3");
+ snd_soc_dapm_disable_pin(codec, "RINPUT3");
+ snd_soc_dapm_disable_pin(codec, "OUT3");
+ snd_soc_dapm_disable_pin(codec, "MONO");
/* Add spitz specific controls */
for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
@@ -313,15 +308,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
}
/* Add spitz specific widgets */
- for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+ snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ ARRAY_SIZE(wm8750_dapm_widgets));
- /* Set up spitz specific audio path audio_map */
- 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]);
+ /* Set up spitz specific audio paths */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 7346d7e5d066..b6edb61a3a30 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -12,9 +12,6 @@
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
- * Revision history
- * 30th Nov 2005 Initial version.
- *
* GPIO's
* 1 - Jack Insertion
* 5 - Hookswitch (headset answer/hang up switch)
@@ -55,29 +52,31 @@ static int tosa_spk_func;
static void tosa_ext_control(struct snd_soc_codec *codec)
{
- int spk = 0, mic_int = 0, hp = 0, hs = 0;
-
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
- hp = 1;
+ snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case TOSA_MIC_INT:
- mic_int = 1;
+ snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case TOSA_HEADSET:
- hs = 1;
+ snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
- spk = 1;
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
- snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
- snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int tosa_startup(struct snd_pcm_substream *substream)
@@ -154,7 +153,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL),
};
/* tosa audio map */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to HPOUTL, HPOUTR */
{"Headphone Jack", NULL, "HPOUTL"},
@@ -173,8 +172,6 @@ static const char *audio_map[][3] = {
{"Headset Jack", NULL, "HPOUTR"},
{"LINEINR", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Headset Jack"},
-
- {NULL, NULL, NULL},
};
static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -196,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
{
int i, err;
- snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
+ snd_soc_dapm_disable_pin(codec, "OUT3");
+ snd_soc_dapm_disable_pin(codec, "MONOOUT");
/* add tosa specific controls */
for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
@@ -208,17 +205,13 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
}
/* add tosa specific widgets */
- for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
- snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
- }
+ snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
+ ARRAY_SIZE(tosa_dapm_widgets));
/* set up tosa specific audio path audio_map */
- 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_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
generated by cgit v1.2.3 (git 2.0.4) at 2024-11-16 08:52:50 +0000