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未验证 提交 05f4434b 编写于 作者: V Vinod Koul 提交者： Mark Brown
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ASoC: Intel: remove mfld_machine 

mfld_machine was not getting compiled due to missed Makefile changes.
Since no one complained it is safe to assume that it is not being used,
so remove it
Signed-off-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>
上级 8fca1583
隐藏空白更改
内联 并排
Showing with 0 addition and 444 deletion
sound/soc/intel/boards/Kconfig
浏览文件 @ 05f4434b
......@@ -12,20 +12,6 @@ menuconfig SND_SOC_INTEL_MACH
if SND_SOC_INTEL_MACH
if SND_SST_ATOM_HIFI2_PLATFORM_PCI
config SND_MFLD_MACHINE
tristate "Medfield (Intel MID)"
depends on INTEL_SCU_IPC
select SND_SOC_SN95031
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
Say Y if you have such a device.
If unsure select "N".
endif ## SND_SST_ATOM_HIFI2_PLATFORM_PCI
if SND_SOC_INTEL_HASWELL
config SND_SOC_INTEL_HASWELL_MACH
......
sound/soc/intel/boards/mfld_machine.c 已删除 100644 → 0
浏览文件 @ 8fca1583
/*
* mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
*
* Copyright (C) 2010 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.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 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.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
#include <linux/init.h>
#include <linux/device.h>
#include <linux/slab.h>
#include <linux/io.h>
#include <linux/module.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "../codecs/sn95031.h"
#define MID_MONO 1
#define MID_STEREO 2
#define MID_MAX_CAP 5
#define MFLD_JACK_INSERT 0x04
enum soc_mic_bias_zones {
MFLD_MV_START = 0,
/* mic bias volutage range for Headphones*/
MFLD_MV_HP = 400,
/* mic bias volutage range for American Headset*/
MFLD_MV_AM_HS = 650,
/* mic bias volutage range for Headset*/
MFLD_MV_HS = 2000,
MFLD_MV_UNDEFINED,
};
static unsigned int hs_switch;
static unsigned int lo_dac;
static struct snd_soc_codec *mfld_codec;
struct mfld_mc_private {
void __iomem *int_base;
u8 interrupt_status;
};
struct snd_soc_jack mfld_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin mfld_jack_pins[] = {
{
.pin = "Headphones",
.mask = SND_JACK_HEADPHONE,
},
{
.pin = "AMIC1",
.mask = SND_JACK_MICROPHONE,
},
};
/* jack detection voltage zones */
static struct snd_soc_jack_zone mfld_zones[] = {
{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
};
/* sound card controls */
static const char * const headset_switch_text[] = {"Earpiece", "Headset"};
static const char * const lo_text[] = {"Vibra", "Headset", "IHF", "None"};
static const struct soc_enum headset_enum =
SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
static const struct soc_enum lo_enum =
SOC_ENUM_SINGLE_EXT(4, lo_text);
static int headset_get_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = hs_switch;
return 0;
}
static int headset_set_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_context *dapm = &card->dapm;
if (ucontrol->value.enumerated.item[0] == hs_switch)
return 0;
snd_soc_dapm_mutex_lock(dapm);
if (ucontrol->value.enumerated.item[0]) {
pr_debug("hs_set HS path\n");
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
pr_debug("hs_set EP path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
hs_switch = ucontrol->value.enumerated.item[0];
return 0;
}
static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm)
{
snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
if (hs_switch) {
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
}
static int lo_get_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = lo_dac;
return 0;
}
static int lo_set_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_context *dapm = &card->dapm;
if (ucontrol->value.enumerated.item[0] == lo_dac)
return 0;
snd_soc_dapm_mutex_lock(dapm);
/* we dont want to work with last state of lineout so just enable all
* pins and then disable pins not required
*/
lo_enable_out_pins(dapm);
switch (ucontrol->value.enumerated.item[0]) {
case 0:
pr_debug("set vibra path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0);
break;
case 1:
pr_debug("set hs path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22);
break;
case 2:
pr_debug("set spkr path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44);
break;
case 3:
pr_debug("set null path\n");
snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66);
break;
}
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
lo_dac = ucontrol->value.enumerated.item[0];
return 0;
}
static const struct snd_kcontrol_new mfld_snd_controls[] = {
SOC_ENUM_EXT("Playback Switch", headset_enum,
headset_get_switch, headset_set_switch),
SOC_ENUM_EXT("Lineout Mux", lo_enum,
lo_get_switch, lo_set_switch),
};
static const struct snd_soc_dapm_widget mfld_widgets[] = {
SND_SOC_DAPM_HP("Headphones", NULL),
SND_SOC_DAPM_MIC("Mic", NULL),
};
static const struct snd_soc_dapm_route mfld_map[] = {
{"Headphones", NULL, "HPOUTR"},
{"Headphones", NULL, "HPOUTL"},
{"Mic", NULL, "AMIC1"},
};
static void mfld_jack_check(unsigned int intr_status)
{
struct mfld_jack_data jack_data;
if (!mfld_codec)
return;
jack_data.mfld_jack = &mfld_jack;
jack_data.intr_id = intr_status;
sn95031_jack_detection(mfld_codec, &jack_data);
/* TODO: add american headset detection post gpiolib support */
}
static int mfld_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
int ret_val;
/* default is earpiece pin, userspace sets it explcitly */
snd_soc_dapm_disable_pin(dapm, "Headphones");
/* default is lineout NC, userspace sets it explcitly */
snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
lo_dac = 3;
hs_switch = 0;
/* we dont use linein in this so set to NC */
snd_soc_dapm_disable_pin(dapm, "LINEINL");
snd_soc_dapm_disable_pin(dapm, "LINEINR");
/* Headset and button jack detection */
ret_val = snd_soc_card_jack_new(runtime->card,
"Intel(R) MID Audio Jack", SND_JACK_HEADSET |
SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack,
mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins));
if (ret_val) {
pr_err("jack creation failed\n");
return ret_val;
}
ret_val = snd_soc_jack_add_zones(&mfld_jack,
ARRAY_SIZE(mfld_zones), mfld_zones);
if (ret_val) {
pr_err("adding jack zones failed\n");
return ret_val;
}
mfld_codec = runtime->codec;
/* we want to check if anything is inserted at boot,
* so send a fake event to codec and it will read adc
* to find if anything is there or not */
mfld_jack_check(MFLD_JACK_INSERT);
return ret_val;
}
static struct snd_soc_dai_link mfld_msic_dailink[] = {
{
.name = "Medfield Headset",
.stream_name = "Headset",
.cpu_dai_name = "Headset-cpu-dai",
.codec_dai_name = "SN95031 Headset",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = mfld_init,
},
{
.name = "Medfield Speaker",
.stream_name = "Speaker",
.cpu_dai_name = "Speaker-cpu-dai",
.codec_dai_name = "SN95031 Speaker",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Vibra",
.stream_name = "Vibra1",
.cpu_dai_name = "Vibra1-cpu-dai",
.codec_dai_name = "SN95031 Vibra1",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Haptics",
.stream_name = "Vibra2",
.cpu_dai_name = "Vibra2-cpu-dai",
.codec_dai_name = "SN95031 Vibra2",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Compress",
.stream_name = "Speaker",
.cpu_dai_name = "Compress-cpu-dai",
.codec_dai_name = "SN95031 Speaker",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
};
/* SoC card */
static struct snd_soc_card snd_soc_card_mfld = {
.name = "medfield_audio",
.owner = THIS_MODULE,
.dai_link = mfld_msic_dailink,
.num_links = ARRAY_SIZE(mfld_msic_dailink),
.controls = mfld_snd_controls,
.num_controls = ARRAY_SIZE(mfld_snd_controls),
.dapm_widgets = mfld_widgets,
.num_dapm_widgets = ARRAY_SIZE(mfld_widgets),
.dapm_routes = mfld_map,
.num_dapm_routes = ARRAY_SIZE(mfld_map),
};
static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
{
struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
memcpy_fromio(&mc_private->interrupt_status,
((void *)(mc_private->int_base)),
sizeof(u8));
return IRQ_WAKE_THREAD;
}
static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
{
struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
mfld_jack_check(mc_drv_ctx->interrupt_status);
return IRQ_HANDLED;
}
static int snd_mfld_mc_probe(struct platform_device *pdev)
{
int ret_val = 0, irq;
struct mfld_mc_private *mc_drv_ctx;
struct resource *irq_mem;
pr_debug("snd_mfld_mc_probe called\n");
/* retrive the irq number */
irq = platform_get_irq(pdev, 0);
if (irq <= 0)
return irq < 0 ? irq : -ENODEV;
/* audio interrupt base of SRAM location where
* interrupts are stored by System FW */
mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
if (!mc_drv_ctx)
return -ENOMEM;
irq_mem = platform_get_resource_byname(
pdev, IORESOURCE_MEM, "IRQ_BASE");
if (!irq_mem) {
pr_err("no mem resource given\n");
return -ENODEV;
}
mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
resource_size(irq_mem));
if (!mc_drv_ctx->int_base) {
pr_err("Mapping of cache failed\n");
return -ENOMEM;
}
/* register for interrupt */
ret_val = devm_request_threaded_irq(&pdev->dev, irq,
snd_mfld_jack_intr_handler,
snd_mfld_jack_detection,
IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
if (ret_val) {
pr_err("cannot register IRQ\n");
return ret_val;
}
/* register the soc card */
snd_soc_card_mfld.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
if (ret_val) {
pr_debug("snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, mc_drv_ctx);
pr_debug("successfully exited probe\n");
return 0;
}
static struct platform_driver snd_mfld_mc_driver = {
.driver = {
.name = "msic_audio",
},
.probe = snd_mfld_mc_probe,
};
module_platform_driver(snd_mfld_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:msic-audio");
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