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-rw-r--r--ANDROID_3.4.5/sound/soc/omap/Kconfig152
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/Makefile44
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/am3517evm.c161
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/ams-delta.c630
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/igep0020.c120
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/mcbsp.c1040
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/mcbsp.h346
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/n810.c384
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-abe-twl6040.c349
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-dmic.c545
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-dmic.h69
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-hdmi.c148
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-hdmi.h36
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.c817
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.h64
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.c524
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.h107
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-pcm.c443
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap-pcm.h40
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap3beagle.c150
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap3evm.c118
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap3pandora.c325
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/omap4-hdmi-card.c121
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/osk5912.c189
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/overo.c122
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/rx51.c451
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/sdp3430.c279
-rw-r--r--ANDROID_3.4.5/sound/soc/omap/zoom2.c219
28 files changed, 0 insertions, 7993 deletions
diff --git a/ANDROID_3.4.5/sound/soc/omap/Kconfig b/ANDROID_3.4.5/sound/soc/omap/Kconfig
deleted file mode 100644
index deafbfaa..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/Kconfig
+++ /dev/null
@@ -1,152 +0,0 @@
-config SND_OMAP_SOC
- tristate "SoC Audio for the Texas Instruments OMAP chips"
- depends on ARCH_OMAP
-
-config SND_OMAP_SOC_DMIC
- tristate
-
-config SND_OMAP_SOC_MCBSP
- tristate
-
-config SND_OMAP_SOC_MCPDM
- tristate
-
-config SND_OMAP_SOC_HDMI
- tristate
-
-config SND_OMAP_SOC_N810
- tristate "SoC Audio support for Nokia N810"
- depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
- depends on OMAP_MUX
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC3X
- help
- Say Y if you want to add support for SoC audio on Nokia N810.
-
-config SND_OMAP_SOC_RX51
- tristate "SoC Audio support for Nokia RX-51"
- depends on SND_OMAP_SOC && MACH_NOKIA_RX51
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC3X
- select SND_SOC_TPA6130A2
- help
- Say Y if you want to add support for SoC audio on Nokia RX-51
- hardware. This is also known as Nokia N900 product.
-
-config SND_OMAP_SOC_AMS_DELTA
- tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
- depends on SND_OMAP_SOC && MACH_AMS_DELTA
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_CX20442
- help
- Say Y if you want to add support for SoC audio device connected to
- a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
-
- Note that in order to get those devices fully supported, you have to
- build the kernel with standard serial port driver included and
- configured for at least 4 ports. Then, from userspace, you must load
- a line discipline #19 on the modem (ttyS3) serial line. The simplest
- way to achieve this is to install util-linux-ng and use the included
- ldattach utility. This can be started automatically from udev,
- a simple rule like this one should do the trick (it does for me):
- ACTION=="add", KERNEL=="controlC0", \
- RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
-
-config SND_OMAP_SOC_OSK5912
- tristate "SoC Audio support for omap osk5912"
- depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
- help
- Say Y if you want to add support for SoC audio on osk5912.
-
-config SND_OMAP_SOC_OVERO
- tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35"
- depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35)
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for SoC audio on the
- Gumstix Overo or CompuLab CM-T35
-
-config SND_OMAP_SOC_OMAP3EVM
- tristate "SoC Audio support for OMAP3EVM board"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for SoC audio on the omap3evm board.
-
-config SND_OMAP_SOC_AM3517EVM
- tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
- depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
- help
- Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
- EVM.
-
-config SND_OMAP_SOC_SDP3430
- tristate "SoC Audio support for Texas Instruments SDP3430"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for SoC audio on Texas Instruments
- SDP3430.
-
-config SND_OMAP_SOC_OMAP_ABE_TWL6040
- tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
- depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4
- select SND_OMAP_SOC_DMIC
- select SND_OMAP_SOC_MCPDM
- select SND_SOC_TWL6040
- select SND_SOC_DMIC
- help
- Say Y if you want to add support for SoC audio on OMAP boards using
- ABE and twl6040 codec. This driver currently supports:
- - SDP4430/Blaze boards
- - PandaBoard (4430)
- - PandaBoardES (4460)
-
-config SND_OMAP_SOC_OMAP4_HDMI
- tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
- depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4
- select SND_OMAP_SOC_HDMI
- help
- Say Y if you want to add support for SoC HDMI audio on Texas Instruments
- OMAP4 chips
-
-config SND_OMAP_SOC_OMAP3_PANDORA
- tristate "SoC Audio support for OMAP3 Pandora"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
-
-config SND_OMAP_SOC_OMAP3_BEAGLE
- tristate "SoC Audio support for OMAP3 Beagle and Devkit8000"
- depends on TWL4030_CORE && SND_OMAP_SOC
- depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000)
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for SoC audio on the Beagleboard or
- the clone Devkit8000.
-
-config SND_OMAP_SOC_ZOOM2
- tristate "SoC Audio support for Zoom2"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for Soc audio on Zoom2 board.
-
-config SND_OMAP_SOC_IGEP0020
- tristate "SoC Audio support for IGEP v2"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for Soc audio on IGEP v2 board.
diff --git a/ANDROID_3.4.5/sound/soc/omap/Makefile b/ANDROID_3.4.5/sound/soc/omap/Makefile
deleted file mode 100644
index 1d656bce..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/Makefile
+++ /dev/null
@@ -1,44 +0,0 @@
-# OMAP Platform Support
-snd-soc-omap-objs := omap-pcm.o
-snd-soc-omap-dmic-objs := omap-dmic.o
-snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o
-snd-soc-omap-mcpdm-objs := omap-mcpdm.o
-snd-soc-omap-hdmi-objs := omap-hdmi.o
-
-obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
-obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o
-obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
-obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
-obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o
-
-# OMAP Machine Support
-snd-soc-n810-objs := n810.o
-snd-soc-rx51-objs := rx51.o
-snd-soc-ams-delta-objs := ams-delta.o
-snd-soc-osk5912-objs := osk5912.o
-snd-soc-overo-objs := overo.o
-snd-soc-omap3evm-objs := omap3evm.o
-snd-soc-am3517evm-objs := am3517evm.o
-snd-soc-sdp3430-objs := sdp3430.o
-snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
-snd-soc-omap3pandora-objs := omap3pandora.o
-snd-soc-omap3beagle-objs := omap3beagle.o
-snd-soc-zoom2-objs := zoom2.o
-snd-soc-igep0020-objs := igep0020.o
-snd-soc-omap4-hdmi-objs := omap4-hdmi-card.o
-
-obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
-obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
-obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
-obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
-obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
-obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
-obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
-obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP4_HDMI) += snd-soc-omap4-hdmi.o
diff --git a/ANDROID_3.4.5/sound/soc/omap/am3517evm.c b/ANDROID_3.4.5/sound/soc/omap/am3517evm.c
deleted file mode 100644
index 009533ab..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/am3517evm.c
+++ /dev/null
@@ -1,161 +0,0 @@
-/*
- * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM
- *
- * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
- *
- * Based on sound/soc/omap/beagle.c by Steve Sakoman
- *
- * Copyright (C) 2009 Texas Instruments Incorporated
- *
- * 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.
- *
- * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
- * whether express or implied; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#include "../codecs/tlv320aic23.h"
-
-#define CODEC_CLOCK 12000000
-
-static int am3517evm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0,
- CODEC_CLOCK, SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n");
- return ret;
- }
-
- snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops am3517evm_ops = {
- .hw_params = am3517evm_hw_params,
-};
-
-/* am3517evm machine dapm widgets */
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Line Out", NULL),
- SND_SOC_DAPM_LINE("Line In", NULL),
- SND_SOC_DAPM_MIC("Mic In", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- /* Line Out connected to LLOUT, RLOUT */
- {"Line Out", NULL, "LOUT"},
- {"Line Out", NULL, "ROUT"},
-
- {"LLINEIN", NULL, "Line In"},
- {"RLINEIN", NULL, "Line In"},
-
- {"MICIN", NULL, "Mic In"},
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link am3517evm_dai = {
- .name = "TLV320AIC23",
- .stream_name = "AIC23",
- .cpu_dai_name = "omap-mcbsp.1",
- .codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "tlv320aic23-codec.2-001a",
- .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &am3517evm_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_am3517evm = {
- .name = "am3517evm",
- .owner = THIS_MODULE,
- .dai_link = &am3517evm_dai,
- .num_links = 1,
-
- .dapm_widgets = tlv320aic23_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *am3517evm_snd_device;
-
-static int __init am3517evm_soc_init(void)
-{
- int ret;
-
- if (!machine_is_omap3517evm())
- return -ENODEV;
- pr_info("OMAP3517 / AM3517 EVM SoC init\n");
-
- am3517evm_snd_device = platform_device_alloc("soc-audio", -1);
- if (!am3517evm_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(am3517evm_snd_device, &snd_soc_am3517evm);
-
- ret = platform_device_add(am3517evm_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(am3517evm_snd_device);
-
- return ret;
-}
-
-static void __exit am3517evm_soc_exit(void)
-{
- platform_device_unregister(am3517evm_snd_device);
-}
-
-module_init(am3517evm_soc_init);
-module_exit(am3517evm_soc_exit);
-
-MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM");
-MODULE_LICENSE("GPL v2");
diff --git a/ANDROID_3.4.5/sound/soc/omap/ams-delta.c b/ANDROID_3.4.5/sound/soc/omap/ams-delta.c
deleted file mode 100644
index 7d4fa8ed..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/ams-delta.c
+++ /dev/null
@@ -1,630 +0,0 @@
-/*
- * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
- *
- * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
- *
- * Initially based on sound/soc/omap/osk5912.x
- * Copyright (C) 2008 Mistral Solutions
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/gpio.h>
-#include <linux/spinlock.h>
-#include <linux/tty.h>
-#include <linux/module.h>
-
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-
-#include <plat/board-ams-delta.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-#include "../codecs/cx20442.h"
-
-
-/* Board specific DAPM widgets */
-static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
- /* Handset */
- SND_SOC_DAPM_MIC("Mouthpiece", NULL),
- SND_SOC_DAPM_HP("Earpiece", NULL),
- /* Handsfree/Speakerphone */
- SND_SOC_DAPM_MIC("Microphone", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
-/* How they are connected to codec pins */
-static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
- {"TELIN", NULL, "Mouthpiece"},
- {"Earpiece", NULL, "TELOUT"},
-
- {"MIC", NULL, "Microphone"},
- {"Speaker", NULL, "SPKOUT"},
-};
-
-/*
- * Controls, functional after the modem line discipline is activated.
- */
-
-/* Virtual switch: audio input/output constellations */
-static const char *ams_delta_audio_mode[] =
- {"Mixed", "Handset", "Handsfree", "Speakerphone"};
-
-/* Selection <-> pin translation */
-#define AMS_DELTA_MOUTHPIECE 0
-#define AMS_DELTA_EARPIECE 1
-#define AMS_DELTA_MICROPHONE 2
-#define AMS_DELTA_SPEAKER 3
-#define AMS_DELTA_AGC 4
-
-#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \
- (1 << AMS_DELTA_MICROPHONE))
-#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \
- (1 << AMS_DELTA_EARPIECE))
-#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \
- (1 << AMS_DELTA_SPEAKER))
-#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
-
-static const unsigned short ams_delta_audio_mode_pins[] = {
- AMS_DELTA_MIXED,
- AMS_DELTA_HANDSET,
- AMS_DELTA_HANDSFREE,
- AMS_DELTA_SPEAKERPHONE,
-};
-
-static unsigned short ams_delta_audio_agc;
-
-static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
- unsigned short pins;
- int pin, changed = 0;
-
- /* Refuse any mode changes if we are not able to control the codec. */
- if (!codec->hw_write)
- return -EUNATCH;
-
- if (ucontrol->value.enumerated.item[0] >= control->max)
- return -EINVAL;
-
- mutex_lock(&codec->mutex);
-
- /* Translate selection to bitmap */
- pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
-
- /* Setup pins after corresponding bits if changed */
- pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
- if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
- changed = 1;
- if (pin)
- snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
- else
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
- }
- pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
- if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
- changed = 1;
- if (pin)
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- else
- snd_soc_dapm_disable_pin(dapm, "Earpiece");
- }
- pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
- if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
- changed = 1;
- if (pin)
- snd_soc_dapm_enable_pin(dapm, "Microphone");
- else
- snd_soc_dapm_disable_pin(dapm, "Microphone");
- }
- pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
- if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
- changed = 1;
- if (pin)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
- else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
- }
- pin = !!(pins & (1 << AMS_DELTA_AGC));
- if (pin != ams_delta_audio_agc) {
- ams_delta_audio_agc = pin;
- changed = 1;
- if (pin)
- snd_soc_dapm_enable_pin(dapm, "AGCIN");
- else
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
- }
- if (changed)
- snd_soc_dapm_sync(dapm);
-
- mutex_unlock(&codec->mutex);
-
- return changed;
-}
-
-static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- unsigned short pins, mode;
-
- pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
- AMS_DELTA_MOUTHPIECE) |
- (snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
- AMS_DELTA_EARPIECE));
- if (pins)
- pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
- AMS_DELTA_MICROPHONE);
- else
- pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
- AMS_DELTA_MICROPHONE) |
- (snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
- AMS_DELTA_SPEAKER) |
- (ams_delta_audio_agc << AMS_DELTA_AGC));
-
- for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
- if (pins == ams_delta_audio_mode_pins[mode])
- break;
-
- if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
- return -EINVAL;
-
- ucontrol->value.enumerated.item[0] = mode;
-
- return 0;
-}
-
-static const struct soc_enum ams_delta_audio_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
- ams_delta_audio_mode),
-};
-
-static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
- SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
- ams_delta_get_audio_mode, ams_delta_set_audio_mode),
-};
-
-/* Hook switch */
-static struct snd_soc_jack ams_delta_hook_switch;
-static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
- {
- .gpio = 4,
- .name = "hook_switch",
- .report = SND_JACK_HEADSET,
- .invert = 1,
- .debounce_time = 150,
- }
-};
-
-/* After we are able to control the codec over the modem,
- * the hook switch can be used for dynamic DAPM reconfiguration. */
-static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
- /* Handset */
- {
- .pin = "Mouthpiece",
- .mask = SND_JACK_MICROPHONE,
- },
- {
- .pin = "Earpiece",
- .mask = SND_JACK_HEADPHONE,
- },
- /* Handsfree */
- {
- .pin = "Microphone",
- .mask = SND_JACK_MICROPHONE,
- .invert = 1,
- },
- {
- .pin = "Speaker",
- .mask = SND_JACK_HEADPHONE,
- .invert = 1,
- },
-};
-
-
-/*
- * Modem line discipline, required for making above controls functional.
- * Activated from userspace with ldattach, possibly invoked from udev rule.
- */
-
-/* To actually apply any modem controlled configuration changes to the codec,
- * we must connect codec DAI pins to the modem for a moment. Be careful not
- * to interfere with our digital mute function that shares the same hardware. */
-static struct timer_list cx81801_timer;
-static bool cx81801_cmd_pending;
-static bool ams_delta_muted;
-static DEFINE_SPINLOCK(ams_delta_lock);
-
-static void cx81801_timeout(unsigned long data)
-{
- int muted;
-
- spin_lock(&ams_delta_lock);
- cx81801_cmd_pending = 0;
- muted = ams_delta_muted;
- spin_unlock(&ams_delta_lock);
-
- /* Reconnect the codec DAI back from the modem to the CPU DAI
- * only if digital mute still off */
- if (!muted)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
-}
-
-/*
- * Used for passing a codec structure pointer
- * from the board initialization code to the tty line discipline.
- */
-static struct snd_soc_codec *cx20442_codec;
-
-/* Line discipline .open() */
-static int cx81801_open(struct tty_struct *tty)
-{
- int ret;
-
- if (!cx20442_codec)
- return -ENODEV;
-
- /*
- * Pass the codec structure pointer for use by other ldisc callbacks,
- * both the card and the codec specific parts.
- */
- tty->disc_data = cx20442_codec;
-
- ret = v253_ops.open(tty);
-
- if (ret < 0)
- tty->disc_data = NULL;
-
- return ret;
-}
-
-/* Line discipline .close() */
-static void cx81801_close(struct tty_struct *tty)
-{
- struct snd_soc_codec *codec = tty->disc_data;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- del_timer_sync(&cx81801_timer);
-
- /* Prevent the hook switch from further changing the DAPM pins */
- INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
-
- if (!codec)
- return;
-
- v253_ops.close(tty);
-
- /* Revert back to default audio input/output constellation */
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- snd_soc_dapm_enable_pin(dapm, "Microphone");
- snd_soc_dapm_disable_pin(dapm, "Speaker");
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
- snd_soc_dapm_sync(dapm);
-}
-
-/* Line discipline .hangup() */
-static int cx81801_hangup(struct tty_struct *tty)
-{
- cx81801_close(tty);
- return 0;
-}
-
-/* Line discipline .receive_buf() */
-static void cx81801_receive(struct tty_struct *tty,
- const unsigned char *cp, char *fp, int count)
-{
- struct snd_soc_codec *codec = tty->disc_data;
- const unsigned char *c;
- int apply, ret;
-
- if (!codec)
- return;
-
- if (!codec->hw_write) {
- /* First modem response, complete setup procedure */
-
- /* Initialize timer used for config pulse generation */
- setup_timer(&cx81801_timer, cx81801_timeout, 0);
-
- v253_ops.receive_buf(tty, cp, fp, count);
-
- /* Link hook switch to DAPM pins */
- ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
- ARRAY_SIZE(ams_delta_hook_switch_pins),
- ams_delta_hook_switch_pins);
- if (ret)
- dev_warn(codec->dev,
- "Failed to link hook switch to DAPM pins, "
- "will continue with hook switch unlinked.\n");
-
- return;
- }
-
- v253_ops.receive_buf(tty, cp, fp, count);
-
- for (c = &cp[count - 1]; c >= cp; c--) {
- if (*c != '\r')
- continue;
- /* Complete modem response received, apply config to codec */
-
- spin_lock_bh(&ams_delta_lock);
- mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
- apply = !ams_delta_muted && !cx81801_cmd_pending;
- cx81801_cmd_pending = 1;
- spin_unlock_bh(&ams_delta_lock);
-
- /* Apply config pulse by connecting the codec to the modem
- * if not already done */
- if (apply)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
- AMS_DELTA_LATCH2_MODEM_CODEC);
- break;
- }
-}
-
-/* Line discipline .write_wakeup() */
-static void cx81801_wakeup(struct tty_struct *tty)
-{
- v253_ops.write_wakeup(tty);
-}
-
-static struct tty_ldisc_ops cx81801_ops = {
- .magic = TTY_LDISC_MAGIC,
- .name = "cx81801",
- .owner = THIS_MODULE,
- .open = cx81801_open,
- .close = cx81801_close,
- .hangup = cx81801_hangup,
- .receive_buf = cx81801_receive,
- .write_wakeup = cx81801_wakeup,
-};
-
-
-/*
- * Even if not very useful, the sound card can still work without any of the
- * above functonality activated. You can still control its audio input/output
- * constellation and speakerphone gain from userspace by issuing AT commands
- * over the modem port.
- */
-
-static int ams_delta_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
-
- /* Set cpu DAI configuration */
- return snd_soc_dai_set_fmt(rtd->cpu_dai,
- SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
-}
-
-static struct snd_soc_ops ams_delta_ops = {
- .hw_params = ams_delta_hw_params,
-};
-
-
-/* Digital mute implemented using modem/CPU multiplexer.
- * Shares hardware with codec config pulse generation */
-static bool ams_delta_muted = 1;
-
-static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
-{
- int apply;
-
- if (ams_delta_muted == mute)
- return 0;
-
- spin_lock_bh(&ams_delta_lock);
- ams_delta_muted = mute;
- apply = !cx81801_cmd_pending;
- spin_unlock_bh(&ams_delta_lock);
-
- if (apply)
- ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
- mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
- return 0;
-}
-
-/* Our codec DAI probably doesn't have its own .ops structure */
-static const struct snd_soc_dai_ops ams_delta_dai_ops = {
- .digital_mute = ams_delta_digital_mute,
-};
-
-/* Will be used if the codec ever has its own digital_mute function */
-static int ams_delta_startup(struct snd_pcm_substream *substream)
-{
- return ams_delta_digital_mute(NULL, 0);
-}
-
-static void ams_delta_shutdown(struct snd_pcm_substream *substream)
-{
- ams_delta_digital_mute(NULL, 1);
-}
-
-
-/*
- * Card initialization
- */
-
-static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_card *card = rtd->card;
- int ret;
- /* Codec is ready, now add/activate board specific controls */
-
- /* Store a pointer to the codec structure for tty ldisc use */
- cx20442_codec = codec;
-
- /* Set up digital mute if not provided by the codec */
- if (!codec_dai->driver->ops) {
- codec_dai->driver->ops = &ams_delta_dai_ops;
- } else {
- ams_delta_ops.startup = ams_delta_startup;
- ams_delta_ops.shutdown = ams_delta_shutdown;
- }
-
- /* Add hook switch - can be used to control the codec from userspace
- * even if line discipline fails */
- ret = snd_soc_jack_new(rtd->codec, "hook_switch",
- SND_JACK_HEADSET, &ams_delta_hook_switch);
- if (ret)
- dev_warn(card->dev,
- "Failed to allocate resources for hook switch, "
- "will continue without one.\n");
- else {
- ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
- ARRAY_SIZE(ams_delta_hook_switch_gpios),
- ams_delta_hook_switch_gpios);
- if (ret)
- dev_warn(card->dev,
- "Failed to set up hook switch GPIO line, "
- "will continue with hook switch inactive.\n");
- }
-
- /* Register optional line discipline for over the modem control */
- ret = tty_register_ldisc(N_V253, &cx81801_ops);
- if (ret) {
- dev_warn(card->dev,
- "Failed to register line discipline, "
- "will continue without any controls.\n");
- return 0;
- }
-
- /* Add board specific DAPM widgets and routes */
- ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
- ARRAY_SIZE(ams_delta_dapm_widgets));
- if (ret) {
- dev_warn(card->dev,
- "Failed to register DAPM controls, "
- "will continue without any.\n");
- return 0;
- }
-
- ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
- ARRAY_SIZE(ams_delta_audio_map));
- if (ret) {
- dev_warn(card->dev,
- "Failed to set up DAPM routes, "
- "will continue with codec default map.\n");
- return 0;
- }
-
- /* Set up initial pin constellation */
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- snd_soc_dapm_enable_pin(dapm, "Microphone");
- snd_soc_dapm_disable_pin(dapm, "Speaker");
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
- snd_soc_dapm_disable_pin(dapm, "AGCOUT");
-
- /* Add virtual switch */
- ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls,
- ARRAY_SIZE(ams_delta_audio_controls));
- if (ret)
- dev_warn(card->dev,
- "Failed to register audio mode control, "
- "will continue without it.\n");
-
- return 0;
-}
-
-/* DAI glue - connects codec <--> CPU */
-static struct snd_soc_dai_link ams_delta_dai_link = {
- .name = "CX20442",
- .stream_name = "CX20442",
- .cpu_dai_name = "omap-mcbsp.1",
- .codec_dai_name = "cx20442-voice",
- .init = ams_delta_cx20442_init,
- .platform_name = "omap-pcm-audio",
- .codec_name = "cx20442-codec",
- .ops = &ams_delta_ops,
-};
-
-/* Audio card driver */
-static struct snd_soc_card ams_delta_audio_card = {
- .name = "AMS_DELTA",
- .owner = THIS_MODULE,
- .dai_link = &ams_delta_dai_link,
- .num_links = 1,
-};
-
-/* Module init/exit */
-static struct platform_device *ams_delta_audio_platform_device;
-static struct platform_device *cx20442_platform_device;
-
-static int __init ams_delta_module_init(void)
-{
- int ret;
-
- if (!(machine_is_ams_delta()))
- return -ENODEV;
-
- ams_delta_audio_platform_device =
- platform_device_alloc("soc-audio", -1);
- if (!ams_delta_audio_platform_device)
- return -ENOMEM;
-
- platform_set_drvdata(ams_delta_audio_platform_device,
- &ams_delta_audio_card);
-
- ret = platform_device_add(ams_delta_audio_platform_device);
- if (ret)
- goto err;
-
- /*
- * Codec platform device could be registered from elsewhere (board?),
- * but I do it here as it makes sense only if used with the card.
- */
- cx20442_platform_device =
- platform_device_register_simple("cx20442-codec", -1, NULL, 0);
- return 0;
-err:
- platform_device_put(ams_delta_audio_platform_device);
- return ret;
-}
-late_initcall(ams_delta_module_init);
-
-static void __exit ams_delta_module_exit(void)
-{
- if (tty_unregister_ldisc(N_V253) != 0)
- dev_warn(&ams_delta_audio_platform_device->dev,
- "failed to unregister V253 line discipline\n");
-
- snd_soc_jack_free_gpios(&ams_delta_hook_switch,
- ARRAY_SIZE(ams_delta_hook_switch_gpios),
- ams_delta_hook_switch_gpios);
-
- platform_device_unregister(cx20442_platform_device);
- platform_device_unregister(ams_delta_audio_platform_device);
-}
-module_exit(ams_delta_module_exit);
-
-MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
-MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/igep0020.c b/ANDROID_3.4.5/sound/soc/omap/igep0020.c
deleted file mode 100644
index e8357819..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/igep0020.c
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * igep0020.c -- SoC audio for IGEP v2
- *
- * Based on sound/soc/omap/overo.c by Steve Sakoman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-static int igep2_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops igep2_ops = {
- .hw_params = igep2_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link igep2_dai = {
- .name = "TWL4030",
- .stream_name = "TWL4030",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &igep2_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_card_igep2 = {
- .name = "igep2",
- .owner = THIS_MODULE,
- .dai_link = &igep2_dai,
- .num_links = 1,
-};
-
-static struct platform_device *igep2_snd_device;
-
-static int __init igep2_soc_init(void)
-{
- int ret;
-
- if (!machine_is_igep0020())
- return -ENODEV;
- printk(KERN_INFO "IGEP v2 SoC init\n");
-
- igep2_snd_device = platform_device_alloc("soc-audio", -1);
- if (!igep2_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(igep2_snd_device, &snd_soc_card_igep2);
-
- ret = platform_device_add(igep2_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(igep2_snd_device);
-
- return ret;
-}
-module_init(igep2_soc_init);
-
-static void __exit igep2_soc_exit(void)
-{
- platform_device_unregister(igep2_snd_device);
-}
-module_exit(igep2_soc_exit);
-
-MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo@iseebcn.com>");
-MODULE_DESCRIPTION("ALSA SoC IGEP v2");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/mcbsp.c b/ANDROID_3.4.5/sound/soc/omap/mcbsp.c
deleted file mode 100644
index e5f44440..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/mcbsp.c
+++ /dev/null
@@ -1,1040 +0,0 @@
-/*
- * sound/soc/omap/mcbsp.c
- *
- * Copyright (C) 2004 Nokia Corporation
- * Author: Samuel Ortiz <samuel.ortiz@nokia.com>
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * Multichannel mode not supported.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/device.h>
-#include <linux/platform_device.h>
-#include <linux/interrupt.h>
-#include <linux/err.h>
-#include <linux/clk.h>
-#include <linux/delay.h>
-#include <linux/io.h>
-#include <linux/slab.h>
-
-#include <plat/mcbsp.h>
-
-#include "mcbsp.h"
-
-static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
-{
- void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step;
-
- if (mcbsp->pdata->reg_size == 2) {
- ((u16 *)mcbsp->reg_cache)[reg] = (u16)val;
- __raw_writew((u16)val, addr);
- } else {
- ((u32 *)mcbsp->reg_cache)[reg] = val;
- __raw_writel(val, addr);
- }
-}
-
-static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache)
-{
- void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step;
-
- if (mcbsp->pdata->reg_size == 2) {
- return !from_cache ? __raw_readw(addr) :
- ((u16 *)mcbsp->reg_cache)[reg];
- } else {
- return !from_cache ? __raw_readl(addr) :
- ((u32 *)mcbsp->reg_cache)[reg];
- }
-}
-
-static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
-{
- __raw_writel(val, mcbsp->st_data->io_base_st + reg);
-}
-
-static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg)
-{
- return __raw_readl(mcbsp->st_data->io_base_st + reg);
-}
-
-#define MCBSP_READ(mcbsp, reg) \
- omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0)
-#define MCBSP_WRITE(mcbsp, reg, val) \
- omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val)
-#define MCBSP_READ_CACHE(mcbsp, reg) \
- omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1)
-
-#define MCBSP_ST_READ(mcbsp, reg) \
- omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg)
-#define MCBSP_ST_WRITE(mcbsp, reg, val) \
- omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val)
-
-static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp)
-{
- dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id);
- dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n",
- MCBSP_READ(mcbsp, DRR2));
- dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n",
- MCBSP_READ(mcbsp, DRR1));
- dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n",
- MCBSP_READ(mcbsp, DXR2));
- dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n",
- MCBSP_READ(mcbsp, DXR1));
- dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n",
- MCBSP_READ(mcbsp, SPCR2));
- dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n",
- MCBSP_READ(mcbsp, SPCR1));
- dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n",
- MCBSP_READ(mcbsp, RCR2));
- dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n",
- MCBSP_READ(mcbsp, RCR1));
- dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n",
- MCBSP_READ(mcbsp, XCR2));
- dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n",
- MCBSP_READ(mcbsp, XCR1));
- dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n",
- MCBSP_READ(mcbsp, SRGR2));
- dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n",
- MCBSP_READ(mcbsp, SRGR1));
- dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n",
- MCBSP_READ(mcbsp, PCR0));
- dev_dbg(mcbsp->dev, "***********************\n");
-}
-
-static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id)
-{
- struct omap_mcbsp *mcbsp_tx = dev_id;
- u16 irqst_spcr2;
-
- irqst_spcr2 = MCBSP_READ(mcbsp_tx, SPCR2);
- dev_dbg(mcbsp_tx->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2);
-
- if (irqst_spcr2 & XSYNC_ERR) {
- dev_err(mcbsp_tx->dev, "TX Frame Sync Error! : 0x%x\n",
- irqst_spcr2);
- /* Writing zero to XSYNC_ERR clears the IRQ */
- MCBSP_WRITE(mcbsp_tx, SPCR2, MCBSP_READ_CACHE(mcbsp_tx, SPCR2));
- }
-
- return IRQ_HANDLED;
-}
-
-static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id)
-{
- struct omap_mcbsp *mcbsp_rx = dev_id;
- u16 irqst_spcr1;
-
- irqst_spcr1 = MCBSP_READ(mcbsp_rx, SPCR1);
- dev_dbg(mcbsp_rx->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1);
-
- if (irqst_spcr1 & RSYNC_ERR) {
- dev_err(mcbsp_rx->dev, "RX Frame Sync Error! : 0x%x\n",
- irqst_spcr1);
- /* Writing zero to RSYNC_ERR clears the IRQ */
- MCBSP_WRITE(mcbsp_rx, SPCR1, MCBSP_READ_CACHE(mcbsp_rx, SPCR1));
- }
-
- return IRQ_HANDLED;
-}
-
-/*
- * omap_mcbsp_config simply write a config to the
- * appropriate McBSP.
- * You either call this function or set the McBSP registers
- * by yourself before calling omap_mcbsp_start().
- */
-void omap_mcbsp_config(struct omap_mcbsp *mcbsp,
- const struct omap_mcbsp_reg_cfg *config)
-{
- dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n",
- mcbsp->id, mcbsp->phys_base);
-
- /* We write the given config */
- MCBSP_WRITE(mcbsp, SPCR2, config->spcr2);
- MCBSP_WRITE(mcbsp, SPCR1, config->spcr1);
- MCBSP_WRITE(mcbsp, RCR2, config->rcr2);
- MCBSP_WRITE(mcbsp, RCR1, config->rcr1);
- MCBSP_WRITE(mcbsp, XCR2, config->xcr2);
- MCBSP_WRITE(mcbsp, XCR1, config->xcr1);
- MCBSP_WRITE(mcbsp, SRGR2, config->srgr2);
- MCBSP_WRITE(mcbsp, SRGR1, config->srgr1);
- MCBSP_WRITE(mcbsp, MCR2, config->mcr2);
- MCBSP_WRITE(mcbsp, MCR1, config->mcr1);
- MCBSP_WRITE(mcbsp, PCR0, config->pcr0);
- if (mcbsp->pdata->has_ccr) {
- MCBSP_WRITE(mcbsp, XCCR, config->xccr);
- MCBSP_WRITE(mcbsp, RCCR, config->rccr);
- }
- /* Enable wakeup behavior */
- if (mcbsp->pdata->has_wakeup)
- MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN);
-}
-
-/**
- * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register
- * @id - mcbsp id
- * @stream - indicates the direction of data flow (rx or tx)
- *
- * Returns the address of mcbsp data transmit register or data receive register
- * to be used by DMA for transferring/receiving data based on the value of
- * @stream for the requested mcbsp given by @id
- */
-static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp,
- unsigned int stream)
-{
- int data_reg;
-
- if (mcbsp->pdata->reg_size == 2) {
- if (stream)
- data_reg = OMAP_MCBSP_REG_DRR1;
- else
- data_reg = OMAP_MCBSP_REG_DXR1;
- } else {
- if (stream)
- data_reg = OMAP_MCBSP_REG_DRR;
- else
- data_reg = OMAP_MCBSP_REG_DXR;
- }
-
- return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step;
-}
-
-static void omap_st_on(struct omap_mcbsp *mcbsp)
-{
- unsigned int w;
-
- if (mcbsp->pdata->enable_st_clock)
- mcbsp->pdata->enable_st_clock(mcbsp->id, 1);
-
- /* Enable McBSP Sidetone */
- w = MCBSP_READ(mcbsp, SSELCR);
- MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN);
-
- /* Enable Sidetone from Sidetone Core */
- w = MCBSP_ST_READ(mcbsp, SSELCR);
- MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN);
-}
-
-static void omap_st_off(struct omap_mcbsp *mcbsp)
-{
- unsigned int w;
-
- w = MCBSP_ST_READ(mcbsp, SSELCR);
- MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN));
-
- w = MCBSP_READ(mcbsp, SSELCR);
- MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN));
-
- if (mcbsp->pdata->enable_st_clock)
- mcbsp->pdata->enable_st_clock(mcbsp->id, 0);
-}
-
-static void omap_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir)
-{
- u16 val, i;
-
- val = MCBSP_ST_READ(mcbsp, SSELCR);
-
- if (val & ST_COEFFWREN)
- MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN));
-
- MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN);
-
- for (i = 0; i < 128; i++)
- MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]);
-
- i = 0;
-
- val = MCBSP_ST_READ(mcbsp, SSELCR);
- while (!(val & ST_COEFFWRDONE) && (++i < 1000))
- val = MCBSP_ST_READ(mcbsp, SSELCR);
-
- MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN));
-
- if (i == 1000)
- dev_err(mcbsp->dev, "McBSP FIR load error!\n");
-}
-
-static void omap_st_chgain(struct omap_mcbsp *mcbsp)
-{
- u16 w;
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
-
- w = MCBSP_ST_READ(mcbsp, SSELCR);
-
- MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | \
- ST_CH1GAIN(st_data->ch1gain));
-}
-
-int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
- int ret = 0;
-
- if (!st_data)
- return -ENOENT;
-
- spin_lock_irq(&mcbsp->lock);
- if (channel == 0)
- st_data->ch0gain = chgain;
- else if (channel == 1)
- st_data->ch1gain = chgain;
- else
- ret = -EINVAL;
-
- if (st_data->enabled)
- omap_st_chgain(mcbsp);
- spin_unlock_irq(&mcbsp->lock);
-
- return ret;
-}
-
-int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
- int ret = 0;
-
- if (!st_data)
- return -ENOENT;
-
- spin_lock_irq(&mcbsp->lock);
- if (channel == 0)
- *chgain = st_data->ch0gain;
- else if (channel == 1)
- *chgain = st_data->ch1gain;
- else
- ret = -EINVAL;
- spin_unlock_irq(&mcbsp->lock);
-
- return ret;
-}
-
-static int omap_st_start(struct omap_mcbsp *mcbsp)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
-
- if (st_data->enabled && !st_data->running) {
- omap_st_fir_write(mcbsp, st_data->taps);
- omap_st_chgain(mcbsp);
-
- if (!mcbsp->free) {
- omap_st_on(mcbsp);
- st_data->running = 1;
- }
- }
-
- return 0;
-}
-
-int omap_st_enable(struct omap_mcbsp *mcbsp)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
-
- if (!st_data)
- return -ENODEV;
-
- spin_lock_irq(&mcbsp->lock);
- st_data->enabled = 1;
- omap_st_start(mcbsp);
- spin_unlock_irq(&mcbsp->lock);
-
- return 0;
-}
-
-static int omap_st_stop(struct omap_mcbsp *mcbsp)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
-
- if (st_data->running) {
- if (!mcbsp->free) {
- omap_st_off(mcbsp);
- st_data->running = 0;
- }
- }
-
- return 0;
-}
-
-int omap_st_disable(struct omap_mcbsp *mcbsp)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
- int ret = 0;
-
- if (!st_data)
- return -ENODEV;
-
- spin_lock_irq(&mcbsp->lock);
- omap_st_stop(mcbsp);
- st_data->enabled = 0;
- spin_unlock_irq(&mcbsp->lock);
-
- return ret;
-}
-
-int omap_st_is_enabled(struct omap_mcbsp *mcbsp)
-{
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
-
- if (!st_data)
- return -ENODEV;
-
- return st_data->enabled;
-}
-
-/*
- * omap_mcbsp_set_rx_threshold configures the transmit threshold in words.
- * The threshold parameter is 1 based, and it is converted (threshold - 1)
- * for the THRSH2 register.
- */
-void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold)
-{
- if (mcbsp->pdata->buffer_size == 0)
- return;
-
- if (threshold && threshold <= mcbsp->max_tx_thres)
- MCBSP_WRITE(mcbsp, THRSH2, threshold - 1);
-}
-
-/*
- * omap_mcbsp_set_rx_threshold configures the receive threshold in words.
- * The threshold parameter is 1 based, and it is converted (threshold - 1)
- * for the THRSH1 register.
- */
-void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold)
-{
- if (mcbsp->pdata->buffer_size == 0)
- return;
-
- if (threshold && threshold <= mcbsp->max_rx_thres)
- MCBSP_WRITE(mcbsp, THRSH1, threshold - 1);
-}
-
-/*
- * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO
- */
-u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp)
-{
- u16 buffstat;
-
- if (mcbsp->pdata->buffer_size == 0)
- return 0;
-
- /* Returns the number of free locations in the buffer */
- buffstat = MCBSP_READ(mcbsp, XBUFFSTAT);
-
- /* Number of slots are different in McBSP ports */
- return mcbsp->pdata->buffer_size - buffstat;
-}
-
-/*
- * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO
- * to reach the threshold value (when the DMA will be triggered to read it)
- */
-u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp)
-{
- u16 buffstat, threshold;
-
- if (mcbsp->pdata->buffer_size == 0)
- return 0;
-
- /* Returns the number of used locations in the buffer */
- buffstat = MCBSP_READ(mcbsp, RBUFFSTAT);
- /* RX threshold */
- threshold = MCBSP_READ(mcbsp, THRSH1);
-
- /* Return the number of location till we reach the threshold limit */
- if (threshold <= buffstat)
- return 0;
- else
- return threshold - buffstat;
-}
-
-int omap_mcbsp_request(struct omap_mcbsp *mcbsp)
-{
- void *reg_cache;
- int err;
-
- reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL);
- if (!reg_cache) {
- return -ENOMEM;
- }
-
- spin_lock(&mcbsp->lock);
- if (!mcbsp->free) {
- dev_err(mcbsp->dev, "McBSP%d is currently in use\n",
- mcbsp->id);
- err = -EBUSY;
- goto err_kfree;
- }
-
- mcbsp->free = false;
- mcbsp->reg_cache = reg_cache;
- spin_unlock(&mcbsp->lock);
-
- if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request)
- mcbsp->pdata->ops->request(mcbsp->id - 1);
-
- /*
- * Make sure that transmitter, receiver and sample-rate generator are
- * not running before activating IRQs.
- */
- MCBSP_WRITE(mcbsp, SPCR1, 0);
- MCBSP_WRITE(mcbsp, SPCR2, 0);
-
- err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler,
- 0, "McBSP", (void *)mcbsp);
- if (err != 0) {
- dev_err(mcbsp->dev, "Unable to request TX IRQ %d "
- "for McBSP%d\n", mcbsp->tx_irq,
- mcbsp->id);
- goto err_clk_disable;
- }
-
- if (mcbsp->rx_irq) {
- err = request_irq(mcbsp->rx_irq,
- omap_mcbsp_rx_irq_handler,
- 0, "McBSP", (void *)mcbsp);
- if (err != 0) {
- dev_err(mcbsp->dev, "Unable to request RX IRQ %d "
- "for McBSP%d\n", mcbsp->rx_irq,
- mcbsp->id);
- goto err_free_irq;
- }
- }
-
- return 0;
-err_free_irq:
- free_irq(mcbsp->tx_irq, (void *)mcbsp);
-err_clk_disable:
- if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free)
- mcbsp->pdata->ops->free(mcbsp->id - 1);
-
- /* Disable wakeup behavior */
- if (mcbsp->pdata->has_wakeup)
- MCBSP_WRITE(mcbsp, WAKEUPEN, 0);
-
- spin_lock(&mcbsp->lock);
- mcbsp->free = true;
- mcbsp->reg_cache = NULL;
-err_kfree:
- spin_unlock(&mcbsp->lock);
- kfree(reg_cache);
-
- return err;
-}
-
-void omap_mcbsp_free(struct omap_mcbsp *mcbsp)
-{
- void *reg_cache;
-
- if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free)
- mcbsp->pdata->ops->free(mcbsp->id - 1);
-
- /* Disable wakeup behavior */
- if (mcbsp->pdata->has_wakeup)
- MCBSP_WRITE(mcbsp, WAKEUPEN, 0);
-
- if (mcbsp->rx_irq)
- free_irq(mcbsp->rx_irq, (void *)mcbsp);
- free_irq(mcbsp->tx_irq, (void *)mcbsp);
-
- reg_cache = mcbsp->reg_cache;
-
- /*
- * Select CLKS source from internal source unconditionally before
- * marking the McBSP port as free.
- * If the external clock source via MCBSP_CLKS pin has been selected the
- * system will refuse to enter idle if the CLKS pin source is not reset
- * back to internal source.
- */
- if (!cpu_class_is_omap1())
- omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC);
-
- spin_lock(&mcbsp->lock);
- if (mcbsp->free)
- dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id);
- else
- mcbsp->free = true;
- mcbsp->reg_cache = NULL;
- spin_unlock(&mcbsp->lock);
-
- if (reg_cache)
- kfree(reg_cache);
-}
-
-/*
- * Here we start the McBSP, by enabling transmitter, receiver or both.
- * If no transmitter or receiver is active prior calling, then sample-rate
- * generator and frame sync are started.
- */
-void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx)
-{
- int enable_srg = 0;
- u16 w;
-
- if (mcbsp->st_data)
- omap_st_start(mcbsp);
-
- /* Only enable SRG, if McBSP is master */
- w = MCBSP_READ_CACHE(mcbsp, PCR0);
- if (w & (FSXM | FSRM | CLKXM | CLKRM))
- enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) |
- MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1);
-
- if (enable_srg) {
- /* Start the sample generator */
- w = MCBSP_READ_CACHE(mcbsp, SPCR2);
- MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6));
- }
-
- /* Enable transmitter and receiver */
- tx &= 1;
- w = MCBSP_READ_CACHE(mcbsp, SPCR2);
- MCBSP_WRITE(mcbsp, SPCR2, w | tx);
-
- rx &= 1;
- w = MCBSP_READ_CACHE(mcbsp, SPCR1);
- MCBSP_WRITE(mcbsp, SPCR1, w | rx);
-
- /*
- * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec
- * REVISIT: 100us may give enough time for two CLKSRG, however
- * due to some unknown PM related, clock gating etc. reason it
- * is now at 500us.
- */
- udelay(500);
-
- if (enable_srg) {
- /* Start frame sync */
- w = MCBSP_READ_CACHE(mcbsp, SPCR2);
- MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7));
- }
-
- if (mcbsp->pdata->has_ccr) {
- /* Release the transmitter and receiver */
- w = MCBSP_READ_CACHE(mcbsp, XCCR);
- w &= ~(tx ? XDISABLE : 0);
- MCBSP_WRITE(mcbsp, XCCR, w);
- w = MCBSP_READ_CACHE(mcbsp, RCCR);
- w &= ~(rx ? RDISABLE : 0);
- MCBSP_WRITE(mcbsp, RCCR, w);
- }
-
- /* Dump McBSP Regs */
- omap_mcbsp_dump_reg(mcbsp);
-}
-
-void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx)
-{
- int idle;
- u16 w;
-
- /* Reset transmitter */
- tx &= 1;
- if (mcbsp->pdata->has_ccr) {
- w = MCBSP_READ_CACHE(mcbsp, XCCR);
- w |= (tx ? XDISABLE : 0);
- MCBSP_WRITE(mcbsp, XCCR, w);
- }
- w = MCBSP_READ_CACHE(mcbsp, SPCR2);
- MCBSP_WRITE(mcbsp, SPCR2, w & ~tx);
-
- /* Reset receiver */
- rx &= 1;
- if (mcbsp->pdata->has_ccr) {
- w = MCBSP_READ_CACHE(mcbsp, RCCR);
- w |= (rx ? RDISABLE : 0);
- MCBSP_WRITE(mcbsp, RCCR, w);
- }
- w = MCBSP_READ_CACHE(mcbsp, SPCR1);
- MCBSP_WRITE(mcbsp, SPCR1, w & ~rx);
-
- idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) |
- MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1);
-
- if (idle) {
- /* Reset the sample rate generator */
- w = MCBSP_READ_CACHE(mcbsp, SPCR2);
- MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6));
- }
-
- if (mcbsp->st_data)
- omap_st_stop(mcbsp);
-}
-
-int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id)
-{
- const char *src;
-
- if (fck_src_id == MCBSP_CLKS_PAD_SRC)
- src = "clks_ext";
- else if (fck_src_id == MCBSP_CLKS_PRCM_SRC)
- src = "clks_fclk";
- else
- return -EINVAL;
-
- if (mcbsp->pdata->set_clk_src)
- return mcbsp->pdata->set_clk_src(mcbsp->dev, mcbsp->fclk, src);
- else
- return -EINVAL;
-}
-
-int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux)
-{
- const char *signal, *src;
-
- if (mcbsp->pdata->mux_signal)
- return -EINVAL;
-
- switch (mux) {
- case CLKR_SRC_CLKR:
- signal = "clkr";
- src = "clkr";
- break;
- case CLKR_SRC_CLKX:
- signal = "clkr";
- src = "clkx";
- break;
- case FSR_SRC_FSR:
- signal = "fsr";
- src = "fsr";
- break;
- case FSR_SRC_FSX:
- signal = "fsr";
- src = "fsx";
- break;
- default:
- return -EINVAL;
- }
-
- return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src);
-}
-
-#define max_thres(m) (mcbsp->pdata->buffer_size)
-#define valid_threshold(m, val) ((val) <= max_thres(m))
-#define THRESHOLD_PROP_BUILDER(prop) \
-static ssize_t prop##_show(struct device *dev, \
- struct device_attribute *attr, char *buf) \
-{ \
- struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \
- \
- return sprintf(buf, "%u\n", mcbsp->prop); \
-} \
- \
-static ssize_t prop##_store(struct device *dev, \
- struct device_attribute *attr, \
- const char *buf, size_t size) \
-{ \
- struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \
- unsigned long val; \
- int status; \
- \
- status = strict_strtoul(buf, 0, &val); \
- if (status) \
- return status; \
- \
- if (!valid_threshold(mcbsp, val)) \
- return -EDOM; \
- \
- mcbsp->prop = val; \
- return size; \
-} \
- \
-static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store);
-
-THRESHOLD_PROP_BUILDER(max_tx_thres);
-THRESHOLD_PROP_BUILDER(max_rx_thres);
-
-static const char *dma_op_modes[] = {
- "element", "threshold", "frame",
-};
-
-static ssize_t dma_op_mode_show(struct device *dev,
- struct device_attribute *attr, char *buf)
-{
- struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
- int dma_op_mode, i = 0;
- ssize_t len = 0;
- const char * const *s;
-
- dma_op_mode = mcbsp->dma_op_mode;
-
- for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) {
- if (dma_op_mode == i)
- len += sprintf(buf + len, "[%s] ", *s);
- else
- len += sprintf(buf + len, "%s ", *s);
- }
- len += sprintf(buf + len, "\n");
-
- return len;
-}
-
-static ssize_t dma_op_mode_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf, size_t size)
-{
- struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
- const char * const *s;
- int i = 0;
-
- for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++)
- if (sysfs_streq(buf, *s))
- break;
-
- if (i == ARRAY_SIZE(dma_op_modes))
- return -EINVAL;
-
- spin_lock_irq(&mcbsp->lock);
- if (!mcbsp->free) {
- size = -EBUSY;
- goto unlock;
- }
- mcbsp->dma_op_mode = i;
-
-unlock:
- spin_unlock_irq(&mcbsp->lock);
-
- return size;
-}
-
-static DEVICE_ATTR(dma_op_mode, 0644, dma_op_mode_show, dma_op_mode_store);
-
-static const struct attribute *additional_attrs[] = {
- &dev_attr_max_tx_thres.attr,
- &dev_attr_max_rx_thres.attr,
- &dev_attr_dma_op_mode.attr,
- NULL,
-};
-
-static const struct attribute_group additional_attr_group = {
- .attrs = (struct attribute **)additional_attrs,
-};
-
-static ssize_t st_taps_show(struct device *dev,
- struct device_attribute *attr, char *buf)
-{
- struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
- ssize_t status = 0;
- int i;
-
- spin_lock_irq(&mcbsp->lock);
- for (i = 0; i < st_data->nr_taps; i++)
- status += sprintf(&buf[status], (i ? ", %d" : "%d"),
- st_data->taps[i]);
- if (i)
- status += sprintf(&buf[status], "\n");
- spin_unlock_irq(&mcbsp->lock);
-
- return status;
-}
-
-static ssize_t st_taps_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf, size_t size)
-{
- struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
- struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
- int val, tmp, status, i = 0;
-
- spin_lock_irq(&mcbsp->lock);
- memset(st_data->taps, 0, sizeof(st_data->taps));
- st_data->nr_taps = 0;
-
- do {
- status = sscanf(buf, "%d%n", &val, &tmp);
- if (status < 0 || status == 0) {
- size = -EINVAL;
- goto out;
- }
- if (val < -32768 || val > 32767) {
- size = -EINVAL;
- goto out;
- }
- st_data->taps[i++] = val;
- buf += tmp;
- if (*buf != ',')
- break;
- buf++;
- } while (1);
-
- st_data->nr_taps = i;
-
-out:
- spin_unlock_irq(&mcbsp->lock);
-
- return size;
-}
-
-static DEVICE_ATTR(st_taps, 0644, st_taps_show, st_taps_store);
-
-static const struct attribute *sidetone_attrs[] = {
- &dev_attr_st_taps.attr,
- NULL,
-};
-
-static const struct attribute_group sidetone_attr_group = {
- .attrs = (struct attribute **)sidetone_attrs,
-};
-
-static int __devinit omap_st_add(struct omap_mcbsp *mcbsp,
- struct resource *res)
-{
- struct omap_mcbsp_st_data *st_data;
- int err;
-
- st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL);
- if (!st_data)
- return -ENOMEM;
-
- st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start,
- resource_size(res));
- if (!st_data->io_base_st)
- return -ENOMEM;
-
- err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group);
- if (err)
- return err;
-
- mcbsp->st_data = st_data;
- return 0;
-}
-
-/*
- * McBSP1 and McBSP3 are directly mapped on 1610 and 1510.
- * 730 has only 2 McBSP, and both of them are MPU peripherals.
- */
-int __devinit omap_mcbsp_init(struct platform_device *pdev)
-{
- struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
- struct resource *res;
- int ret = 0;
-
- spin_lock_init(&mcbsp->lock);
- mcbsp->free = true;
-
- res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
- if (!res) {
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!res) {
- dev_err(mcbsp->dev, "invalid memory resource\n");
- return -ENOMEM;
- }
- }
- if (!devm_request_mem_region(&pdev->dev, res->start, resource_size(res),
- dev_name(&pdev->dev))) {
- dev_err(mcbsp->dev, "memory region already claimed\n");
- return -ENODEV;
- }
-
- mcbsp->phys_base = res->start;
- mcbsp->reg_cache_size = resource_size(res);
- mcbsp->io_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!mcbsp->io_base)
- return -ENOMEM;
-
- res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma");
- if (!res)
- mcbsp->phys_dma_base = mcbsp->phys_base;
- else
- mcbsp->phys_dma_base = res->start;
-
- mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx");
- mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx");
-
- /* From OMAP4 there will be a single irq line */
- if (mcbsp->tx_irq == -ENXIO) {
- mcbsp->tx_irq = platform_get_irq(pdev, 0);
- mcbsp->rx_irq = 0;
- }
-
- res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx");
- if (!res) {
- dev_err(&pdev->dev, "invalid rx DMA channel\n");
- return -ENODEV;
- }
- /* RX DMA request number, and port address configuration */
- mcbsp->dma_data[1].name = "Audio Capture";
- mcbsp->dma_data[1].dma_req = res->start;
- mcbsp->dma_data[1].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 1);
-
- res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx");
- if (!res) {
- dev_err(&pdev->dev, "invalid tx DMA channel\n");
- return -ENODEV;
- }
- /* TX DMA request number, and port address configuration */
- mcbsp->dma_data[0].name = "Audio Playback";
- mcbsp->dma_data[0].dma_req = res->start;
- mcbsp->dma_data[0].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 0);
-
- mcbsp->fclk = clk_get(&pdev->dev, "fck");
- if (IS_ERR(mcbsp->fclk)) {
- ret = PTR_ERR(mcbsp->fclk);
- dev_err(mcbsp->dev, "unable to get fck: %d\n", ret);
- return ret;
- }
-
- mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT;
- if (mcbsp->pdata->buffer_size) {
- /*
- * Initially configure the maximum thresholds to a safe value.
- * The McBSP FIFO usage with these values should not go under
- * 16 locations.
- * If the whole FIFO without safety buffer is used, than there
- * is a possibility that the DMA will be not able to push the
- * new data on time, causing channel shifts in runtime.
- */
- mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
- mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
-
- ret = sysfs_create_group(&mcbsp->dev->kobj,
- &additional_attr_group);
- if (ret) {
- dev_err(mcbsp->dev,
- "Unable to create additional controls\n");
- goto err_thres;
- }
- } else {
- mcbsp->max_tx_thres = -EINVAL;
- mcbsp->max_rx_thres = -EINVAL;
- }
-
- res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone");
- if (res) {
- ret = omap_st_add(mcbsp, res);
- if (ret) {
- dev_err(mcbsp->dev,
- "Unable to create sidetone controls\n");
- goto err_st;
- }
- }
-
- return 0;
-
-err_st:
- if (mcbsp->pdata->buffer_size)
- sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-err_thres:
- clk_put(mcbsp->fclk);
- return ret;
-}
-
-void __devexit omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp)
-{
- if (mcbsp->pdata->buffer_size)
- sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-
- if (mcbsp->st_data)
- sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group);
-}
diff --git a/ANDROID_3.4.5/sound/soc/omap/mcbsp.h b/ANDROID_3.4.5/sound/soc/omap/mcbsp.h
deleted file mode 100644
index a944fcc9..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/mcbsp.h
+++ /dev/null
@@ -1,346 +0,0 @@
-/*
- * sound/soc/omap/mcbsp.h
- *
- * OMAP Multi-Channel Buffered Serial Port
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.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.
- *
- * 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
- *
- */
-#ifndef __ASOC_MCBSP_H
-#define __ASOC_MCBSP_H
-
-#include "omap-pcm.h"
-
-/* McBSP register numbers. Register address offset = num * reg_step */
-enum {
- /* Common registers */
- OMAP_MCBSP_REG_SPCR2 = 4,
- OMAP_MCBSP_REG_SPCR1,
- OMAP_MCBSP_REG_RCR2,
- OMAP_MCBSP_REG_RCR1,
- OMAP_MCBSP_REG_XCR2,
- OMAP_MCBSP_REG_XCR1,
- OMAP_MCBSP_REG_SRGR2,
- OMAP_MCBSP_REG_SRGR1,
- OMAP_MCBSP_REG_MCR2,
- OMAP_MCBSP_REG_MCR1,
- OMAP_MCBSP_REG_RCERA,
- OMAP_MCBSP_REG_RCERB,
- OMAP_MCBSP_REG_XCERA,
- OMAP_MCBSP_REG_XCERB,
- OMAP_MCBSP_REG_PCR0,
- OMAP_MCBSP_REG_RCERC,
- OMAP_MCBSP_REG_RCERD,
- OMAP_MCBSP_REG_XCERC,
- OMAP_MCBSP_REG_XCERD,
- OMAP_MCBSP_REG_RCERE,
- OMAP_MCBSP_REG_RCERF,
- OMAP_MCBSP_REG_XCERE,
- OMAP_MCBSP_REG_XCERF,
- OMAP_MCBSP_REG_RCERG,
- OMAP_MCBSP_REG_RCERH,
- OMAP_MCBSP_REG_XCERG,
- OMAP_MCBSP_REG_XCERH,
-
- /* OMAP1-OMAP2420 registers */
- OMAP_MCBSP_REG_DRR2 = 0,
- OMAP_MCBSP_REG_DRR1,
- OMAP_MCBSP_REG_DXR2,
- OMAP_MCBSP_REG_DXR1,
-
- /* OMAP2430 and onwards */
- OMAP_MCBSP_REG_DRR = 0,
- OMAP_MCBSP_REG_DXR = 2,
- OMAP_MCBSP_REG_SYSCON = 35,
- OMAP_MCBSP_REG_THRSH2,
- OMAP_MCBSP_REG_THRSH1,
- OMAP_MCBSP_REG_IRQST = 40,
- OMAP_MCBSP_REG_IRQEN,
- OMAP_MCBSP_REG_WAKEUPEN,
- OMAP_MCBSP_REG_XCCR,
- OMAP_MCBSP_REG_RCCR,
- OMAP_MCBSP_REG_XBUFFSTAT,
- OMAP_MCBSP_REG_RBUFFSTAT,
- OMAP_MCBSP_REG_SSELCR,
-};
-
-/* OMAP3 sidetone control registers */
-#define OMAP_ST_REG_REV 0x00
-#define OMAP_ST_REG_SYSCONFIG 0x10
-#define OMAP_ST_REG_IRQSTATUS 0x18
-#define OMAP_ST_REG_IRQENABLE 0x1C
-#define OMAP_ST_REG_SGAINCR 0x24
-#define OMAP_ST_REG_SFIRCR 0x28
-#define OMAP_ST_REG_SSELCR 0x2C
-
-/************************** McBSP SPCR1 bit definitions ***********************/
-#define RRST BIT(0)
-#define RRDY BIT(1)
-#define RFULL BIT(2)
-#define RSYNC_ERR BIT(3)
-#define RINTM(value) (((value) & 0x3) << 4) /* bits 4:5 */
-#define ABIS BIT(6)
-#define DXENA BIT(7)
-#define CLKSTP(value) (((value) & 0x3) << 11) /* bits 11:12 */
-#define RJUST(value) (((value) & 0x3) << 13) /* bits 13:14 */
-#define ALB BIT(15)
-#define DLB BIT(15)
-
-/************************** McBSP SPCR2 bit definitions ***********************/
-#define XRST BIT(0)
-#define XRDY BIT(1)
-#define XEMPTY BIT(2)
-#define XSYNC_ERR BIT(3)
-#define XINTM(value) (((value) & 0x3) << 4) /* bits 4:5 */
-#define GRST BIT(6)
-#define FRST BIT(7)
-#define SOFT BIT(8)
-#define FREE BIT(9)
-
-/************************** McBSP PCR bit definitions *************************/
-#define CLKRP BIT(0)
-#define CLKXP BIT(1)
-#define FSRP BIT(2)
-#define FSXP BIT(3)
-#define DR_STAT BIT(4)
-#define DX_STAT BIT(5)
-#define CLKS_STAT BIT(6)
-#define SCLKME BIT(7)
-#define CLKRM BIT(8)
-#define CLKXM BIT(9)
-#define FSRM BIT(10)
-#define FSXM BIT(11)
-#define RIOEN BIT(12)
-#define XIOEN BIT(13)
-#define IDLE_EN BIT(14)
-
-/************************** McBSP RCR1 bit definitions ************************/
-#define RWDLEN1(value) (((value) & 0x7) << 5) /* Bits 5:7 */
-#define RFRLEN1(value) (((value) & 0x7f) << 8) /* Bits 8:14 */
-
-/************************** McBSP XCR1 bit definitions ************************/
-#define XWDLEN1(value) (((value) & 0x7) << 5) /* Bits 5:7 */
-#define XFRLEN1(value) (((value) & 0x7f) << 8) /* Bits 8:14 */
-
-/*************************** McBSP RCR2 bit definitions ***********************/
-#define RDATDLY(value) ((value) & 0x3) /* Bits 0:1 */
-#define RFIG BIT(2)
-#define RCOMPAND(value) (((value) & 0x3) << 3) /* Bits 3:4 */
-#define RWDLEN2(value) (((value) & 0x7) << 5) /* Bits 5:7 */
-#define RFRLEN2(value) (((value) & 0x7f) << 8) /* Bits 8:14 */
-#define RPHASE BIT(15)
-
-/*************************** McBSP XCR2 bit definitions ***********************/
-#define XDATDLY(value) ((value) & 0x3) /* Bits 0:1 */
-#define XFIG BIT(2)
-#define XCOMPAND(value) (((value) & 0x3) << 3) /* Bits 3:4 */
-#define XWDLEN2(value) (((value) & 0x7) << 5) /* Bits 5:7 */
-#define XFRLEN2(value) (((value) & 0x7f) << 8) /* Bits 8:14 */
-#define XPHASE BIT(15)
-
-/************************* McBSP SRGR1 bit definitions ************************/
-#define CLKGDV(value) ((value) & 0x7f) /* Bits 0:7 */
-#define FWID(value) (((value) & 0xff) << 8) /* Bits 8:15 */
-
-/************************* McBSP SRGR2 bit definitions ************************/
-#define FPER(value) ((value) & 0x0fff) /* Bits 0:11 */
-#define FSGM BIT(12)
-#define CLKSM BIT(13)
-#define CLKSP BIT(14)
-#define GSYNC BIT(15)
-
-/************************* McBSP MCR1 bit definitions *************************/
-#define RMCM BIT(0)
-#define RCBLK(value) (((value) & 0x7) << 2) /* Bits 2:4 */
-#define RPABLK(value) (((value) & 0x3) << 5) /* Bits 5:6 */
-#define RPBBLK(value) (((value) & 0x3) << 7) /* Bits 7:8 */
-
-/************************* McBSP MCR2 bit definitions *************************/
-#define XMCM(value) ((value) & 0x3) /* Bits 0:1 */
-#define XCBLK(value) (((value) & 0x7) << 2) /* Bits 2:4 */
-#define XPABLK(value) (((value) & 0x3) << 5) /* Bits 5:6 */
-#define XPBBLK(value) (((value) & 0x3) << 7) /* Bits 7:8 */
-
-/*********************** McBSP XCCR bit definitions *************************/
-#define XDISABLE BIT(0)
-#define XDMAEN BIT(3)
-#define DILB BIT(5)
-#define XFULL_CYCLE BIT(11)
-#define DXENDLY(value) (((value) & 0x3) << 12) /* Bits 12:13 */
-#define PPCONNECT BIT(14)
-#define EXTCLKGATE BIT(15)
-
-/********************** McBSP RCCR bit definitions *************************/
-#define RDISABLE BIT(0)
-#define RDMAEN BIT(3)
-#define RFULL_CYCLE BIT(11)
-
-/********************** McBSP SYSCONFIG bit definitions ********************/
-#define SOFTRST BIT(1)
-#define ENAWAKEUP BIT(2)
-#define SIDLEMODE(value) (((value) & 0x3) << 3)
-#define CLOCKACTIVITY(value) (((value) & 0x3) << 8)
-
-/********************** McBSP SSELCR bit definitions ***********************/
-#define SIDETONEEN BIT(10)
-
-/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/
-#define ST_AUTOIDLE BIT(0)
-
-/********************** McBSP Sidetone SGAINCR bit definitions *************/
-#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */
-#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */
-
-/********************** McBSP Sidetone SFIRCR bit definitions **************/
-#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */
-
-/********************** McBSP Sidetone SSELCR bit definitions **************/
-#define ST_SIDETONEEN BIT(0)
-#define ST_COEFFWREN BIT(1)
-#define ST_COEFFWRDONE BIT(2)
-
-/********************** McBSP DMA operating modes **************************/
-#define MCBSP_DMA_MODE_ELEMENT 0
-#define MCBSP_DMA_MODE_THRESHOLD 1
-#define MCBSP_DMA_MODE_FRAME 2
-
-/********************** McBSP WAKEUPEN bit definitions *********************/
-#define RSYNCERREN BIT(0)
-#define RFSREN BIT(1)
-#define REOFEN BIT(2)
-#define RRDYEN BIT(3)
-#define XSYNCERREN BIT(7)
-#define XFSXEN BIT(8)
-#define XEOFEN BIT(9)
-#define XRDYEN BIT(10)
-#define XEMPTYEOFEN BIT(14)
-
-/* Clock signal muxing options */
-#define CLKR_SRC_CLKR 0 /* CLKR signal is from the CLKR pin */
-#define CLKR_SRC_CLKX 1 /* CLKR signal is from the CLKX pin */
-#define FSR_SRC_FSR 2 /* FSR signal is from the FSR pin */
-#define FSR_SRC_FSX 3 /* FSR signal is from the FSX pin */
-
-/* McBSP functional clock sources */
-#define MCBSP_CLKS_PRCM_SRC 0
-#define MCBSP_CLKS_PAD_SRC 1
-
-/* we don't do multichannel for now */
-struct omap_mcbsp_reg_cfg {
- u16 spcr2;
- u16 spcr1;
- u16 rcr2;
- u16 rcr1;
- u16 xcr2;
- u16 xcr1;
- u16 srgr2;
- u16 srgr1;
- u16 mcr2;
- u16 mcr1;
- u16 pcr0;
- u16 rcerc;
- u16 rcerd;
- u16 xcerc;
- u16 xcerd;
- u16 rcere;
- u16 rcerf;
- u16 xcere;
- u16 xcerf;
- u16 rcerg;
- u16 rcerh;
- u16 xcerg;
- u16 xcerh;
- u16 xccr;
- u16 rccr;
-};
-
-struct omap_mcbsp_st_data {
- void __iomem *io_base_st;
- bool running;
- bool enabled;
- s16 taps[128]; /* Sidetone filter coefficients */
- int nr_taps; /* Number of filter coefficients in use */
- s16 ch0gain;
- s16 ch1gain;
-};
-
-struct omap_mcbsp {
- struct device *dev;
- struct clk *fclk;
- spinlock_t lock;
- unsigned long phys_base;
- unsigned long phys_dma_base;
- void __iomem *io_base;
- u8 id;
- /*
- * Flags indicating is the bus already activated and configured by
- * another substream
- */
- int active;
- int configured;
- u8 free;
-
- int rx_irq;
- int tx_irq;
-
- /* Protect the field .free, while checking if the mcbsp is in use */
- struct omap_mcbsp_platform_data *pdata;
- struct omap_mcbsp_st_data *st_data;
- struct omap_mcbsp_reg_cfg cfg_regs;
- struct omap_pcm_dma_data dma_data[2];
- int dma_op_mode;
- u16 max_tx_thres;
- u16 max_rx_thres;
- void *reg_cache;
- int reg_cache_size;
-
- unsigned int fmt;
- unsigned int in_freq;
- int clk_div;
- int wlen;
-};
-
-void omap_mcbsp_config(struct omap_mcbsp *mcbsp,
- const struct omap_mcbsp_reg_cfg *config);
-void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold);
-void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold);
-u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp);
-u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp);
-int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp);
-int omap_mcbsp_request(struct omap_mcbsp *mcbsp);
-void omap_mcbsp_free(struct omap_mcbsp *mcbsp);
-void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx);
-void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx);
-
-/* McBSP functional clock source changing function */
-int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id);
-
-/* McBSP signal muxing API */
-int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux);
-
-/* Sidetone specific API */
-int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain);
-int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain);
-int omap_st_enable(struct omap_mcbsp *mcbsp);
-int omap_st_disable(struct omap_mcbsp *mcbsp);
-int omap_st_is_enabled(struct omap_mcbsp *mcbsp);
-
-int __devinit omap_mcbsp_init(struct platform_device *pdev);
-void __devexit omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp);
-
-#endif /* __ASOC_MCBSP_H */
diff --git a/ANDROID_3.4.5/sound/soc/omap/n810.c b/ANDROID_3.4.5/sound/soc/omap/n810.c
deleted file mode 100644
index abac4b69..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/n810.c
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
- * n810.c -- SoC audio for Nokia N810
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/i2c.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <linux/gpio.h>
-#include <linux/module.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#define N810_HEADSET_AMP_GPIO 10
-#define N810_SPEAKER_AMP_GPIO 101
-
-enum {
- N810_JACK_DISABLED,
- N810_JACK_HP,
- N810_JACK_HS,
- N810_JACK_MIC,
-};
-
-static struct clk *sys_clkout2;
-static struct clk *sys_clkout2_src;
-static struct clk *func96m_clk;
-
-static int n810_spk_func;
-static int n810_jack_func;
-static int n810_dmic_func;
-
-static void n810_ext_control(struct snd_soc_dapm_context *dapm)
-{
- int hp = 0, line1l = 0;
-
- switch (n810_jack_func) {
- case N810_JACK_HS:
- line1l = 1;
- case N810_JACK_HP:
- hp = 1;
- break;
- case N810_JACK_MIC:
- line1l = 1;
- break;
- }
-
- if (n810_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
-
- if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- if (line1l)
- snd_soc_dapm_enable_pin(dapm, "LINE1L");
- else
- snd_soc_dapm_disable_pin(dapm, "LINE1L");
-
- if (n810_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
- else
- snd_soc_dapm_disable_pin(dapm, "DMic");
-
- snd_soc_dapm_sync(dapm);
-}
-
-static int n810_startup(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
-
- n810_ext_control(&codec->dapm);
- return clk_enable(sys_clkout2);
-}
-
-static void n810_shutdown(struct snd_pcm_substream *substream)
-{
- clk_disable(sys_clkout2);
-}
-
-static int n810_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int err;
-
- /* Set the codec system clock for DAC and ADC */
- err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
- SND_SOC_CLOCK_IN);
-
- return err;
-}
-
-static struct snd_soc_ops n810_ops = {
- .startup = n810_startup,
- .hw_params = n810_hw_params,
- .shutdown = n810_shutdown,
-};
-
-static int n810_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = n810_spk_func;
-
- return 0;
-}
-
-static int n810_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (n810_spk_func == ucontrol->value.integer.value[0])
- return 0;
-
- n810_spk_func = ucontrol->value.integer.value[0];
- n810_ext_control(&card->dapm);
-
- return 1;
-}
-
-static int n810_get_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = n810_jack_func;
-
- return 0;
-}
-
-static int n810_set_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (n810_jack_func == ucontrol->value.integer.value[0])
- return 0;
-
- n810_jack_func = ucontrol->value.integer.value[0];
- n810_ext_control(&card->dapm);
-
- return 1;
-}
-
-static int n810_get_input(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = n810_dmic_func;
-
- return 0;
-}
-
-static int n810_set_input(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (n810_dmic_func == ucontrol->value.integer.value[0])
- return 0;
-
- n810_dmic_func = ucontrol->value.integer.value[0];
- n810_ext_control(&card->dapm);
-
- return 1;
-}
-
-static int n810_spk_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- if (SND_SOC_DAPM_EVENT_ON(event))
- gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
- else
- gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
-
- return 0;
-}
-
-static int n810_jack_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- if (SND_SOC_DAPM_EVENT_ON(event))
- gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
- else
- gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
-
- return 0;
-}
-
-static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
- SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
- SND_SOC_DAPM_MIC("DMic", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- {"Headphone Jack", NULL, "HPLOUT"},
- {"Headphone Jack", NULL, "HPROUT"},
-
- {"Ext Spk", NULL, "LLOUT"},
- {"Ext Spk", NULL, "RLOUT"},
-
- {"DMic Rate 64", NULL, "Mic Bias 2V"},
- {"Mic Bias 2V", NULL, "DMic"},
-};
-
-static const char *spk_function[] = {"Off", "On"};
-static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
-static const char *input_function[] = {"ADC", "Digital Mic"};
-static const struct soc_enum n810_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
-};
-
-static const struct snd_kcontrol_new aic33_n810_controls[] = {
- SOC_ENUM_EXT("Speaker Function", n810_enum[0],
- n810_get_spk, n810_set_spk),
- SOC_ENUM_EXT("Jack Function", n810_enum[1],
- n810_get_jack, n810_set_jack),
- SOC_ENUM_EXT("Input Select", n810_enum[2],
- n810_get_input, n810_set_input),
-};
-
-static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* Not connected */
- snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
- snd_soc_dapm_nc_pin(dapm, "HPLCOM");
- snd_soc_dapm_nc_pin(dapm, "HPRCOM");
- snd_soc_dapm_nc_pin(dapm, "MIC3L");
- snd_soc_dapm_nc_pin(dapm, "MIC3R");
- snd_soc_dapm_nc_pin(dapm, "LINE1R");
- snd_soc_dapm_nc_pin(dapm, "LINE2L");
- snd_soc_dapm_nc_pin(dapm, "LINE2R");
-
- return 0;
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link n810_dai = {
- .name = "TLV320AIC33",
- .stream_name = "AIC33",
- .cpu_dai_name = "omap-mcbsp.2",
- .platform_name = "omap-pcm-audio",
- .codec_name = "tlv320aic3x-codec.2-0018",
- .codec_dai_name = "tlv320aic3x-hifi",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = n810_aic33_init,
- .ops = &n810_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_n810 = {
- .name = "N810",
- .owner = THIS_MODULE,
- .dai_link = &n810_dai,
- .num_links = 1,
-
- .controls = aic33_n810_controls,
- .num_controls = ARRAY_SIZE(aic33_n810_controls),
- .dapm_widgets = aic33_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *n810_snd_device;
-
-static int __init n810_soc_init(void)
-{
- int err;
- struct device *dev;
-
- if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
- return -ENODEV;
-
- n810_snd_device = platform_device_alloc("soc-audio", -1);
- if (!n810_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(n810_snd_device, &snd_soc_n810);
- err = platform_device_add(n810_snd_device);
- if (err)
- goto err1;
-
- dev = &n810_snd_device->dev;
-
- sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
- if (IS_ERR(sys_clkout2_src)) {
- dev_err(dev, "Could not get sys_clkout2_src clock\n");
- err = PTR_ERR(sys_clkout2_src);
- goto err2;
- }
- sys_clkout2 = clk_get(dev, "sys_clkout2");
- if (IS_ERR(sys_clkout2)) {
- dev_err(dev, "Could not get sys_clkout2\n");
- err = PTR_ERR(sys_clkout2);
- goto err3;
- }
- /*
- * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
- * 96 MHz as its parent in order to get 12 MHz
- */
- func96m_clk = clk_get(dev, "func_96m_ck");
- if (IS_ERR(func96m_clk)) {
- dev_err(dev, "Could not get func 96M clock\n");
- err = PTR_ERR(func96m_clk);
- goto err4;
- }
- clk_set_parent(sys_clkout2_src, func96m_clk);
- clk_set_rate(sys_clkout2, 12000000);
-
- BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
- (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
-
- gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
- gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
-
- return 0;
-err4:
- clk_put(sys_clkout2);
-err3:
- clk_put(sys_clkout2_src);
-err2:
- platform_device_del(n810_snd_device);
-err1:
- platform_device_put(n810_snd_device);
-
- return err;
-}
-
-static void __exit n810_soc_exit(void)
-{
- gpio_free(N810_SPEAKER_AMP_GPIO);
- gpio_free(N810_HEADSET_AMP_GPIO);
- clk_put(sys_clkout2_src);
- clk_put(sys_clkout2);
- clk_put(func96m_clk);
-
- platform_device_unregister(n810_snd_device);
-}
-
-module_init(n810_soc_init);
-module_exit(n810_soc_exit);
-
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
-MODULE_DESCRIPTION("ALSA SoC Nokia N810");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-abe-twl6040.c b/ANDROID_3.4.5/sound/soc/omap/omap-abe-twl6040.c
deleted file mode 100644
index 93bb8eee..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-abe-twl6040.c
+++ /dev/null
@@ -1,349 +0,0 @@
-/*
- * omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
- * twl6040 codec
- *
- * Author: Misael Lopez Cruz <misael.lopez@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/mfd/twl6040.h>
-#include <linux/platform_data/omap-abe-twl6040.h>
-#include <linux/module.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <plat/hardware.h>
-#include <plat/mux.h>
-
-#include "omap-dmic.h"
-#include "omap-mcpdm.h"
-#include "omap-pcm.h"
-#include "../codecs/twl6040.h"
-
-static int omap_abe_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_card *card = codec->card;
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
- int clk_id, freq;
- int ret;
-
- clk_id = twl6040_get_clk_id(rtd->codec);
- if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
- freq = pdata->mclk_freq;
- else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
- freq = 32768;
- else
- return -EINVAL;
-
- /* set the codec mclk */
- ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
- SND_SOC_CLOCK_IN);
- if (ret) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
- return ret;
-}
-
-static struct snd_soc_ops omap_abe_ops = {
- .hw_params = omap_abe_hw_params,
-};
-
-static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret = 0;
-
- ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
- 19200000, SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set DMIC cpu system clock\n");
- return ret;
- }
- ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
- SND_SOC_CLOCK_OUT);
- if (ret < 0) {
- printk(KERN_ERR "can't set DMIC output clock\n");
- return ret;
- }
- return 0;
-}
-
-static struct snd_soc_ops omap_abe_dmic_ops = {
- .hw_params = omap_abe_dmic_hw_params,
-};
-
-/* Headset jack */
-static struct snd_soc_jack hs_jack;
-
-/*Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
- {
- .pin = "Headset Stereophone",
- .mask = SND_JACK_HEADPHONE,
- },
-};
-
-/* SDP4430 machine DAPM */
-static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
- /* Outputs */
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
- SND_SOC_DAPM_SPK("Earphone Spk", NULL),
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
- SND_SOC_DAPM_LINE("Line Out", NULL),
- SND_SOC_DAPM_SPK("Vibrator", NULL),
-
- /* Inputs */
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
- SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
- SND_SOC_DAPM_LINE("Line In", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- /* Routings for outputs */
- {"Headset Stereophone", NULL, "HSOL"},
- {"Headset Stereophone", NULL, "HSOR"},
-
- {"Earphone Spk", NULL, "EP"},
-
- {"Ext Spk", NULL, "HFL"},
- {"Ext Spk", NULL, "HFR"},
-
- {"Line Out", NULL, "AUXL"},
- {"Line Out", NULL, "AUXR"},
-
- {"Vibrator", NULL, "VIBRAL"},
- {"Vibrator", NULL, "VIBRAR"},
-
- /* Routings for inputs */
- {"HSMIC", NULL, "Headset Mic"},
- {"Headset Mic", NULL, "Headset Mic Bias"},
-
- {"MAINMIC", NULL, "Main Handset Mic"},
- {"Main Handset Mic", NULL, "Main Mic Bias"},
-
- {"SUBMIC", NULL, "Sub Handset Mic"},
- {"Sub Handset Mic", NULL, "Main Mic Bias"},
-
- {"AFML", NULL, "Line In"},
- {"AFMR", NULL, "Line In"},
-};
-
-static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
- int connected, char *pin)
-{
- if (!connected)
- snd_soc_dapm_disable_pin(dapm, pin);
-}
-
-static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_card *card = codec->card;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
- int hs_trim;
- int ret = 0;
-
- /* Disable not connected paths if not used */
- twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
- twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
- twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
- twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
- twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
- twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
-
- /*
- * Configure McPDM offset cancellation based on the HSOTRIM value from
- * twl6040.
- */
- hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
- omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
- TWL6040_HSF_TRIM_RIGHT(hs_trim));
-
- /* Headset jack detection only if it is supported */
- if (pdata->jack_detection) {
- ret = snd_soc_jack_new(codec, "Headset Jack",
- SND_JACK_HEADSET, &hs_jack);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
- hs_jack_pins);
- twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
- }
-
- return ret;
-}
-
-static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Digital Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route dmic_audio_map[] = {
- {"DMic", NULL, "Digital Mic"},
- {"Digital Mic", NULL, "Digital Mic1 Bias"},
-};
-
-static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
- ARRAY_SIZE(dmic_dapm_widgets));
- if (ret)
- return ret;
-
- return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
- ARRAY_SIZE(dmic_audio_map));
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link twl6040_dmic_dai[] = {
- {
- .name = "TWL6040",
- .stream_name = "TWL6040",
- .cpu_dai_name = "omap-mcpdm",
- .codec_dai_name = "twl6040-legacy",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl6040-codec",
- .init = omap_abe_twl6040_init,
- .ops = &omap_abe_ops,
- },
- {
- .name = "DMIC",
- .stream_name = "DMIC Capture",
- .cpu_dai_name = "omap-dmic",
- .codec_dai_name = "dmic-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "dmic-codec",
- .init = omap_abe_dmic_init,
- .ops = &omap_abe_dmic_ops,
- },
-};
-
-static struct snd_soc_dai_link twl6040_only_dai[] = {
- {
- .name = "TWL6040",
- .stream_name = "TWL6040",
- .cpu_dai_name = "omap-mcpdm",
- .codec_dai_name = "twl6040-legacy",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl6040-codec",
- .init = omap_abe_twl6040_init,
- .ops = &omap_abe_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card omap_abe_card = {
- .owner = THIS_MODULE,
-
- .dapm_widgets = twl6040_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static __devinit int omap_abe_probe(struct platform_device *pdev)
-{
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
- struct snd_soc_card *card = &omap_abe_card;
- int ret;
-
- card->dev = &pdev->dev;
-
- if (!pdata) {
- dev_err(&pdev->dev, "Missing pdata\n");
- return -ENODEV;
- }
-
- if (pdata->card_name) {
- card->name = pdata->card_name;
- } else {
- dev_err(&pdev->dev, "Card name is not provided\n");
- return -ENODEV;
- }
-
- if (!pdata->mclk_freq) {
- dev_err(&pdev->dev, "MCLK frequency missing\n");
- return -ENODEV;
- }
-
- if (pdata->has_dmic) {
- card->dai_link = twl6040_dmic_dai;
- card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
- } else {
- card->dai_link = twl6040_only_dai;
- card->num_links = ARRAY_SIZE(twl6040_only_dai);
- }
-
- ret = snd_soc_register_card(card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
-
- return ret;
-}
-
-static int __devexit omap_abe_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
-}
-
-static struct platform_driver omap_abe_driver = {
- .driver = {
- .name = "omap-abe-twl6040",
- .owner = THIS_MODULE,
- .pm = &snd_soc_pm_ops,
- },
- .probe = omap_abe_probe,
- .remove = __devexit_p(omap_abe_remove),
-};
-
-module_platform_driver(omap_abe_driver);
-
-MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:omap-abe-twl6040");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-dmic.c b/ANDROID_3.4.5/sound/soc/omap/omap-dmic.c
deleted file mode 100644
index 4dcb5a7e..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-dmic.c
+++ /dev/null
@@ -1,545 +0,0 @@
-/*
- * omap-dmic.c -- OMAP ASoC DMIC DAI driver
- *
- * Copyright (C) 2010 - 2011 Texas Instruments
- *
- * Author: David Lambert <dlambert@ti.com>
- * Misael Lopez Cruz <misael.lopez@ti.com>
- * Liam Girdwood <lrg@ti.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/err.h>
-#include <linux/clk.h>
-#include <linux/io.h>
-#include <linux/slab.h>
-#include <linux/pm_runtime.h>
-#include <plat/dma.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include "omap-pcm.h"
-#include "omap-dmic.h"
-
-struct omap_dmic {
- struct device *dev;
- void __iomem *io_base;
- struct clk *fclk;
- int fclk_freq;
- int out_freq;
- int clk_div;
- int sysclk;
- int threshold;
- u32 ch_enabled;
- bool active;
- struct mutex mutex;
-};
-
-/*
- * Stream DMA parameters
- */
-static struct omap_pcm_dma_data omap_dmic_dai_dma_params = {
- .name = "DMIC capture",
- .data_type = OMAP_DMA_DATA_TYPE_S32,
- .sync_mode = OMAP_DMA_SYNC_PACKET,
-};
-
-static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val)
-{
- __raw_writel(val, dmic->io_base + reg);
-}
-
-static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg)
-{
- return __raw_readl(dmic->io_base + reg);
-}
-
-static inline void omap_dmic_start(struct omap_dmic *dmic)
-{
- u32 ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG);
-
- /* Configure DMA controller */
- omap_dmic_write(dmic, OMAP_DMIC_DMAENABLE_SET_REG,
- OMAP_DMIC_DMA_ENABLE);
-
- omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, ctrl | dmic->ch_enabled);
-}
-
-static inline void omap_dmic_stop(struct omap_dmic *dmic)
-{
- u32 ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG);
- omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG,
- ctrl & ~OMAP_DMIC_UP_ENABLE_MASK);
-
- /* Disable DMA request generation */
- omap_dmic_write(dmic, OMAP_DMIC_DMAENABLE_CLR_REG,
- OMAP_DMIC_DMA_ENABLE);
-
-}
-
-static inline int dmic_is_enabled(struct omap_dmic *dmic)
-{
- return omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG) &
- OMAP_DMIC_UP_ENABLE_MASK;
-}
-
-static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
- int ret = 0;
-
- mutex_lock(&dmic->mutex);
-
- if (!dai->active)
- dmic->active = 1;
- else
- ret = -EBUSY;
-
- mutex_unlock(&dmic->mutex);
-
- return ret;
-}
-
-static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
-
- mutex_lock(&dmic->mutex);
-
- if (!dai->active)
- dmic->active = 0;
-
- mutex_unlock(&dmic->mutex);
-}
-
-static int omap_dmic_select_divider(struct omap_dmic *dmic, int sample_rate)
-{
- int divider = -EINVAL;
-
- /*
- * 192KHz rate is only supported with 19.2MHz/3.84MHz clock
- * configuration.
- */
- if (sample_rate == 192000) {
- if (dmic->fclk_freq == 19200000 && dmic->out_freq == 3840000)
- divider = 0x6; /* Divider: 5 (192KHz sampling rate) */
- else
- dev_err(dmic->dev,
- "invalid clock configuration for 192KHz\n");
-
- return divider;
- }
-
- switch (dmic->out_freq) {
- case 1536000:
- if (dmic->fclk_freq != 24576000)
- goto div_err;
- divider = 0x4; /* Divider: 16 */
- break;
- case 2400000:
- switch (dmic->fclk_freq) {
- case 12000000:
- divider = 0x5; /* Divider: 5 */
- break;
- case 19200000:
- divider = 0x0; /* Divider: 8 */
- break;
- case 24000000:
- divider = 0x2; /* Divider: 10 */
- break;
- default:
- goto div_err;
- }
- break;
- case 3072000:
- if (dmic->fclk_freq != 24576000)
- goto div_err;
- divider = 0x3; /* Divider: 8 */
- break;
- case 3840000:
- if (dmic->fclk_freq != 19200000)
- goto div_err;
- divider = 0x1; /* Divider: 5 (96KHz sampling rate) */
- break;
- default:
- dev_err(dmic->dev, "invalid out frequency: %dHz\n",
- dmic->out_freq);
- break;
- }
-
- return divider;
-
-div_err:
- dev_err(dmic->dev, "invalid out frequency %dHz for %dHz input\n",
- dmic->out_freq, dmic->fclk_freq);
- return -EINVAL;
-}
-
-static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
- int channels;
-
- dmic->clk_div = omap_dmic_select_divider(dmic, params_rate(params));
- if (dmic->clk_div < 0) {
- dev_err(dmic->dev, "no valid divider for %dHz from %dHz\n",
- dmic->out_freq, dmic->fclk_freq);
- return -EINVAL;
- }
-
- dmic->ch_enabled = 0;
- channels = params_channels(params);
- switch (channels) {
- case 6:
- dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
- case 4:
- dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
- case 2:
- dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
- break;
- default:
- dev_err(dmic->dev, "invalid number of legacy channels\n");
- return -EINVAL;
- }
-
- /* packet size is threshold * channels */
- omap_dmic_dai_dma_params.packet_size = dmic->threshold * channels;
- snd_soc_dai_set_dma_data(dai, substream, &omap_dmic_dai_dma_params);
-
- return 0;
-}
-
-static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
- u32 ctrl;
-
- /* Configure uplink threshold */
- omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
-
- ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG);
-
- /* Set dmic out format */
- ctrl &= ~(OMAP_DMIC_FORMAT | OMAP_DMIC_POLAR_MASK);
- ctrl |= (OMAP_DMICOUTFORMAT_LJUST | OMAP_DMIC_POLAR1 |
- OMAP_DMIC_POLAR2 | OMAP_DMIC_POLAR3);
-
- /* Configure dmic clock divider */
- ctrl &= ~OMAP_DMIC_CLK_DIV_MASK;
- ctrl |= OMAP_DMIC_CLK_DIV(dmic->clk_div);
-
- omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, ctrl);
-
- omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG,
- ctrl | OMAP_DMICOUTFORMAT_LJUST | OMAP_DMIC_POLAR1 |
- OMAP_DMIC_POLAR2 | OMAP_DMIC_POLAR3);
-
- return 0;
-}
-
-static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream,
- int cmd, struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- omap_dmic_start(dmic);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- omap_dmic_stop(dmic);
- break;
- default:
- break;
- }
-
- return 0;
-}
-
-static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
- unsigned int freq)
-{
- struct clk *parent_clk;
- char *parent_clk_name;
- int ret = 0;
-
- switch (freq) {
- case 12000000:
- case 19200000:
- case 24000000:
- case 24576000:
- break;
- default:
- dev_err(dmic->dev, "invalid input frequency: %dHz\n", freq);
- dmic->fclk_freq = 0;
- return -EINVAL;
- }
-
- if (dmic->sysclk == clk_id) {
- dmic->fclk_freq = freq;
- return 0;
- }
-
- /* re-parent not allowed if a stream is ongoing */
- if (dmic->active && dmic_is_enabled(dmic)) {
- dev_err(dmic->dev, "can't re-parent when DMIC active\n");
- return -EBUSY;
- }
-
- switch (clk_id) {
- case OMAP_DMIC_SYSCLK_PAD_CLKS:
- parent_clk_name = "pad_clks_ck";
- break;
- case OMAP_DMIC_SYSCLK_SLIMBLUS_CLKS:
- parent_clk_name = "slimbus_clk";
- break;
- case OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS:
- parent_clk_name = "dmic_sync_mux_ck";
- break;
- default:
- dev_err(dmic->dev, "fclk clk_id (%d) not supported\n", clk_id);
- return -EINVAL;
- }
-
- parent_clk = clk_get(dmic->dev, parent_clk_name);
- if (IS_ERR(parent_clk)) {
- dev_err(dmic->dev, "can't get %s\n", parent_clk_name);
- return -ENODEV;
- }
-
- mutex_lock(&dmic->mutex);
- if (dmic->active) {
- /* disable clock while reparenting */
- pm_runtime_put_sync(dmic->dev);
- ret = clk_set_parent(dmic->fclk, parent_clk);
- pm_runtime_get_sync(dmic->dev);
- } else {
- ret = clk_set_parent(dmic->fclk, parent_clk);
- }
- mutex_unlock(&dmic->mutex);
-
- if (ret < 0) {
- dev_err(dmic->dev, "re-parent failed\n");
- goto err_busy;
- }
-
- dmic->sysclk = clk_id;
- dmic->fclk_freq = freq;
-
-err_busy:
- clk_put(parent_clk);
-
- return ret;
-}
-
-static int omap_dmic_select_outclk(struct omap_dmic *dmic, int clk_id,
- unsigned int freq)
-{
- int ret = 0;
-
- if (clk_id != OMAP_DMIC_ABE_DMIC_CLK) {
- dev_err(dmic->dev, "output clk_id (%d) not supported\n",
- clk_id);
- return -EINVAL;
- }
-
- switch (freq) {
- case 1536000:
- case 2400000:
- case 3072000:
- case 3840000:
- dmic->out_freq = freq;
- break;
- default:
- dev_err(dmic->dev, "invalid out frequency: %dHz\n", freq);
- dmic->out_freq = 0;
- ret = -EINVAL;
- }
-
- return ret;
-}
-
-static int omap_dmic_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
- unsigned int freq, int dir)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
-
- if (dir == SND_SOC_CLOCK_IN)
- return omap_dmic_select_fclk(dmic, clk_id, freq);
- else if (dir == SND_SOC_CLOCK_OUT)
- return omap_dmic_select_outclk(dmic, clk_id, freq);
-
- dev_err(dmic->dev, "invalid clock direction (%d)\n", dir);
- return -EINVAL;
-}
-
-static const struct snd_soc_dai_ops omap_dmic_dai_ops = {
- .startup = omap_dmic_dai_startup,
- .shutdown = omap_dmic_dai_shutdown,
- .hw_params = omap_dmic_dai_hw_params,
- .prepare = omap_dmic_dai_prepare,
- .trigger = omap_dmic_dai_trigger,
- .set_sysclk = omap_dmic_set_dai_sysclk,
-};
-
-static int omap_dmic_probe(struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
-
- pm_runtime_enable(dmic->dev);
-
- /* Disable lines while request is ongoing */
- pm_runtime_get_sync(dmic->dev);
- omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, 0x00);
- pm_runtime_put_sync(dmic->dev);
-
- /* Configure DMIC threshold value */
- dmic->threshold = OMAP_DMIC_THRES_MAX - 3;
- return 0;
-}
-
-static int omap_dmic_remove(struct snd_soc_dai *dai)
-{
- struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
-
- pm_runtime_disable(dmic->dev);
-
- return 0;
-}
-
-static struct snd_soc_dai_driver omap_dmic_dai = {
- .name = "omap-dmic",
- .probe = omap_dmic_probe,
- .remove = omap_dmic_remove,
- .capture = {
- .channels_min = 2,
- .channels_max = 6,
- .rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
- .formats = SNDRV_PCM_FMTBIT_S32_LE,
- .sig_bits = 24,
- },
- .ops = &omap_dmic_dai_ops,
-};
-
-static __devinit int asoc_dmic_probe(struct platform_device *pdev)
-{
- struct omap_dmic *dmic;
- struct resource *res;
- int ret;
-
- dmic = devm_kzalloc(&pdev->dev, sizeof(struct omap_dmic), GFP_KERNEL);
- if (!dmic)
- return -ENOMEM;
-
- platform_set_drvdata(pdev, dmic);
- dmic->dev = &pdev->dev;
- dmic->sysclk = OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS;
-
- mutex_init(&dmic->mutex);
-
- dmic->fclk = clk_get(dmic->dev, "dmic_fck");
- if (IS_ERR(dmic->fclk)) {
- dev_err(dmic->dev, "cant get dmic_fck\n");
- return -ENODEV;
- }
-
- res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma");
- if (!res) {
- dev_err(dmic->dev, "invalid dma memory resource\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
- omap_dmic_dai_dma_params.port_addr = res->start + OMAP_DMIC_DATA_REG;
-
- res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!res) {
- dev_err(dmic->dev, "invalid dma resource\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
- omap_dmic_dai_dma_params.dma_req = res->start;
-
- res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
- if (!res) {
- dev_err(dmic->dev, "invalid memory resource\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
-
- if (!devm_request_mem_region(&pdev->dev, res->start,
- resource_size(res), pdev->name)) {
- dev_err(dmic->dev, "memory region already claimed\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
-
- dmic->io_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!dmic->io_base) {
- ret = -ENOMEM;
- goto err_put_clk;
- }
-
- ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai);
- if (ret)
- goto err_put_clk;
-
- return 0;
-
-err_put_clk:
- clk_put(dmic->fclk);
- return ret;
-}
-
-static int __devexit asoc_dmic_remove(struct platform_device *pdev)
-{
- struct omap_dmic *dmic = platform_get_drvdata(pdev);
-
- snd_soc_unregister_dai(&pdev->dev);
- clk_put(dmic->fclk);
-
- return 0;
-}
-
-static struct platform_driver asoc_dmic_driver = {
- .driver = {
- .name = "omap-dmic",
- .owner = THIS_MODULE,
- },
- .probe = asoc_dmic_probe,
- .remove = __devexit_p(asoc_dmic_remove),
-};
-
-module_platform_driver(asoc_dmic_driver);
-
-MODULE_ALIAS("platform:omap-dmic");
-MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
-MODULE_DESCRIPTION("OMAP DMIC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-dmic.h b/ANDROID_3.4.5/sound/soc/omap/omap-dmic.h
deleted file mode 100644
index 231e728b..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-dmic.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * omap-dmic.h -- OMAP Digital Microphone Controller
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _OMAP_DMIC_H
-#define _OMAP_DMIC_H
-
-#define OMAP_DMIC_REVISION_REG 0x00
-#define OMAP_DMIC_SYSCONFIG_REG 0x10
-#define OMAP_DMIC_IRQSTATUS_RAW_REG 0x24
-#define OMAP_DMIC_IRQSTATUS_REG 0x28
-#define OMAP_DMIC_IRQENABLE_SET_REG 0x2C
-#define OMAP_DMIC_IRQENABLE_CLR_REG 0x30
-#define OMAP_DMIC_IRQWAKE_EN_REG 0x34
-#define OMAP_DMIC_DMAENABLE_SET_REG 0x38
-#define OMAP_DMIC_DMAENABLE_CLR_REG 0x3C
-#define OMAP_DMIC_DMAWAKEEN_REG 0x40
-#define OMAP_DMIC_CTRL_REG 0x44
-#define OMAP_DMIC_DATA_REG 0x48
-#define OMAP_DMIC_FIFO_CTRL_REG 0x4C
-#define OMAP_DMIC_FIFO_DMIC1R_DATA_REG 0x50
-#define OMAP_DMIC_FIFO_DMIC1L_DATA_REG 0x54
-#define OMAP_DMIC_FIFO_DMIC2R_DATA_REG 0x58
-#define OMAP_DMIC_FIFO_DMIC2L_DATA_REG 0x5C
-#define OMAP_DMIC_FIFO_DMIC3R_DATA_REG 0x60
-#define OMAP_DMIC_FIFO_DMIC3L_DATA_REG 0x64
-
-/* IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR bit fields */
-#define OMAP_DMIC_IRQ (1 << 0)
-#define OMAP_DMIC_IRQ_FULL (1 << 1)
-#define OMAP_DMIC_IRQ_ALMST_EMPTY (1 << 2)
-#define OMAP_DMIC_IRQ_EMPTY (1 << 3)
-#define OMAP_DMIC_IRQ_MASK 0x07
-
-/* DMIC_DMAENABLE bit fields */
-#define OMAP_DMIC_DMA_ENABLE 0x1
-
-/* DMIC_CTRL bit fields */
-#define OMAP_DMIC_UP1_ENABLE (1 << 0)
-#define OMAP_DMIC_UP2_ENABLE (1 << 1)
-#define OMAP_DMIC_UP3_ENABLE (1 << 2)
-#define OMAP_DMIC_UP_ENABLE_MASK 0x7
-#define OMAP_DMIC_FORMAT (1 << 3)
-#define OMAP_DMIC_POLAR1 (1 << 4)
-#define OMAP_DMIC_POLAR2 (1 << 5)
-#define OMAP_DMIC_POLAR3 (1 << 6)
-#define OMAP_DMIC_POLAR_MASK (0x7 << 4)
-#define OMAP_DMIC_CLK_DIV(x) (((x) & 0x7) << 7)
-#define OMAP_DMIC_CLK_DIV_MASK (0x7 << 7)
-#define OMAP_DMIC_RESET (1 << 10)
-
-#define OMAP_DMICOUTFORMAT_LJUST (0 << 3)
-#define OMAP_DMICOUTFORMAT_RJUST (1 << 3)
-
-/* DMIC_FIFO_CTRL bit fields */
-#define OMAP_DMIC_THRES_MAX 0xF
-
-enum omap_dmic_clk {
- OMAP_DMIC_SYSCLK_PAD_CLKS, /* PAD_CLKS */
- OMAP_DMIC_SYSCLK_SLIMBLUS_CLKS, /* SLIMBUS_CLK */
- OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS, /* DMIC_SYNC_MUX_CLK */
- OMAP_DMIC_ABE_DMIC_CLK, /* abe_dmic_clk */
-};
-
-#endif
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-hdmi.c b/ANDROID_3.4.5/sound/soc/omap/omap-hdmi.c
deleted file mode 100644
index 38e0defa..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-hdmi.c
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * omap-hdmi.c
- *
- * OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors.
- * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/
- * Authors: Jorge Candelaria <jorge.candelaria@ti.com>
- * Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include <plat/dma.h>
-#include "omap-pcm.h"
-#include "omap-hdmi.h"
-
-#define DRV_NAME "hdmi-audio-dai"
-
-static struct omap_pcm_dma_data omap_hdmi_dai_dma_params = {
- .name = "HDMI playback",
- .sync_mode = OMAP_DMA_SYNC_PACKET,
-};
-
-static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- int err;
- /*
- * Make sure that the period bytes are multiple of the DMA packet size.
- * Largest packet size we use is 32 32-bit words = 128 bytes
- */
- err = snd_pcm_hw_constraint_step(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128);
- if (err < 0)
- return err;
-
- return 0;
-}
-
-static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
-{
- int err = 0;
-
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- omap_hdmi_dai_dma_params.packet_size = 16;
- break;
- case SNDRV_PCM_FORMAT_S24_LE:
- omap_hdmi_dai_dma_params.packet_size = 32;
- break;
- default:
- err = -EINVAL;
- }
-
- omap_hdmi_dai_dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
-
- snd_soc_dai_set_dma_data(dai, substream,
- &omap_hdmi_dai_dma_params);
-
- return err;
-}
-
-static const struct snd_soc_dai_ops omap_hdmi_dai_ops = {
- .startup = omap_hdmi_dai_startup,
- .hw_params = omap_hdmi_dai_hw_params,
-};
-
-static struct snd_soc_dai_driver omap_hdmi_dai = {
- .playback = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = OMAP_HDMI_RATES,
- .formats = OMAP_HDMI_FORMATS,
- },
- .ops = &omap_hdmi_dai_ops,
-};
-
-static __devinit int omap_hdmi_probe(struct platform_device *pdev)
-{
- int ret;
- struct resource *hdmi_rsrc;
-
- hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (!hdmi_rsrc) {
- dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n");
- return -EINVAL;
- }
-
- omap_hdmi_dai_dma_params.port_addr = hdmi_rsrc->start
- + OMAP_HDMI_AUDIO_DMA_PORT;
-
- hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!hdmi_rsrc) {
- dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n");
- return -EINVAL;
- }
-
- omap_hdmi_dai_dma_params.dma_req = hdmi_rsrc->start;
-
- ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai);
- return ret;
-}
-
-static int __devexit omap_hdmi_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_dai(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver hdmi_dai_driver = {
- .driver = {
- .name = DRV_NAME,
- .owner = THIS_MODULE,
- },
- .probe = omap_hdmi_probe,
- .remove = __devexit_p(omap_hdmi_remove),
-};
-
-module_platform_driver(hdmi_dai_driver);
-
-MODULE_AUTHOR("Jorge Candelaria <jorge.candelaria@ti.com>");
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("OMAP HDMI SoC Interface");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-hdmi.h b/ANDROID_3.4.5/sound/soc/omap/omap-hdmi.h
deleted file mode 100644
index 34c298d5..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-hdmi.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * omap-hdmi.h
- *
- * Definitions for OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors.
- * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/
- * Authors: Jorge Candelaria <jorge.candelaria@ti.com>
- * Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#ifndef __OMAP_HDMI_H__
-#define __OMAP_HDMI_H__
-
-#define OMAP_HDMI_AUDIO_DMA_PORT 0x8c
-
-#define OMAP_HDMI_RATES (SNDRV_PCM_RATE_32000 | \
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-
-#define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
- SNDRV_PCM_FMTBIT_S24_LE)
-
-#endif
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.c b/ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.c
deleted file mode 100644
index 6912ac7c..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.c
+++ /dev/null
@@ -1,817 +0,0 @@
-/*
- * omap-mcbsp.c -- OMAP ALSA SoC DAI driver using McBSP port
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
-#include <linux/pm_runtime.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include <plat/dma.h>
-#include <plat/mcbsp.h>
-#include "mcbsp.h"
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000)
-
-#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \
- xhandler_get, xhandler_put) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = omap_mcbsp_st_info_volsw, \
- .get = xhandler_get, .put = xhandler_put, \
- .private_value = (unsigned long) &(struct soc_mixer_control) \
- {.min = xmin, .max = xmax} }
-
-enum {
- OMAP_MCBSP_WORD_8 = 0,
- OMAP_MCBSP_WORD_12,
- OMAP_MCBSP_WORD_16,
- OMAP_MCBSP_WORD_20,
- OMAP_MCBSP_WORD_24,
- OMAP_MCBSP_WORD_32,
-};
-
-/*
- * Stream DMA parameters. DMA request line and port address are set runtime
- * since they are different between OMAP1 and later OMAPs
- */
-static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_pcm_dma_data *dma_data;
- int words;
-
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
- if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
- /*
- * Configure McBSP threshold based on either:
- * packet_size, when the sDMA is in packet mode, or
- * based on the period size.
- */
- if (dma_data->packet_size)
- words = dma_data->packet_size;
- else
- words = snd_pcm_lib_period_bytes(substream) /
- (mcbsp->wlen / 8);
- else
- words = 1;
-
- /* Configure McBSP internal buffer usage */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- omap_mcbsp_set_tx_threshold(mcbsp, words);
- else
- omap_mcbsp_set_rx_threshold(mcbsp, words);
-}
-
-static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- struct snd_interval *buffer_size = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE);
- struct snd_interval *channels = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
- struct omap_mcbsp *mcbsp = rule->private;
- struct snd_interval frames;
- int size;
-
- snd_interval_any(&frames);
- size = mcbsp->pdata->buffer_size;
-
- frames.min = size / channels->min;
- frames.integer = 1;
- return snd_interval_refine(buffer_size, &frames);
-}
-
-static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- int err = 0;
-
- if (!cpu_dai->active)
- err = omap_mcbsp_request(mcbsp);
-
- /*
- * OMAP3 McBSP FIFO is word structured.
- * McBSP2 has 1024 + 256 = 1280 word long buffer,
- * McBSP1,3,4,5 has 128 word long buffer
- * This means that the size of the FIFO depends on the sample format.
- * For example on McBSP3:
- * 16bit samples: size is 128 * 2 = 256 bytes
- * 32bit samples: size is 128 * 4 = 512 bytes
- * It is simpler to place constraint for buffer and period based on
- * channels.
- * McBSP3 as example again (16 or 32 bit samples):
- * 1 channel (mono): size is 128 frames (128 words)
- * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
- * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
- */
- if (mcbsp->pdata->buffer_size) {
- /*
- * Rule for the buffer size. We should not allow
- * smaller buffer than the FIFO size to avoid underruns
- */
- snd_pcm_hw_rule_add(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- omap_mcbsp_hwrule_min_buffersize,
- mcbsp,
- SNDRV_PCM_HW_PARAM_CHANNELS, -1);
-
- /* Make sure, that the period size is always even */
- snd_pcm_hw_constraint_step(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
- }
-
- return err;
-}
-
-static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
-
- if (!cpu_dai->active) {
- omap_mcbsp_free(mcbsp);
- mcbsp->configured = 0;
- }
-}
-
-static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
- struct snd_soc_dai *cpu_dai)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- mcbsp->active++;
- omap_mcbsp_start(mcbsp, play, !play);
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- omap_mcbsp_stop(mcbsp, play, !play);
- mcbsp->active--;
- break;
- default:
- err = -EINVAL;
- }
-
- return err;
-}
-
-static snd_pcm_sframes_t omap_mcbsp_dai_delay(
- struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- u16 fifo_use;
- snd_pcm_sframes_t delay;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- fifo_use = omap_mcbsp_get_tx_delay(mcbsp);
- else
- fifo_use = omap_mcbsp_get_rx_delay(mcbsp);
-
- /*
- * Divide the used locations with the channel count to get the
- * FIFO usage in samples (don't care about partial samples in the
- * buffer).
- */
- delay = fifo_use / substream->runtime->channels;
-
- return delay;
-}
-
-static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
- struct omap_pcm_dma_data *dma_data;
- int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
- int pkt_size = 0;
- unsigned int format, div, framesize, master;
-
- dma_data = &mcbsp->dma_data[substream->stream];
-
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- dma_data->data_type = OMAP_DMA_DATA_TYPE_S16;
- wlen = 16;
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- dma_data->data_type = OMAP_DMA_DATA_TYPE_S32;
- wlen = 32;
- break;
- default:
- return -EINVAL;
- }
- if (mcbsp->pdata->buffer_size) {
- dma_data->set_threshold = omap_mcbsp_set_threshold;
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
- if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
- int period_words, max_thrsh;
-
- period_words = params_period_bytes(params) / (wlen / 8);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- max_thrsh = mcbsp->max_tx_thres;
- else
- max_thrsh = mcbsp->max_rx_thres;
- /*
- * If the period contains less or equal number of words,
- * we are using the original threshold mode setup:
- * McBSP threshold = sDMA frame size = period_size
- * Otherwise we switch to sDMA packet mode:
- * McBSP threshold = sDMA packet size
- * sDMA frame size = period size
- */
- if (period_words > max_thrsh) {
- int divider = 0;
-
- /*
- * Look for the biggest threshold value, which
- * divides the period size evenly.
- */
- divider = period_words / max_thrsh;
- if (period_words % max_thrsh)
- divider++;
- while (period_words % divider &&
- divider < period_words)
- divider++;
- if (divider == period_words)
- return -EINVAL;
-
- pkt_size = period_words / divider;
- sync_mode = OMAP_DMA_SYNC_PACKET;
- } else {
- sync_mode = OMAP_DMA_SYNC_FRAME;
- }
- }
- }
-
- dma_data->sync_mode = sync_mode;
- dma_data->packet_size = pkt_size;
-
- snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
-
- if (mcbsp->configured) {
- /* McBSP already configured by another stream */
- return 0;
- }
-
- regs->rcr2 &= ~(RPHASE | RFRLEN2(0x7f) | RWDLEN2(7));
- regs->xcr2 &= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7));
- regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7));
- regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7));
- format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
- wpf = channels = params_channels(params);
- if (channels == 2 && (format == SND_SOC_DAIFMT_I2S ||
- format == SND_SOC_DAIFMT_LEFT_J)) {
- /* Use dual-phase frames */
- regs->rcr2 |= RPHASE;
- regs->xcr2 |= XPHASE;
- /* Set 1 word per (McBSP) frame for phase1 and phase2 */
- wpf--;
- regs->rcr2 |= RFRLEN2(wpf - 1);
- regs->xcr2 |= XFRLEN2(wpf - 1);
- }
-
- regs->rcr1 |= RFRLEN1(wpf - 1);
- regs->xcr1 |= XFRLEN1(wpf - 1);
-
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- /* Set word lengths */
- regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16);
- regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16);
- regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16);
- regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16);
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- /* Set word lengths */
- regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32);
- regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32);
- regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32);
- regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_32);
- break;
- default:
- /* Unsupported PCM format */
- return -EINVAL;
- }
-
- /* In McBSP master modes, FRAME (i.e. sample rate) is generated
- * by _counting_ BCLKs. Calculate frame size in BCLKs */
- master = mcbsp->fmt & SND_SOC_DAIFMT_MASTER_MASK;
- if (master == SND_SOC_DAIFMT_CBS_CFS) {
- div = mcbsp->clk_div ? mcbsp->clk_div : 1;
- framesize = (mcbsp->in_freq / div) / params_rate(params);
-
- if (framesize < wlen * channels) {
- printk(KERN_ERR "%s: not enough bandwidth for desired rate and "
- "channels\n", __func__);
- return -EINVAL;
- }
- } else
- framesize = wlen * channels;
-
- /* Set FS period and length in terms of bit clock periods */
- regs->srgr2 &= ~FPER(0xfff);
- regs->srgr1 &= ~FWID(0xff);
- switch (format) {
- case SND_SOC_DAIFMT_I2S:
- case SND_SOC_DAIFMT_LEFT_J:
- regs->srgr2 |= FPER(framesize - 1);
- regs->srgr1 |= FWID((framesize >> 1) - 1);
- break;
- case SND_SOC_DAIFMT_DSP_A:
- case SND_SOC_DAIFMT_DSP_B:
- regs->srgr2 |= FPER(framesize - 1);
- regs->srgr1 |= FWID(0);
- break;
- }
-
- omap_mcbsp_config(mcbsp, &mcbsp->cfg_regs);
- mcbsp->wlen = wlen;
- mcbsp->configured = 1;
-
- return 0;
-}
-
-/*
- * This must be called before _set_clkdiv and _set_sysclk since McBSP register
- * cache is initialized here
- */
-static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
- unsigned int fmt)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
- bool inv_fs = false;
-
- if (mcbsp->configured)
- return 0;
-
- mcbsp->fmt = fmt;
- memset(regs, 0, sizeof(*regs));
- /* Generic McBSP register settings */
- regs->spcr2 |= XINTM(3) | FREE;
- regs->spcr1 |= RINTM(3);
- /* RFIG and XFIG are not defined in 34xx */
- if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) {
- regs->rcr2 |= RFIG;
- regs->xcr2 |= XFIG;
- }
- if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) {
- regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
- regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
- }
-
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_I2S:
- /* 1-bit data delay */
- regs->rcr2 |= RDATDLY(1);
- regs->xcr2 |= XDATDLY(1);
- break;
- case SND_SOC_DAIFMT_LEFT_J:
- /* 0-bit data delay */
- regs->rcr2 |= RDATDLY(0);
- regs->xcr2 |= XDATDLY(0);
- regs->spcr1 |= RJUST(2);
- /* Invert FS polarity configuration */
- inv_fs = true;
- break;
- case SND_SOC_DAIFMT_DSP_A:
- /* 1-bit data delay */
- regs->rcr2 |= RDATDLY(1);
- regs->xcr2 |= XDATDLY(1);
- /* Invert FS polarity configuration */
- inv_fs = true;
- break;
- case SND_SOC_DAIFMT_DSP_B:
- /* 0-bit data delay */
- regs->rcr2 |= RDATDLY(0);
- regs->xcr2 |= XDATDLY(0);
- /* Invert FS polarity configuration */
- inv_fs = true;
- break;
- default:
- /* Unsupported data format */
- return -EINVAL;
- }
-
- switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
- /* McBSP master. Set FS and bit clocks as outputs */
- regs->pcr0 |= FSXM | FSRM |
- CLKXM | CLKRM;
- /* Sample rate generator drives the FS */
- regs->srgr2 |= FSGM;
- break;
- case SND_SOC_DAIFMT_CBM_CFM:
- /* McBSP slave */
- break;
- default:
- /* Unsupported master/slave configuration */
- return -EINVAL;
- }
-
- /* Set bit clock (CLKX/CLKR) and FS polarities */
- switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_NB_NF:
- /*
- * Normal BCLK + FS.
- * FS active low. TX data driven on falling edge of bit clock
- * and RX data sampled on rising edge of bit clock.
- */
- regs->pcr0 |= FSXP | FSRP |
- CLKXP | CLKRP;
- break;
- case SND_SOC_DAIFMT_NB_IF:
- regs->pcr0 |= CLKXP | CLKRP;
- break;
- case SND_SOC_DAIFMT_IB_NF:
- regs->pcr0 |= FSXP | FSRP;
- break;
- case SND_SOC_DAIFMT_IB_IF:
- break;
- default:
- return -EINVAL;
- }
- if (inv_fs == true)
- regs->pcr0 ^= FSXP | FSRP;
-
- return 0;
-}
-
-static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
- int div_id, int div)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
-
- if (div_id != OMAP_MCBSP_CLKGDV)
- return -ENODEV;
-
- mcbsp->clk_div = div;
- regs->srgr1 &= ~CLKGDV(0xff);
- regs->srgr1 |= CLKGDV(div - 1);
-
- return 0;
-}
-
-static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
- int clk_id, unsigned int freq,
- int dir)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
- int err = 0;
-
- if (mcbsp->active) {
- if (freq == mcbsp->in_freq)
- return 0;
- else
- return -EBUSY;
- }
-
- if (clk_id == OMAP_MCBSP_SYSCLK_CLK ||
- clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK ||
- clk_id == OMAP_MCBSP_SYSCLK_CLKS_EXT ||
- clk_id == OMAP_MCBSP_SYSCLK_CLKX_EXT ||
- clk_id == OMAP_MCBSP_SYSCLK_CLKR_EXT) {
- mcbsp->in_freq = freq;
- regs->srgr2 &= ~CLKSM;
- regs->pcr0 &= ~SCLKME;
- } else if (cpu_class_is_omap1()) {
- /*
- * McBSP CLKR/FSR signal muxing functions are only available on
- * OMAP2 or newer versions
- */
- return -EINVAL;
- }
-
- switch (clk_id) {
- case OMAP_MCBSP_SYSCLK_CLK:
- regs->srgr2 |= CLKSM;
- break;
- case OMAP_MCBSP_SYSCLK_CLKS_FCLK:
- if (cpu_class_is_omap1()) {
- err = -EINVAL;
- break;
- }
- err = omap2_mcbsp_set_clks_src(mcbsp,
- MCBSP_CLKS_PRCM_SRC);
- break;
- case OMAP_MCBSP_SYSCLK_CLKS_EXT:
- if (cpu_class_is_omap1()) {
- err = 0;
- break;
- }
- err = omap2_mcbsp_set_clks_src(mcbsp,
- MCBSP_CLKS_PAD_SRC);
- break;
-
- case OMAP_MCBSP_SYSCLK_CLKX_EXT:
- regs->srgr2 |= CLKSM;
- case OMAP_MCBSP_SYSCLK_CLKR_EXT:
- regs->pcr0 |= SCLKME;
- break;
-
-
- case OMAP_MCBSP_CLKR_SRC_CLKR:
- err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKR);
- break;
- case OMAP_MCBSP_CLKR_SRC_CLKX:
- err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKX);
- break;
- case OMAP_MCBSP_FSR_SRC_FSR:
- err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSR);
- break;
- case OMAP_MCBSP_FSR_SRC_FSX:
- err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSX);
- break;
- default:
- err = -ENODEV;
- }
-
- return err;
-}
-
-static const struct snd_soc_dai_ops mcbsp_dai_ops = {
- .startup = omap_mcbsp_dai_startup,
- .shutdown = omap_mcbsp_dai_shutdown,
- .trigger = omap_mcbsp_dai_trigger,
- .delay = omap_mcbsp_dai_delay,
- .hw_params = omap_mcbsp_dai_hw_params,
- .set_fmt = omap_mcbsp_dai_set_dai_fmt,
- .set_clkdiv = omap_mcbsp_dai_set_clkdiv,
- .set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
-};
-
-static int omap_mcbsp_probe(struct snd_soc_dai *dai)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai);
-
- pm_runtime_enable(mcbsp->dev);
-
- return 0;
-}
-
-static int omap_mcbsp_remove(struct snd_soc_dai *dai)
-{
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai);
-
- pm_runtime_disable(mcbsp->dev);
-
- return 0;
-}
-
-static struct snd_soc_dai_driver omap_mcbsp_dai = {
- .probe = omap_mcbsp_probe,
- .remove = omap_mcbsp_remove,
- .playback = {
- .channels_min = 1,
- .channels_max = 16,
- .rates = OMAP_MCBSP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
- },
- .capture = {
- .channels_min = 1,
- .channels_max = 16,
- .rates = OMAP_MCBSP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
- },
- .ops = &mcbsp_dai_ops,
-};
-
-static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- int max = mc->max;
- int min = mc->min;
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 1;
- uinfo->value.integer.min = min;
- uinfo->value.integer.max = max;
- return 0;
-}
-
-#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(channel) \
-static int \
-omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \
- struct snd_ctl_elem_value *uc) \
-{ \
- struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \
- struct soc_mixer_control *mc = \
- (struct soc_mixer_control *)kc->private_value; \
- int max = mc->max; \
- int min = mc->min; \
- int val = uc->value.integer.value[0]; \
- \
- if (val < min || val > max) \
- return -EINVAL; \
- \
- /* OMAP McBSP implementation uses index values 0..4 */ \
- return omap_st_set_chgain(mcbsp, channel, val); \
-}
-
-#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(channel) \
-static int \
-omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \
- struct snd_ctl_elem_value *uc) \
-{ \
- struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \
- s16 chgain; \
- \
- if (omap_st_get_chgain(mcbsp, channel, &chgain)) \
- return -EAGAIN; \
- \
- uc->value.integer.value[0] = chgain; \
- return 0; \
-}
-
-OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(0)
-OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(1)
-OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(0)
-OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(1)
-
-static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- u8 value = ucontrol->value.integer.value[0];
-
- if (value == omap_st_is_enabled(mcbsp))
- return 0;
-
- if (value)
- omap_st_enable(mcbsp);
- else
- omap_st_disable(mcbsp);
-
- return 1;
-}
-
-static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
-
- ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp);
- return 0;
-}
-
-static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = {
- SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0,
- omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
- OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume",
- -32768, 32767,
- omap_mcbsp_get_st_ch0_volume,
- omap_mcbsp_set_st_ch0_volume),
- OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume",
- -32768, 32767,
- omap_mcbsp_get_st_ch1_volume,
- omap_mcbsp_set_st_ch1_volume),
-};
-
-static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
- SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0,
- omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
- OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume",
- -32768, 32767,
- omap_mcbsp_get_st_ch0_volume,
- omap_mcbsp_set_st_ch0_volume),
- OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume",
- -32768, 32767,
- omap_mcbsp_get_st_ch1_volume,
- omap_mcbsp_set_st_ch1_volume),
-};
-
-int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
-
- if (!mcbsp->st_data)
- return -ENODEV;
-
- switch (cpu_dai->id) {
- case 2: /* McBSP 2 */
- return snd_soc_add_dai_controls(cpu_dai,
- omap_mcbsp2_st_controls,
- ARRAY_SIZE(omap_mcbsp2_st_controls));
- case 3: /* McBSP 3 */
- return snd_soc_add_dai_controls(cpu_dai,
- omap_mcbsp3_st_controls,
- ARRAY_SIZE(omap_mcbsp3_st_controls));
- default:
- break;
- }
-
- return -EINVAL;
-}
-EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls);
-
-static __devinit int asoc_mcbsp_probe(struct platform_device *pdev)
-{
- struct omap_mcbsp_platform_data *pdata = dev_get_platdata(&pdev->dev);
- struct omap_mcbsp *mcbsp;
- int ret;
-
- if (!pdata) {
- dev_err(&pdev->dev, "missing platform data.\n");
- return -EINVAL;
- }
- mcbsp = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcbsp), GFP_KERNEL);
- if (!mcbsp)
- return -ENOMEM;
-
- mcbsp->id = pdev->id;
- mcbsp->pdata = pdata;
- mcbsp->dev = &pdev->dev;
- platform_set_drvdata(pdev, mcbsp);
-
- ret = omap_mcbsp_init(pdev);
- if (!ret)
- return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai);
-
- return ret;
-}
-
-static int __devexit asoc_mcbsp_remove(struct platform_device *pdev)
-{
- struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
-
- snd_soc_unregister_dai(&pdev->dev);
-
- if (mcbsp->pdata->ops && mcbsp->pdata->ops->free)
- mcbsp->pdata->ops->free(mcbsp->id);
-
- omap_mcbsp_sysfs_remove(mcbsp);
-
- clk_put(mcbsp->fclk);
-
- platform_set_drvdata(pdev, NULL);
-
- return 0;
-}
-
-static struct platform_driver asoc_mcbsp_driver = {
- .driver = {
- .name = "omap-mcbsp",
- .owner = THIS_MODULE,
- },
-
- .probe = asoc_mcbsp_probe,
- .remove = __devexit_p(asoc_mcbsp_remove),
-};
-
-module_platform_driver(asoc_mcbsp_driver);
-
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
-MODULE_DESCRIPTION("OMAP I2S SoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.h b/ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.h
deleted file mode 100644
index f877b16f..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-mcbsp.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * omap-mcbsp.h
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#ifndef __OMAP_I2S_H__
-#define __OMAP_I2S_H__
-
-/* Source clocks for McBSP sample rate generator */
-enum omap_mcbsp_clksrg_clk {
- OMAP_MCBSP_SYSCLK_CLKS_FCLK, /* Internal FCLK */
- OMAP_MCBSP_SYSCLK_CLKS_EXT, /* External CLKS pin */
- OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
- OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
- OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
- OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */
- OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */
- OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */
- OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */
-};
-
-/* McBSP dividers */
-enum omap_mcbsp_div {
- OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */
-};
-
-#if defined(CONFIG_SOC_OMAP2420)
-#define NUM_LINKS 2
-#endif
-#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
-#undef NUM_LINKS
-#define NUM_LINKS 3
-#endif
-#if defined(CONFIG_ARCH_OMAP4)
-#undef NUM_LINKS
-#define NUM_LINKS 4
-#endif
-#if defined(CONFIG_ARCH_OMAP3) || defined(CONFIG_SOC_OMAP2430)
-#undef NUM_LINKS
-#define NUM_LINKS 5
-#endif
-
-int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd);
-
-#endif
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.c b/ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.c
deleted file mode 100644
index 39705561..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.c
+++ /dev/null
@@ -1,524 +0,0 @@
-/*
- * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port
- *
- * Copyright (C) 2009 - 2011 Texas Instruments
- *
- * Author: Misael Lopez Cruz <misael.lopez@ti.com>
- * Contact: Jorge Eduardo Candelaria <x0107209@ti.com>
- * Margarita Olaya <magi.olaya@ti.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/interrupt.h>
-#include <linux/err.h>
-#include <linux/io.h>
-#include <linux/irq.h>
-#include <linux/slab.h>
-#include <linux/pm_runtime.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <plat/dma.h>
-#include <plat/omap_hwmod.h>
-#include "omap-mcpdm.h"
-#include "omap-pcm.h"
-
-struct omap_mcpdm {
- struct device *dev;
- unsigned long phys_base;
- void __iomem *io_base;
- int irq;
-
- struct mutex mutex;
-
- /* channel data */
- u32 dn_channels;
- u32 up_channels;
-
- /* McPDM FIFO thresholds */
- u32 dn_threshold;
- u32 up_threshold;
-
- /* McPDM dn offsets for rx1, and 2 channels */
- u32 dn_rx_offset;
-};
-
-/*
- * Stream DMA parameters
- */
-static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = {
- {
- .name = "Audio playback",
- .dma_req = OMAP44XX_DMA_MCPDM_DL,
- .data_type = OMAP_DMA_DATA_TYPE_S32,
- .sync_mode = OMAP_DMA_SYNC_PACKET,
- .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_DN_DATA,
- },
- {
- .name = "Audio capture",
- .dma_req = OMAP44XX_DMA_MCPDM_UP,
- .data_type = OMAP_DMA_DATA_TYPE_S32,
- .sync_mode = OMAP_DMA_SYNC_PACKET,
- .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_UP_DATA,
- },
-};
-
-static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val)
-{
- __raw_writel(val, mcpdm->io_base + reg);
-}
-
-static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg)
-{
- return __raw_readl(mcpdm->io_base + reg);
-}
-
-#ifdef DEBUG
-static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm)
-{
- dev_dbg(mcpdm->dev, "***********************\n");
- dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS_RAW));
- dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS));
- dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_SET));
- dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_CLR));
- dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_IRQWAKE_EN));
- dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_SET));
- dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_CLR));
- dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_DMAWAKEEN));
- dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL));
- dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_DN_DATA));
- dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_UP_DATA));
- dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_DN));
- dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n",
- omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_UP));
- dev_dbg(mcpdm->dev, "***********************\n");
-}
-#else
-static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {}
-#endif
-
-/*
- * Enables the transfer through the PDM interface to/from the Phoenix
- * codec by enabling the corresponding UP or DN channels.
- */
-static void omap_mcpdm_start(struct omap_mcpdm *mcpdm)
-{
- u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
-
- ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
-
- ctrl |= mcpdm->dn_channels | mcpdm->up_channels;
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
-
- ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
-}
-
-/*
- * Disables the transfer through the PDM interface to/from the Phoenix
- * codec by disabling the corresponding UP or DN channels.
- */
-static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm)
-{
- u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
-
- ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
-
- ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels);
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
-
- ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
-
-}
-
-/*
- * Is the physical McPDM interface active.
- */
-static inline int omap_mcpdm_active(struct omap_mcpdm *mcpdm)
-{
- return omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL) &
- (MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK);
-}
-
-/*
- * Configures McPDM uplink, and downlink for audio.
- * This function should be called before omap_mcpdm_start.
- */
-static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm)
-{
- omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_SET,
- MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL |
- MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL);
-
- /* Enable DN RX1/2 offset cancellation feature, if configured */
- if (mcpdm->dn_rx_offset) {
- u32 dn_offset = mcpdm->dn_rx_offset;
-
- omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset);
- dn_offset |= (MCPDM_DN_OFST_RX1_EN | MCPDM_DN_OFST_RX2_EN);
- omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset);
- }
-
- omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold);
- omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold);
-
- omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET,
- MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE);
-}
-
-/*
- * Cleans McPDM uplink, and downlink configuration.
- * This function should be called when the stream is closed.
- */
-static void omap_mcpdm_close_streams(struct omap_mcpdm *mcpdm)
-{
- /* Disable irq request generation for downlink */
- omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR,
- MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL);
-
- /* Disable DMA request generation for downlink */
- omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_DN_ENABLE);
-
- /* Disable irq request generation for uplink */
- omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR,
- MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL);
-
- /* Disable DMA request generation for uplink */
- omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_UP_ENABLE);
-
- /* Disable RX1/2 offset cancellation */
- if (mcpdm->dn_rx_offset)
- omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, 0);
-}
-
-static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id)
-{
- struct omap_mcpdm *mcpdm = dev_id;
- int irq_status;
-
- irq_status = omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS);
-
- /* Acknowledge irq event */
- omap_mcpdm_write(mcpdm, MCPDM_REG_IRQSTATUS, irq_status);
-
- if (irq_status & MCPDM_DN_IRQ_FULL)
- dev_dbg(mcpdm->dev, "DN (playback) FIFO Full\n");
-
- if (irq_status & MCPDM_DN_IRQ_EMPTY)
- dev_dbg(mcpdm->dev, "DN (playback) FIFO Empty\n");
-
- if (irq_status & MCPDM_DN_IRQ)
- dev_dbg(mcpdm->dev, "DN (playback) write request\n");
-
- if (irq_status & MCPDM_UP_IRQ_FULL)
- dev_dbg(mcpdm->dev, "UP (capture) FIFO Full\n");
-
- if (irq_status & MCPDM_UP_IRQ_EMPTY)
- dev_dbg(mcpdm->dev, "UP (capture) FIFO Empty\n");
-
- if (irq_status & MCPDM_UP_IRQ)
- dev_dbg(mcpdm->dev, "UP (capture) write request\n");
-
- return IRQ_HANDLED;
-}
-
-static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
-
- mutex_lock(&mcpdm->mutex);
-
- if (!dai->active) {
- /* Enable watch dog for ES above ES 1.0 to avoid saturation */
- if (omap_rev() != OMAP4430_REV_ES1_0) {
- u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
-
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL,
- ctrl | MCPDM_WD_EN);
- }
- omap_mcpdm_open_streams(mcpdm);
- }
-
- mutex_unlock(&mcpdm->mutex);
-
- return 0;
-}
-
-static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
-
- mutex_lock(&mcpdm->mutex);
-
- if (!dai->active) {
- if (omap_mcpdm_active(mcpdm)) {
- omap_mcpdm_stop(mcpdm);
- omap_mcpdm_close_streams(mcpdm);
- }
- }
-
- mutex_unlock(&mcpdm->mutex);
-}
-
-static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
- int stream = substream->stream;
- struct omap_pcm_dma_data *dma_data;
- int channels;
- int link_mask = 0;
-
- channels = params_channels(params);
- switch (channels) {
- case 5:
- if (stream == SNDRV_PCM_STREAM_CAPTURE)
- /* up to 3 channels for capture */
- return -EINVAL;
- link_mask |= 1 << 4;
- case 4:
- if (stream == SNDRV_PCM_STREAM_CAPTURE)
- /* up to 3 channels for capture */
- return -EINVAL;
- link_mask |= 1 << 3;
- case 3:
- link_mask |= 1 << 2;
- case 2:
- link_mask |= 1 << 1;
- case 1:
- link_mask |= 1 << 0;
- break;
- default:
- /* unsupported number of channels */
- return -EINVAL;
- }
-
- dma_data = &omap_mcpdm_dai_dma_params[stream];
-
- /* Configure McPDM channels, and DMA packet size */
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- mcpdm->dn_channels = link_mask << 3;
- dma_data->packet_size =
- (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels;
- } else {
- mcpdm->up_channels = link_mask << 0;
- dma_data->packet_size = mcpdm->up_threshold * channels;
- }
-
- snd_soc_dai_set_dma_data(dai, substream, dma_data);
-
- return 0;
-}
-
-static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
-
- if (!omap_mcpdm_active(mcpdm)) {
- omap_mcpdm_start(mcpdm);
- omap_mcpdm_reg_dump(mcpdm);
- }
-
- return 0;
-}
-
-static const struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
- .startup = omap_mcpdm_dai_startup,
- .shutdown = omap_mcpdm_dai_shutdown,
- .hw_params = omap_mcpdm_dai_hw_params,
- .prepare = omap_mcpdm_prepare,
-};
-
-static int omap_mcpdm_probe(struct snd_soc_dai *dai)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
- int ret;
-
- pm_runtime_enable(mcpdm->dev);
-
- /* Disable lines while request is ongoing */
- pm_runtime_get_sync(mcpdm->dev);
- omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00);
-
- ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
- 0, "McPDM", (void *)mcpdm);
-
- pm_runtime_put_sync(mcpdm->dev);
-
- if (ret) {
- dev_err(mcpdm->dev, "Request for IRQ failed\n");
- pm_runtime_disable(mcpdm->dev);
- }
-
- /* Configure McPDM threshold values */
- mcpdm->dn_threshold = 2;
- mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3;
- return ret;
-}
-
-static int omap_mcpdm_remove(struct snd_soc_dai *dai)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
-
- free_irq(mcpdm->irq, (void *)mcpdm);
- pm_runtime_disable(mcpdm->dev);
-
- return 0;
-}
-
-#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-#define OMAP_MCPDM_FORMATS SNDRV_PCM_FMTBIT_S32_LE
-
-static struct snd_soc_dai_driver omap_mcpdm_dai = {
- .probe = omap_mcpdm_probe,
- .remove = omap_mcpdm_remove,
- .probe_order = SND_SOC_COMP_ORDER_LATE,
- .remove_order = SND_SOC_COMP_ORDER_EARLY,
- .playback = {
- .channels_min = 1,
- .channels_max = 5,
- .rates = OMAP_MCPDM_RATES,
- .formats = OMAP_MCPDM_FORMATS,
- .sig_bits = 24,
- },
- .capture = {
- .channels_min = 1,
- .channels_max = 3,
- .rates = OMAP_MCPDM_RATES,
- .formats = OMAP_MCPDM_FORMATS,
- .sig_bits = 24,
- },
- .ops = &omap_mcpdm_dai_ops,
-};
-
-void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
- u8 rx1, u8 rx2)
-{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-
- mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2);
-}
-EXPORT_SYMBOL_GPL(omap_mcpdm_configure_dn_offsets);
-
-static __devinit int asoc_mcpdm_probe(struct platform_device *pdev)
-{
- struct omap_mcpdm *mcpdm;
- struct resource *res;
- int ret = 0;
-
- mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
- if (!mcpdm)
- return -ENOMEM;
-
- platform_set_drvdata(pdev, mcpdm);
-
- mutex_init(&mcpdm->mutex);
-
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res == NULL) {
- dev_err(&pdev->dev, "no resource\n");
- goto err_res;
- }
-
- if (!request_mem_region(res->start, resource_size(res), "McPDM")) {
- ret = -EBUSY;
- goto err_res;
- }
-
- mcpdm->io_base = ioremap(res->start, resource_size(res));
- if (!mcpdm->io_base) {
- ret = -ENOMEM;
- goto err_iomap;
- }
-
- mcpdm->irq = platform_get_irq(pdev, 0);
- if (mcpdm->irq < 0) {
- ret = mcpdm->irq;
- goto err_irq;
- }
-
- mcpdm->dev = &pdev->dev;
-
- ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai);
- if (!ret)
- return 0;
-
-err_irq:
- iounmap(mcpdm->io_base);
-err_iomap:
- release_mem_region(res->start, resource_size(res));
-err_res:
- kfree(mcpdm);
- return ret;
-}
-
-static int __devexit asoc_mcpdm_remove(struct platform_device *pdev)
-{
- struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev);
- struct resource *res;
-
- snd_soc_unregister_dai(&pdev->dev);
-
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- iounmap(mcpdm->io_base);
- release_mem_region(res->start, resource_size(res));
-
- kfree(mcpdm);
- return 0;
-}
-
-static struct platform_driver asoc_mcpdm_driver = {
- .driver = {
- .name = "omap-mcpdm",
- .owner = THIS_MODULE,
- },
-
- .probe = asoc_mcpdm_probe,
- .remove = __devexit_p(asoc_mcpdm_remove),
-};
-
-module_platform_driver(asoc_mcpdm_driver);
-
-MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
-MODULE_DESCRIPTION("OMAP PDM SoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.h b/ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.h
deleted file mode 100644
index de8cf265..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-mcpdm.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * omap-mcpdm.h
- *
- * Copyright (C) 2009 - 2011 Texas Instruments
- *
- * Contact: Misael Lopez Cruz <misael.lopez@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#ifndef __OMAP_MCPDM_H__
-#define __OMAP_MCPDM_H__
-
-#define MCPDM_REG_REVISION 0x00
-#define MCPDM_REG_SYSCONFIG 0x10
-#define MCPDM_REG_IRQSTATUS_RAW 0x24
-#define MCPDM_REG_IRQSTATUS 0x28
-#define MCPDM_REG_IRQENABLE_SET 0x2C
-#define MCPDM_REG_IRQENABLE_CLR 0x30
-#define MCPDM_REG_IRQWAKE_EN 0x34
-#define MCPDM_REG_DMAENABLE_SET 0x38
-#define MCPDM_REG_DMAENABLE_CLR 0x3C
-#define MCPDM_REG_DMAWAKEEN 0x40
-#define MCPDM_REG_CTRL 0x44
-#define MCPDM_REG_DN_DATA 0x48
-#define MCPDM_REG_UP_DATA 0x4C
-#define MCPDM_REG_FIFO_CTRL_DN 0x50
-#define MCPDM_REG_FIFO_CTRL_UP 0x54
-#define MCPDM_REG_DN_OFFSET 0x58
-
-/*
- * MCPDM_IRQ bit fields
- * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR
- */
-
-#define MCPDM_DN_IRQ (1 << 0)
-#define MCPDM_DN_IRQ_EMPTY (1 << 1)
-#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2)
-#define MCPDM_DN_IRQ_FULL (1 << 3)
-
-#define MCPDM_UP_IRQ (1 << 8)
-#define MCPDM_UP_IRQ_EMPTY (1 << 9)
-#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10)
-#define MCPDM_UP_IRQ_FULL (1 << 11)
-
-#define MCPDM_DOWNLINK_IRQ_MASK 0x00F
-#define MCPDM_UPLINK_IRQ_MASK 0xF00
-
-/*
- * MCPDM_DMAENABLE bit fields
- */
-
-#define MCPDM_DMA_DN_ENABLE (1 << 0)
-#define MCPDM_DMA_UP_ENABLE (1 << 1)
-
-/*
- * MCPDM_CTRL bit fields
- */
-
-#define MCPDM_PDM_UPLINK_EN(x) (1 << (x - 1)) /* ch1 is at bit 0 */
-#define MCPDM_PDM_DOWNLINK_EN(x) (1 << (x + 2)) /* ch1 is at bit 3 */
-#define MCPDM_PDMOUTFORMAT (1 << 8)
-#define MCPDM_CMD_INT (1 << 9)
-#define MCPDM_STATUS_INT (1 << 10)
-#define MCPDM_SW_UP_RST (1 << 11)
-#define MCPDM_SW_DN_RST (1 << 12)
-#define MCPDM_WD_EN (1 << 14)
-#define MCPDM_PDM_UP_MASK 0x7
-#define MCPDM_PDM_DN_MASK (0x1f << 3)
-
-
-#define MCPDM_PDMOUTFORMAT_LJUST (0 << 8)
-#define MCPDM_PDMOUTFORMAT_RJUST (1 << 8)
-
-/*
- * MCPDM_FIFO_CTRL bit fields
- */
-
-#define MCPDM_UP_THRES_MAX 0xF
-#define MCPDM_DN_THRES_MAX 0xF
-
-/*
- * MCPDM_DN_OFFSET bit fields
- */
-
-#define MCPDM_DN_OFST_RX1_EN (1 << 0)
-#define MCPDM_DNOFST_RX1(x) ((x & 0x1f) << 1)
-#define MCPDM_DN_OFST_RX2_EN (1 << 8)
-#define MCPDM_DNOFST_RX2(x) ((x & 0x1f) << 9)
-
-void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
- u8 rx1, u8 rx2);
-
-#endif /* End of __OMAP_MCPDM_H__ */
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-pcm.c b/ANDROID_3.4.5/sound/soc/omap/omap-pcm.c
deleted file mode 100644
index 5a649da9..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-pcm.c
+++ /dev/null
@@ -1,443 +0,0 @@
-/*
- * omap-pcm.c -- ALSA PCM interface for the OMAP SoC
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/dma-mapping.h>
-#include <linux/slab.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <plat/dma.h>
-#include "omap-pcm.h"
-
-static const struct snd_pcm_hardware omap_pcm_hardware = {
- .info = SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
- .period_bytes_min = 32,
- .period_bytes_max = 64 * 1024,
- .periods_min = 2,
- .periods_max = 255,
- .buffer_bytes_max = 128 * 1024,
-};
-
-struct omap_runtime_data {
- spinlock_t lock;
- struct omap_pcm_dma_data *dma_data;
- int dma_ch;
- int period_index;
-};
-
-static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
-{
- struct snd_pcm_substream *substream = data;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct omap_runtime_data *prtd = runtime->private_data;
- unsigned long flags;
-
- if ((cpu_is_omap1510())) {
- /*
- * OMAP1510 doesn't fully support DMA progress counter
- * and there is no software emulation implemented yet,
- * so have to maintain our own progress counters
- * that can be used by omap_pcm_pointer() instead.
- */
- spin_lock_irqsave(&prtd->lock, flags);
- if ((stat == OMAP_DMA_LAST_IRQ) &&
- (prtd->period_index == runtime->periods - 1)) {
- /* we are in sync, do nothing */
- spin_unlock_irqrestore(&prtd->lock, flags);
- return;
- }
- if (prtd->period_index >= 0) {
- if (stat & OMAP_DMA_BLOCK_IRQ) {
- /* end of buffer reached, loop back */
- prtd->period_index = 0;
- } else if (stat & OMAP_DMA_LAST_IRQ) {
- /* update the counter for the last period */
- prtd->period_index = runtime->periods - 1;
- } else if (++prtd->period_index >= runtime->periods) {
- /* end of buffer missed? loop back */
- prtd->period_index = 0;
- }
- }
- spin_unlock_irqrestore(&prtd->lock, flags);
- }
-
- snd_pcm_period_elapsed(substream);
-}
-
-/* this may get called several times by oss emulation */
-static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct omap_runtime_data *prtd = runtime->private_data;
- struct omap_pcm_dma_data *dma_data;
-
- int err = 0;
-
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- /* return if this is a bufferless transfer e.g.
- * codec <--> BT codec or GSM modem -- lg FIXME */
- if (!dma_data)
- return 0;
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- runtime->dma_bytes = params_buffer_bytes(params);
-
- if (prtd->dma_data)
- return 0;
- prtd->dma_data = dma_data;
- err = omap_request_dma(dma_data->dma_req, dma_data->name,
- omap_pcm_dma_irq, substream, &prtd->dma_ch);
- if (!err) {
- /*
- * Link channel with itself so DMA doesn't need any
- * reprogramming while looping the buffer
- */
- omap_dma_link_lch(prtd->dma_ch, prtd->dma_ch);
- }
-
- return err;
-}
-
-static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct omap_runtime_data *prtd = runtime->private_data;
-
- if (prtd->dma_data == NULL)
- return 0;
-
- omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
- omap_free_dma(prtd->dma_ch);
- prtd->dma_data = NULL;
-
- snd_pcm_set_runtime_buffer(substream, NULL);
-
- return 0;
-}
-
-static int omap_pcm_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct omap_runtime_data *prtd = runtime->private_data;
- struct omap_pcm_dma_data *dma_data = prtd->dma_data;
- struct omap_dma_channel_params dma_params;
- int bytes;
-
- /* return if this is a bufferless transfer e.g.
- * codec <--> BT codec or GSM modem -- lg FIXME */
- if (!prtd->dma_data)
- return 0;
-
- memset(&dma_params, 0, sizeof(dma_params));
- dma_params.data_type = dma_data->data_type;
- dma_params.trigger = dma_data->dma_req;
- dma_params.sync_mode = dma_data->sync_mode;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
- dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
- dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC;
- dma_params.src_start = runtime->dma_addr;
- dma_params.dst_start = dma_data->port_addr;
- dma_params.dst_port = OMAP_DMA_PORT_MPUI;
- dma_params.dst_fi = dma_data->packet_size;
- } else {
- dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT;
- dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
- dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
- dma_params.src_start = dma_data->port_addr;
- dma_params.dst_start = runtime->dma_addr;
- dma_params.src_port = OMAP_DMA_PORT_MPUI;
- dma_params.src_fi = dma_data->packet_size;
- }
- /*
- * Set DMA transfer frame size equal to ALSA period size and frame
- * count as no. of ALSA periods. Then with DMA frame interrupt enabled,
- * we can transfer the whole ALSA buffer with single DMA transfer but
- * still can get an interrupt at each period bounary
- */
- bytes = snd_pcm_lib_period_bytes(substream);
- dma_params.elem_count = bytes >> dma_data->data_type;
- dma_params.frame_count = runtime->periods;
- omap_set_dma_params(prtd->dma_ch, &dma_params);
-
- if ((cpu_is_omap1510()))
- omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
- OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
- else if (!substream->runtime->no_period_wakeup)
- omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
- else {
- /*
- * No period wakeup:
- * we need to disable BLOCK_IRQ, which is enabled by the omap
- * dma core at request dma time.
- */
- omap_disable_dma_irq(prtd->dma_ch, OMAP_DMA_BLOCK_IRQ);
- }
-
- if (!(cpu_class_is_omap1())) {
- omap_set_dma_src_burst_mode(prtd->dma_ch,
- OMAP_DMA_DATA_BURST_16);
- omap_set_dma_dest_burst_mode(prtd->dma_ch,
- OMAP_DMA_DATA_BURST_16);
- }
-
- return 0;
-}
-
-static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct omap_runtime_data *prtd = runtime->private_data;
- struct omap_pcm_dma_data *dma_data = prtd->dma_data;
- unsigned long flags;
- int ret = 0;
-
- spin_lock_irqsave(&prtd->lock, flags);
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- prtd->period_index = 0;
- /* Configure McBSP internal buffer usage */
- if (dma_data->set_threshold)
- dma_data->set_threshold(substream);
-
- omap_start_dma(prtd->dma_ch);
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- prtd->period_index = -1;
- omap_stop_dma(prtd->dma_ch);
- break;
- default:
- ret = -EINVAL;
- }
- spin_unlock_irqrestore(&prtd->lock, flags);
-
- return ret;
-}
-
-static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct omap_runtime_data *prtd = runtime->private_data;
- dma_addr_t ptr;
- snd_pcm_uframes_t offset;
-
- if (cpu_is_omap1510()) {
- offset = prtd->period_index * runtime->period_size;
- } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- ptr = omap_get_dma_dst_pos(prtd->dma_ch);
- offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
- } else {
- ptr = omap_get_dma_src_pos(prtd->dma_ch);
- offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
- }
-
- if (offset >= runtime->buffer_size)
- offset = 0;
-
- return offset;
-}
-
-static int omap_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct omap_runtime_data *prtd;
- int ret;
-
- snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
-
- /* Ensure that buffer size is a multiple of period size */
- ret = snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS);
- if (ret < 0)
- goto out;
-
- prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
- if (prtd == NULL) {
- ret = -ENOMEM;
- goto out;
- }
- spin_lock_init(&prtd->lock);
- runtime->private_data = prtd;
-
-out:
- return ret;
-}
-
-static int omap_pcm_close(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- kfree(runtime->private_data);
- return 0;
-}
-
-static int omap_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- return dma_mmap_writecombine(substream->pcm->card->dev, vma,
- runtime->dma_area,
- runtime->dma_addr,
- runtime->dma_bytes);
-}
-
-static struct snd_pcm_ops omap_pcm_ops = {
- .open = omap_pcm_open,
- .close = omap_pcm_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = omap_pcm_hw_params,
- .hw_free = omap_pcm_hw_free,
- .prepare = omap_pcm_prepare,
- .trigger = omap_pcm_trigger,
- .pointer = omap_pcm_pointer,
- .mmap = omap_pcm_mmap,
-};
-
-static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
-
-static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
- int stream)
-{
- struct snd_pcm_substream *substream = pcm->streams[stream].substream;
- struct snd_dma_buffer *buf = &substream->dma_buffer;
- size_t size = omap_pcm_hardware.buffer_bytes_max;
-
- buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = pcm->card->dev;
- buf->private_data = NULL;
- buf->area = dma_alloc_writecombine(pcm->card->dev, size,
- &buf->addr, GFP_KERNEL);
- if (!buf->area)
- return -ENOMEM;
-
- buf->bytes = size;
- return 0;
-}
-
-static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (!substream)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
-
- dma_free_writecombine(pcm->card->dev, buf->bytes,
- buf->area, buf->addr);
- buf->area = NULL;
- }
-}
-
-static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_card *card = rtd->card->snd_card;
- struct snd_pcm *pcm = rtd->pcm;
- int ret = 0;
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &omap_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
-
- if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
- ret = omap_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_PLAYBACK);
- if (ret)
- goto out;
- }
-
- if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
- ret = omap_pcm_preallocate_dma_buffer(pcm,
- SNDRV_PCM_STREAM_CAPTURE);
- if (ret)
- goto out;
- }
-
-out:
- /* free preallocated buffers in case of error */
- if (ret)
- omap_pcm_free_dma_buffers(pcm);
-
- return ret;
-}
-
-static struct snd_soc_platform_driver omap_soc_platform = {
- .ops = &omap_pcm_ops,
- .pcm_new = omap_pcm_new,
- .pcm_free = omap_pcm_free_dma_buffers,
-};
-
-static __devinit int omap_pcm_probe(struct platform_device *pdev)
-{
- return snd_soc_register_platform(&pdev->dev,
- &omap_soc_platform);
-}
-
-static int __devexit omap_pcm_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
-}
-
-static struct platform_driver omap_pcm_driver = {
- .driver = {
- .name = "omap-pcm-audio",
- .owner = THIS_MODULE,
- },
-
- .probe = omap_pcm_probe,
- .remove = __devexit_p(omap_pcm_remove),
-};
-
-module_platform_driver(omap_pcm_driver);
-
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
-MODULE_DESCRIPTION("OMAP PCM DMA module");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap-pcm.h b/ANDROID_3.4.5/sound/soc/omap/omap-pcm.h
deleted file mode 100644
index b92248cb..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap-pcm.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * omap-pcm.h
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#ifndef __OMAP_PCM_H__
-#define __OMAP_PCM_H__
-
-struct snd_pcm_substream;
-
-struct omap_pcm_dma_data {
- char *name; /* stream identifier */
- int dma_req; /* DMA request line */
- unsigned long port_addr; /* transmit/receive register */
- void (*set_threshold)(struct snd_pcm_substream *substream);
- int data_type; /* data type 8,16,32 */
- int sync_mode; /* DMA sync mode */
- int packet_size; /* packet size only in PACKET mode */
-};
-
-#endif
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap3beagle.c b/ANDROID_3.4.5/sound/soc/omap/omap3beagle.c
deleted file mode 100644
index 2830dfd0..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap3beagle.c
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * omap3beagle.c -- SoC audio for OMAP3 Beagle
- *
- * Author: Steve Sakoman <steve@sakoman.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- unsigned int fmt;
- int ret;
-
- switch (params_channels(params)) {
- case 2: /* Stereo I2S mode */
- fmt = SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM;
- break;
- case 4: /* Four channel TDM mode */
- fmt = SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBM_CFM;
- break;
- default:
- return -EINVAL;
- }
-
- /* Set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, fmt);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec DAI configuration\n");
- return ret;
- }
-
- /* Set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
- if (ret < 0) {
- printk(KERN_ERR "can't set cpu DAI configuration\n");
- return ret;
- }
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops omap3beagle_ops = {
- .hw_params = omap3beagle_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link omap3beagle_dai = {
- .name = "TWL4030",
- .stream_name = "TWL4030",
- .cpu_dai_name = "omap-mcbsp.2",
- .platform_name = "omap-pcm-audio",
- .codec_dai_name = "twl4030-hifi",
- .codec_name = "twl4030-codec",
- .ops = &omap3beagle_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_omap3beagle = {
- .name = "omap3beagle",
- .owner = THIS_MODULE,
- .dai_link = &omap3beagle_dai,
- .num_links = 1,
-};
-
-static struct platform_device *omap3beagle_snd_device;
-
-static int __init omap3beagle_soc_init(void)
-{
- int ret;
-
- if (!(machine_is_omap3_beagle() || machine_is_devkit8000()))
- return -ENODEV;
- pr_info("OMAP3 Beagle/Devkit8000 SoC init\n");
-
- omap3beagle_snd_device = platform_device_alloc("soc-audio", -1);
- if (!omap3beagle_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(omap3beagle_snd_device, &snd_soc_omap3beagle);
-
- ret = platform_device_add(omap3beagle_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(omap3beagle_snd_device);
-
- return ret;
-}
-
-static void __exit omap3beagle_soc_exit(void)
-{
- platform_device_unregister(omap3beagle_snd_device);
-}
-
-module_init(omap3beagle_soc_init);
-module_exit(omap3beagle_soc_exit);
-
-MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
-MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap3evm.c b/ANDROID_3.4.5/sound/soc/omap/omap3evm.c
deleted file mode 100644
index 3d468c91..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap3evm.c
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * omap3evm.c -- ALSA SoC support for OMAP3 EVM
- *
- * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
- *
- * Based on sound/soc/omap/beagle.c by Steve Sakoman
- *
- * Copyright (C) 2008 Texas Instruments, Incorporated
- *
- * 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.
- *
- * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
- * whether express or implied; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-static int omap3evm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "Can't set codec system clock\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops omap3evm_ops = {
- .hw_params = omap3evm_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link omap3evm_dai = {
- .name = "TWL4030",
- .stream_name = "TWL4030",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &omap3evm_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_omap3evm = {
- .name = "omap3evm",
- .owner = THIS_MODULE,
- .dai_link = &omap3evm_dai,
- .num_links = 1,
-};
-
-static struct platform_device *omap3evm_snd_device;
-
-static int __init omap3evm_soc_init(void)
-{
- int ret;
-
- if (!machine_is_omap3evm())
- return -ENODEV;
- pr_info("OMAP3 EVM SoC init\n");
-
- omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
- if (!omap3evm_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(omap3evm_snd_device, &snd_soc_omap3evm);
- ret = platform_device_add(omap3evm_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(omap3evm_snd_device);
-
- return ret;
-}
-
-static void __exit omap3evm_soc_exit(void)
-{
- platform_device_unregister(omap3evm_snd_device);
-}
-
-module_init(omap3evm_soc_init);
-module_exit(omap3evm_soc_exit);
-
-MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPL v2");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap3pandora.c b/ANDROID_3.4.5/sound/soc/omap/omap3pandora.c
deleted file mode 100644
index 4c3a0978..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap3pandora.c
+++ /dev/null
@@ -1,325 +0,0 @@
-/*
- * omap3pandora.c -- SoC audio for Pandora Handheld Console
- *
- * Author: Gražvydas Ignotas <notasas@gmail.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/gpio.h>
-#include <linux/delay.h>
-#include <linux/regulator/consumer.h>
-#include <linux/module.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#define OMAP3_PANDORA_DAC_POWER_GPIO 118
-#define OMAP3_PANDORA_AMP_POWER_GPIO 14
-
-#define PREFIX "ASoC omap3pandora: "
-
-static struct regulator *omap3pandora_dac_reg;
-
-static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- pr_err(PREFIX "can't set codec system clock\n");
- return ret;
- }
-
- /* Set McBSP clock to external */
- ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT,
- 256 * params_rate(params),
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- pr_err(PREFIX "can't set cpu system clock\n");
- return ret;
- }
-
- ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
- if (ret < 0) {
- pr_err(PREFIX "can't set SRG clock divider\n");
- return ret;
- }
-
- return 0;
-}
-
-static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- /*
- * The PCM1773 DAC datasheet requires 1ms delay between switching
- * VCC power on/off and /PD pin high/low
- */
- if (SND_SOC_DAPM_EVENT_ON(event)) {
- regulator_enable(omap3pandora_dac_reg);
- mdelay(1);
- gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
- } else {
- gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
- mdelay(1);
- regulator_disable(omap3pandora_dac_reg);
- }
-
- return 0;
-}
-
-static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- if (SND_SOC_DAPM_EVENT_ON(event))
- gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
- else
- gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
-
- return 0;
-}
-
-/*
- * Audio paths on Pandora board:
- *
- * |O| ---> PCM DAC +-> AMP -> Headphone Jack
- * |M| A +--------> Line Out
- * |A| <~~clk~~+
- * |P| <--- TWL4030 <--------- Line In and MICs
- */
-static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
- SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM,
- 0, 0, omap3pandora_dac_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
- 0, 0, NULL, 0, omap3pandora_hp_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_LINE("Line Out", NULL),
-};
-
-static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Mic (internal)", NULL),
- SND_SOC_DAPM_MIC("Mic (external)", NULL),
- SND_SOC_DAPM_LINE("Line In", NULL),
-};
-
-static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
- {"PCM DAC", NULL, "APLL Enable"},
- {"Headphone Amplifier", NULL, "PCM DAC"},
- {"Line Out", NULL, "PCM DAC"},
- {"Headphone Jack", NULL, "Headphone Amplifier"},
-};
-
-static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
- {"AUXL", NULL, "Line In"},
- {"AUXR", NULL, "Line In"},
-
- {"MAINMIC", NULL, "Mic Bias 1"},
- {"Mic Bias 1", NULL, "Mic (internal)"},
-
- {"SUBMIC", NULL, "Mic Bias 2"},
- {"Mic Bias 2", NULL, "Mic (external)"},
-};
-
-static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- /* All TWL4030 output pins are floating */
- snd_soc_dapm_nc_pin(dapm, "EARPIECE");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
- snd_soc_dapm_nc_pin(dapm, "HSOL");
- snd_soc_dapm_nc_pin(dapm, "HSOR");
- snd_soc_dapm_nc_pin(dapm, "CARKITL");
- snd_soc_dapm_nc_pin(dapm, "CARKITR");
- snd_soc_dapm_nc_pin(dapm, "HFL");
- snd_soc_dapm_nc_pin(dapm, "HFR");
- snd_soc_dapm_nc_pin(dapm, "VIBRA");
-
- ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets,
- ARRAY_SIZE(omap3pandora_out_dapm_widgets));
- if (ret < 0)
- return ret;
-
- return snd_soc_dapm_add_routes(dapm, omap3pandora_out_map,
- ARRAY_SIZE(omap3pandora_out_map));
-}
-
-static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- /* Not comnnected */
- snd_soc_dapm_nc_pin(dapm, "HSMIC");
- snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
-
- ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets,
- ARRAY_SIZE(omap3pandora_in_dapm_widgets));
- if (ret < 0)
- return ret;
-
- return snd_soc_dapm_add_routes(dapm, omap3pandora_in_map,
- ARRAY_SIZE(omap3pandora_in_map));
-}
-
-static struct snd_soc_ops omap3pandora_ops = {
- .hw_params = omap3pandora_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link omap3pandora_dai[] = {
- {
- .name = "PCM1773",
- .stream_name = "HiFi Out",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &omap3pandora_ops,
- .init = omap3pandora_out_init,
- }, {
- .name = "TWL4030",
- .stream_name = "Line/Mic In",
- .cpu_dai_name = "omap-mcbsp.4",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &omap3pandora_ops,
- .init = omap3pandora_in_init,
- }
-};
-
-/* SoC card */
-static struct snd_soc_card snd_soc_card_omap3pandora = {
- .name = "omap3pandora",
- .owner = THIS_MODULE,
- .dai_link = omap3pandora_dai,
- .num_links = ARRAY_SIZE(omap3pandora_dai),
-};
-
-static struct platform_device *omap3pandora_snd_device;
-
-static int __init omap3pandora_soc_init(void)
-{
- int ret;
-
- if (!machine_is_omap3_pandora())
- return -ENODEV;
-
- pr_info("OMAP3 Pandora SoC init\n");
-
- ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
- if (ret) {
- pr_err(PREFIX "Failed to get DAC power GPIO\n");
- return ret;
- }
-
- ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
- if (ret) {
- pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
- goto fail0;
- }
-
- ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
- if (ret) {
- pr_err(PREFIX "Failed to get amp power GPIO\n");
- goto fail0;
- }
-
- ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
- if (ret) {
- pr_err(PREFIX "Failed to set amp power GPIO direction\n");
- goto fail1;
- }
-
- omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
- if (omap3pandora_snd_device == NULL) {
- pr_err(PREFIX "Platform device allocation failed\n");
- ret = -ENOMEM;
- goto fail1;
- }
-
- platform_set_drvdata(omap3pandora_snd_device, &snd_soc_card_omap3pandora);
-
- ret = platform_device_add(omap3pandora_snd_device);
- if (ret) {
- pr_err(PREFIX "Unable to add platform device\n");
- goto fail2;
- }
-
- omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc");
- if (IS_ERR(omap3pandora_dac_reg)) {
- pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n",
- dev_name(&omap3pandora_snd_device->dev),
- PTR_ERR(omap3pandora_dac_reg));
- ret = PTR_ERR(omap3pandora_dac_reg);
- goto fail3;
- }
-
- return 0;
-
-fail3:
- platform_device_del(omap3pandora_snd_device);
-fail2:
- platform_device_put(omap3pandora_snd_device);
-fail1:
- gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
-fail0:
- gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
- return ret;
-}
-module_init(omap3pandora_soc_init);
-
-static void __exit omap3pandora_soc_exit(void)
-{
- regulator_put(omap3pandora_dac_reg);
- platform_device_unregister(omap3pandora_snd_device);
- gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
- gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
-}
-module_exit(omap3pandora_soc_exit);
-
-MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
-MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/omap4-hdmi-card.c b/ANDROID_3.4.5/sound/soc/omap/omap4-hdmi-card.c
deleted file mode 100644
index 28d689b2..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/omap4-hdmi-card.c
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * omap4-hdmi-card.c
- *
- * OMAP ALSA SoC machine driver for TI OMAP4 HDMI
- * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/module.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <asm/mach-types.h>
-#include <video/omapdss.h>
-
-#define DRV_NAME "omap4-hdmi-audio"
-
-static int omap4_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- int i;
- struct omap_overlay_manager *mgr = NULL;
- struct device *dev = substream->pcm->card->dev;
-
- /* Find DSS HDMI device */
- for (i = 0; i < omap_dss_get_num_overlay_managers(); i++) {
- mgr = omap_dss_get_overlay_manager(i);
- if (mgr && mgr->device
- && mgr->device->type == OMAP_DISPLAY_TYPE_HDMI)
- break;
- }
-
- if (i == omap_dss_get_num_overlay_managers()) {
- dev_err(dev, "HDMI display device not found!\n");
- return -ENODEV;
- }
-
- /* Make sure HDMI is power-on to avoid L3 interconnect errors */
- if (mgr->device->state != OMAP_DSS_DISPLAY_ACTIVE) {
- dev_err(dev, "HDMI display is not active!\n");
- return -EIO;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops omap4_hdmi_dai_ops = {
- .hw_params = omap4_hdmi_dai_hw_params,
-};
-
-static struct snd_soc_dai_link omap4_hdmi_dai = {
- .name = "HDMI",
- .stream_name = "HDMI",
- .cpu_dai_name = "hdmi-audio-dai",
- .platform_name = "omap-pcm-audio",
- .codec_name = "omapdss_hdmi",
- .codec_dai_name = "hdmi-audio-codec",
- .ops = &omap4_hdmi_dai_ops,
-};
-
-static struct snd_soc_card snd_soc_omap4_hdmi = {
- .name = "OMAP4HDMI",
- .owner = THIS_MODULE,
- .dai_link = &omap4_hdmi_dai,
- .num_links = 1,
-};
-
-static __devinit int omap4_hdmi_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &snd_soc_omap4_hdmi;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = snd_soc_register_card(card);
- if (ret) {
- dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
- card->dev = NULL;
- return ret;
- }
- return 0;
-}
-
-static int __devexit omap4_hdmi_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- card->dev = NULL;
- return 0;
-}
-
-static struct platform_driver omap4_hdmi_driver = {
- .driver = {
- .name = "omap4-hdmi-audio",
- .owner = THIS_MODULE,
- },
- .probe = omap4_hdmi_probe,
- .remove = __devexit_p(omap4_hdmi_remove),
-};
-
-module_platform_driver(omap4_hdmi_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/ANDROID_3.4.5/sound/soc/omap/osk5912.c b/ANDROID_3.4.5/sound/soc/omap/osk5912.c
deleted file mode 100644
index b1a9d64c..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/osk5912.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * osk5912.c -- SoC audio for OSK 5912
- *
- * Copyright (C) 2008 Mistral Solutions
- *
- * Contact: Arun KS <arunks@mistralsolutions.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <linux/gpio.h>
-#include <linux/module.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-#include "../codecs/tlv320aic23.h"
-
-#define CODEC_CLOCK 12000000
-
-static struct clk *tlv320aic23_mclk;
-
-static int osk_startup(struct snd_pcm_substream *substream)
-{
- return clk_enable(tlv320aic23_mclk);
-}
-
-static void osk_shutdown(struct snd_pcm_substream *substream)
-{
- clk_disable(tlv320aic23_mclk);
-}
-
-static int osk_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int err;
-
- /* Set the codec system clock for DAC and ADC */
- err =
- snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
-
- if (err < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return err;
- }
-
- return err;
-}
-
-static struct snd_soc_ops osk_ops = {
- .startup = osk_startup,
- .hw_params = osk_hw_params,
- .shutdown = osk_shutdown,
-};
-
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_LINE("Line In", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- {"Headphone Jack", NULL, "LHPOUT"},
- {"Headphone Jack", NULL, "RHPOUT"},
-
- {"LLINEIN", NULL, "Line In"},
- {"RLINEIN", NULL, "Line In"},
-
- {"MICIN", NULL, "Mic Jack"},
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link osk_dai = {
- .name = "TLV320AIC23",
- .stream_name = "AIC23",
- .cpu_dai_name = "omap-mcbsp.1",
- .codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "tlv320aic23-codec",
- .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &osk_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_card_osk = {
- .name = "OSK5912",
- .owner = THIS_MODULE,
- .dai_link = &osk_dai,
- .num_links = 1,
-
- .dapm_widgets = tlv320aic23_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *osk_snd_device;
-
-static int __init osk_soc_init(void)
-{
- int err;
- u32 curRate;
- struct device *dev;
-
- if (!(machine_is_omap_osk()))
- return -ENODEV;
-
- osk_snd_device = platform_device_alloc("soc-audio", -1);
- if (!osk_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(osk_snd_device, &snd_soc_card_osk);
- err = platform_device_add(osk_snd_device);
- if (err)
- goto err1;
-
- dev = &osk_snd_device->dev;
-
- tlv320aic23_mclk = clk_get(dev, "mclk");
- if (IS_ERR(tlv320aic23_mclk)) {
- printk(KERN_ERR "Could not get mclk clock\n");
- err = PTR_ERR(tlv320aic23_mclk);
- goto err2;
- }
-
- /*
- * Configure 12 MHz output on MCLK.
- */
- curRate = (uint) clk_get_rate(tlv320aic23_mclk);
- if (curRate != CODEC_CLOCK) {
- if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
- printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
- err = -ECANCELED;
- goto err3;
- }
- }
-
- printk(KERN_INFO "MCLK = %d [%d]\n",
- (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
-
- return 0;
-
-err3:
- clk_put(tlv320aic23_mclk);
-err2:
- platform_device_del(osk_snd_device);
-err1:
- platform_device_put(osk_snd_device);
-
- return err;
-
-}
-
-static void __exit osk_soc_exit(void)
-{
- clk_put(tlv320aic23_mclk);
- platform_device_unregister(osk_snd_device);
-}
-
-module_init(osk_soc_init);
-module_exit(osk_soc_exit);
-
-MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
-MODULE_DESCRIPTION("ALSA SoC OSK 5912");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/overo.c b/ANDROID_3.4.5/sound/soc/omap/overo.c
deleted file mode 100644
index 6ac3e0c3..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/overo.c
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * overo.c -- SoC audio for Gumstix Overo
- *
- * Author: Steve Sakoman <steve@sakoman.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-static int overo_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops overo_ops = {
- .hw_params = overo_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link overo_dai = {
- .name = "TWL4030",
- .stream_name = "TWL4030",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &overo_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_card_overo = {
- .name = "overo",
- .owner = THIS_MODULE,
- .dai_link = &overo_dai,
- .num_links = 1,
-};
-
-static struct platform_device *overo_snd_device;
-
-static int __init overo_soc_init(void)
-{
- int ret;
-
- if (!(machine_is_overo() || machine_is_cm_t35())) {
- pr_debug("Incomatible machine!\n");
- return -ENODEV;
- }
- printk(KERN_INFO "overo SoC init\n");
-
- overo_snd_device = platform_device_alloc("soc-audio", -1);
- if (!overo_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(overo_snd_device, &snd_soc_card_overo);
-
- ret = platform_device_add(overo_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(overo_snd_device);
-
- return ret;
-}
-module_init(overo_soc_init);
-
-static void __exit overo_soc_exit(void)
-{
- platform_device_unregister(overo_snd_device);
-}
-module_exit(overo_soc_exit);
-
-MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
-MODULE_DESCRIPTION("ALSA SoC overo");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/rx51.c b/ANDROID_3.4.5/sound/soc/omap/rx51.c
deleted file mode 100644
index 2712dd23..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/rx51.c
+++ /dev/null
@@ -1,451 +0,0 @@
-/*
- * rx51.c -- SoC audio for Nokia RX-51
- *
- * Copyright (C) 2008 - 2009 Nokia Corporation
- *
- * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
- * Eduardo Valentin <eduardo.valentin@nokia.com>
- * Jarkko Nikula <jarkko.nikula@bitmer.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/delay.h>
-#include <linux/gpio.h>
-#include <linux/platform_device.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/jack.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <plat/mcbsp.h>
-#include "../codecs/tpa6130a2.h"
-
-#include <asm/mach-types.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#define RX51_TVOUT_SEL_GPIO 40
-#define RX51_JACK_DETECT_GPIO 177
-#define RX51_ECI_SW_GPIO 182
-/*
- * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
- * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
- */
-#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7)
-
-enum {
- RX51_JACK_DISABLED,
- RX51_JACK_TVOUT, /* tv-out with stereo output */
- RX51_JACK_HP, /* headphone: stereo output, no mic */
- RX51_JACK_HS, /* headset: stereo output with mic */
-};
-
-static int rx51_spk_func;
-static int rx51_dmic_func;
-static int rx51_jack_func;
-
-static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
-{
- int hp = 0, hs = 0, tvout = 0;
-
- switch (rx51_jack_func) {
- case RX51_JACK_TVOUT:
- tvout = 1;
- hp = 1;
- break;
- case RX51_JACK_HS:
- hs = 1;
- case RX51_JACK_HP:
- hp = 1;
- break;
- }
-
- if (rx51_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
- if (rx51_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
- else
- snd_soc_dapm_disable_pin(dapm, "DMic");
- if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- if (hs)
- snd_soc_dapm_enable_pin(dapm, "HS Mic");
- else
- snd_soc_dapm_disable_pin(dapm, "HS Mic");
-
- gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
-
- snd_soc_dapm_sync(dapm);
-}
-
-static int rx51_startup(struct snd_pcm_substream *substream)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_card *card = rtd->card;
-
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
- rx51_ext_control(&card->dapm);
-
- return 0;
-}
-
-static int rx51_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
- /* Set the codec system clock for DAC and ADC */
- return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
- SND_SOC_CLOCK_IN);
-}
-
-static struct snd_soc_ops rx51_ops = {
- .startup = rx51_startup,
- .hw_params = rx51_hw_params,
-};
-
-static int rx51_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = rx51_spk_func;
-
- return 0;
-}
-
-static int rx51_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (rx51_spk_func == ucontrol->value.integer.value[0])
- return 0;
-
- rx51_spk_func = ucontrol->value.integer.value[0];
- rx51_ext_control(&card->dapm);
-
- return 1;
-}
-
-static int rx51_spk_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- if (SND_SOC_DAPM_EVENT_ON(event))
- gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 1);
- else
- gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 0);
-
- return 0;
-}
-
-static int rx51_hp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- struct snd_soc_codec *codec = w->dapm->codec;
-
- if (SND_SOC_DAPM_EVENT_ON(event))
- tpa6130a2_stereo_enable(codec, 1);
- else
- tpa6130a2_stereo_enable(codec, 0);
-
- return 0;
-}
-
-static int rx51_get_input(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = rx51_dmic_func;
-
- return 0;
-}
-
-static int rx51_set_input(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (rx51_dmic_func == ucontrol->value.integer.value[0])
- return 0;
-
- rx51_dmic_func = ucontrol->value.integer.value[0];
- rx51_ext_control(&card->dapm);
-
- return 1;
-}
-
-static int rx51_get_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = rx51_jack_func;
-
- return 0;
-}
-
-static int rx51_set_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (rx51_jack_func == ucontrol->value.integer.value[0])
- return 0;
-
- rx51_jack_func = ucontrol->value.integer.value[0];
- rx51_ext_control(&card->dapm);
-
- return 1;
-}
-
-static struct snd_soc_jack rx51_av_jack;
-
-static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
- {
- .gpio = RX51_JACK_DETECT_GPIO,
- .name = "avdet-gpio",
- .report = SND_JACK_HEADSET,
- .invert = 1,
- .debounce_time = 200,
- },
-};
-
-static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
- SND_SOC_DAPM_MIC("DMic", NULL),
- SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event),
- SND_SOC_DAPM_MIC("HS Mic", NULL),
- SND_SOC_DAPM_LINE("FM Transmitter", NULL),
-};
-
-static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = {
- SND_SOC_DAPM_SPK("Earphone", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- {"Ext Spk", NULL, "HPLOUT"},
- {"Ext Spk", NULL, "HPROUT"},
- {"Headphone Jack", NULL, "LLOUT"},
- {"Headphone Jack", NULL, "RLOUT"},
- {"FM Transmitter", NULL, "LLOUT"},
- {"FM Transmitter", NULL, "RLOUT"},
-
- {"DMic Rate 64", NULL, "Mic Bias 2V"},
- {"Mic Bias 2V", NULL, "DMic"},
-};
-
-static const struct snd_soc_dapm_route audio_mapb[] = {
- {"b LINE2R", NULL, "MONO_LOUT"},
- {"Earphone", NULL, "b HPLOUT"},
-
- {"LINE1L", NULL, "b Mic Bias 2.5V"},
- {"b Mic Bias 2.5V", NULL, "HS Mic"}
-};
-
-static const char *spk_function[] = {"Off", "On"};
-static const char *input_function[] = {"ADC", "Digital Mic"};
-static const char *jack_function[] = {"Off", "TV-OUT", "Headphone", "Headset"};
-
-static const struct soc_enum rx51_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
-};
-
-static const struct snd_kcontrol_new aic34_rx51_controls[] = {
- SOC_ENUM_EXT("Speaker Function", rx51_enum[0],
- rx51_get_spk, rx51_set_spk),
- SOC_ENUM_EXT("Input Select", rx51_enum[1],
- rx51_get_input, rx51_set_input),
- SOC_ENUM_EXT("Jack Function", rx51_enum[2],
- rx51_get_jack, rx51_set_jack),
- SOC_DAPM_PIN_SWITCH("FM Transmitter"),
-};
-
-static const struct snd_kcontrol_new aic34_rx51_controlsb[] = {
- SOC_DAPM_PIN_SWITCH("Earphone"),
-};
-
-static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
-
- /* Set up NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "MIC3L");
- snd_soc_dapm_nc_pin(dapm, "MIC3R");
- snd_soc_dapm_nc_pin(dapm, "LINE1R");
-
- /* Add RX-51 specific controls */
- err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls,
- ARRAY_SIZE(aic34_rx51_controls));
- if (err < 0)
- return err;
-
- /* Add RX-51 specific widgets */
- snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets,
- ARRAY_SIZE(aic34_dapm_widgets));
-
- /* Set up RX-51 specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- err = tpa6130a2_add_controls(codec);
- if (err < 0)
- return err;
- snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
-
- err = omap_mcbsp_st_add_controls(rtd);
- if (err < 0)
- return err;
-
- /* AV jack detection */
- err = snd_soc_jack_new(codec, "AV Jack",
- SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
- &rx51_av_jack);
- if (err)
- return err;
- err = snd_soc_jack_add_gpios(&rx51_av_jack,
- ARRAY_SIZE(rx51_av_jack_gpios),
- rx51_av_jack_gpios);
-
- return err;
-}
-
-static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
-{
- int err;
-
- err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb,
- ARRAY_SIZE(aic34_rx51_controlsb));
- if (err < 0)
- return err;
-
- err = snd_soc_dapm_new_controls(dapm, aic34_dapm_widgetsb,
- ARRAY_SIZE(aic34_dapm_widgetsb));
- if (err < 0)
- return 0;
-
- return snd_soc_dapm_add_routes(dapm, audio_mapb,
- ARRAY_SIZE(audio_mapb));
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link rx51_dai[] = {
- {
- .name = "TLV320AIC34",
- .stream_name = "AIC34",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "tlv320aic3x-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "tlv320aic3x-codec.2-0018",
- .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = rx51_aic34_init,
- .ops = &rx51_ops,
- },
-};
-
-static struct snd_soc_aux_dev rx51_aux_dev[] = {
- {
- .name = "TLV320AIC34b",
- .codec_name = "tlv320aic3x-codec.2-0019",
- .init = rx51_aic34b_init,
- },
-};
-
-static struct snd_soc_codec_conf rx51_codec_conf[] = {
- {
- .dev_name = "tlv320aic3x-codec.2-0019",
- .name_prefix = "b",
- },
-};
-
-/* Audio card */
-static struct snd_soc_card rx51_sound_card = {
- .name = "RX-51",
- .owner = THIS_MODULE,
- .dai_link = rx51_dai,
- .num_links = ARRAY_SIZE(rx51_dai),
- .aux_dev = rx51_aux_dev,
- .num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
- .codec_conf = rx51_codec_conf,
- .num_configs = ARRAY_SIZE(rx51_codec_conf),
-};
-
-static struct platform_device *rx51_snd_device;
-
-static int __init rx51_soc_init(void)
-{
- int err;
-
- if (!machine_is_nokia_rx51())
- return -ENODEV;
-
- err = gpio_request_one(RX51_TVOUT_SEL_GPIO,
- GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel");
- if (err)
- goto err_gpio_tvout_sel;
- err = gpio_request_one(RX51_ECI_SW_GPIO,
- GPIOF_DIR_OUT | GPIOF_INIT_HIGH, "eci_sw");
- if (err)
- goto err_gpio_eci_sw;
-
- rx51_snd_device = platform_device_alloc("soc-audio", -1);
- if (!rx51_snd_device) {
- err = -ENOMEM;
- goto err1;
- }
-
- platform_set_drvdata(rx51_snd_device, &rx51_sound_card);
-
- err = platform_device_add(rx51_snd_device);
- if (err)
- goto err2;
-
- return 0;
-err2:
- platform_device_put(rx51_snd_device);
-err1:
- gpio_free(RX51_ECI_SW_GPIO);
-err_gpio_eci_sw:
- gpio_free(RX51_TVOUT_SEL_GPIO);
-err_gpio_tvout_sel:
-
- return err;
-}
-
-static void __exit rx51_soc_exit(void)
-{
- snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
- rx51_av_jack_gpios);
-
- platform_device_unregister(rx51_snd_device);
- gpio_free(RX51_ECI_SW_GPIO);
- gpio_free(RX51_TVOUT_SEL_GPIO);
-}
-
-module_init(rx51_soc_init);
-module_exit(rx51_soc_exit);
-
-MODULE_AUTHOR("Nokia Corporation");
-MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
-MODULE_LICENSE("GPL");
diff --git a/ANDROID_3.4.5/sound/soc/omap/sdp3430.c b/ANDROID_3.4.5/sound/soc/omap/sdp3430.c
deleted file mode 100644
index 0e283226..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/sdp3430.c
+++ /dev/null
@@ -1,279 +0,0 @@
-/*
- * sdp3430.c -- SoC audio for TI OMAP3430 SDP
- *
- * Author: Misael Lopez Cruz <x0052729@ti.com>
- *
- * Based on:
- * Author: Steve Sakoman <steve@sakoman.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <linux/i2c/twl.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-/* Register descriptions for twl4030 codec part */
-#include <linux/mfd/twl4030-audio.h>
-#include <linux/module.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-/* TWL4030 PMBR1 Register */
-#define TWL4030_INTBR_PMBR1 0x0D
-/* TWL4030 PMBR1 Register GPIO6 mux bit */
-#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2)
-
-static struct snd_soc_card snd_soc_sdp3430;
-
-static int sdp3430_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops sdp3430_ops = {
- .hw_params = sdp3430_hw_params,
-};
-
-/* Headset jack */
-static struct snd_soc_jack hs_jack;
-
-/* Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
- {
- .pin = "Headset Stereophone",
- .mask = SND_JACK_HEADPHONE,
- },
-};
-
-/* Headset jack detection gpios */
-static struct snd_soc_jack_gpio hs_jack_gpios[] = {
- {
- .gpio = (OMAP_MAX_GPIO_LINES + 2),
- .name = "hsdet-gpio",
- .report = SND_JACK_HEADSET,
- .debounce_time = 200,
- },
-};
-
-/* SDP3430 machine DAPM */
-static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Ext Mic", NULL),
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- /* External Mics: MAINMIC, SUBMIC with bias*/
- {"MAINMIC", NULL, "Mic Bias 1"},
- {"SUBMIC", NULL, "Mic Bias 2"},
- {"Mic Bias 1", NULL, "Ext Mic"},
- {"Mic Bias 2", NULL, "Ext Mic"},
-
- /* External Speakers: HFL, HFR */
- {"Ext Spk", NULL, "HFL"},
- {"Ext Spk", NULL, "HFR"},
-
- /* Headset Mic: HSMIC with bias */
- {"HSMIC", NULL, "Headset Mic Bias"},
- {"Headset Mic Bias", NULL, "Headset Mic"},
-
- /* Headset Stereophone (Headphone): HSOL, HSOR */
- {"Headset Stereophone", NULL, "HSOL"},
- {"Headset Stereophone", NULL, "HSOR"},
-};
-
-static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- /* SDP3430 connected pins */
- snd_soc_dapm_enable_pin(dapm, "Ext Mic");
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
-
- /* TWL4030 not connected pins */
- snd_soc_dapm_nc_pin(dapm, "AUXL");
- snd_soc_dapm_nc_pin(dapm, "AUXR");
- snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
-
- snd_soc_dapm_nc_pin(dapm, "OUTL");
- snd_soc_dapm_nc_pin(dapm, "OUTR");
- snd_soc_dapm_nc_pin(dapm, "EARPIECE");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
- snd_soc_dapm_nc_pin(dapm, "CARKITL");
- snd_soc_dapm_nc_pin(dapm, "CARKITR");
-
- /* Headset jack detection */
- ret = snd_soc_jack_new(codec, "Headset Jack",
- SND_JACK_HEADSET, &hs_jack);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
- hs_jack_pins);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
-
- return ret;
-}
-
-static int sdp3430_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- unsigned short reg;
-
- /* Enable voice interface */
- reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
- reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
- codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg);
-
- return 0;
-}
-
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp3430_dai[] = {
- {
- .name = "TWL4030 I2S",
- .stream_name = "TWL4030 Audio",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = sdp3430_twl4030_init,
- .ops = &sdp3430_ops,
- },
- {
- .name = "TWL4030 PCM",
- .stream_name = "TWL4030 Voice",
- .cpu_dai_name = "omap-mcbsp.3",
- .codec_dai_name = "twl4030-voice",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = sdp3430_twl4030_voice_init,
- .ops = &sdp3430_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_sdp3430 = {
- .name = "SDP3430",
- .owner = THIS_MODULE,
- .dai_link = sdp3430_dai,
- .num_links = ARRAY_SIZE(sdp3430_dai),
-
- .dapm_widgets = sdp3430_twl4030_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(sdp3430_twl4030_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *sdp3430_snd_device;
-
-static int __init sdp3430_soc_init(void)
-{
- int ret;
- u8 pin_mux;
-
- if (!machine_is_omap_3430sdp())
- return -ENODEV;
- printk(KERN_INFO "SDP3430 SoC init\n");
-
- sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
- if (!sdp3430_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(sdp3430_snd_device, &snd_soc_sdp3430);
-
- /* Set TWL4030 GPIO6 as EXTMUTE signal */
- twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
- TWL4030_INTBR_PMBR1);
- pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
- pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
- twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
- TWL4030_INTBR_PMBR1);
-
- ret = platform_device_add(sdp3430_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(sdp3430_snd_device);
-
- return ret;
-}
-module_init(sdp3430_soc_init);
-
-static void __exit sdp3430_soc_exit(void)
-{
- snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
-
- platform_device_unregister(sdp3430_snd_device);
-}
-module_exit(sdp3430_soc_exit);
-
-MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC SDP3430");
-MODULE_LICENSE("GPL");
-
diff --git a/ANDROID_3.4.5/sound/soc/omap/zoom2.c b/ANDROID_3.4.5/sound/soc/omap/zoom2.c
deleted file mode 100644
index 920e0d9e..00000000
--- a/ANDROID_3.4.5/sound/soc/omap/zoom2.c
+++ /dev/null
@@ -1,219 +0,0 @@
-/*
- * zoom2.c -- SoC audio for Zoom2
- *
- * Author: Misael Lopez Cruz <x0052729@ti.com>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * 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., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <mach/board-zoom.h>
-#include <plat/mcbsp.h>
-
-/* Register descriptions for twl4030 codec part */
-#include <linux/mfd/twl4030-audio.h>
-#include <linux/module.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15)
-
-static int zoom2_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- /* Set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
- SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops zoom2_ops = {
- .hw_params = zoom2_hw_params,
-};
-
-/* Zoom2 machine DAPM */
-static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Ext Mic", NULL),
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
- SND_SOC_DAPM_LINE("Aux In", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- /* External Mics: MAINMIC, SUBMIC with bias*/
- {"MAINMIC", NULL, "Mic Bias 1"},
- {"SUBMIC", NULL, "Mic Bias 2"},
- {"Mic Bias 1", NULL, "Ext Mic"},
- {"Mic Bias 2", NULL, "Ext Mic"},
-
- /* External Speakers: HFL, HFR */
- {"Ext Spk", NULL, "HFL"},
- {"Ext Spk", NULL, "HFR"},
-
- /* Headset Stereophone: HSOL, HSOR */
- {"Headset Stereophone", NULL, "HSOL"},
- {"Headset Stereophone", NULL, "HSOR"},
-
- /* Headset Mic: HSMIC with bias */
- {"HSMIC", NULL, "Headset Mic Bias"},
- {"Headset Mic Bias", NULL, "Headset Mic"},
-
- /* Aux In: AUXL, AUXR */
- {"Aux In", NULL, "AUXL"},
- {"Aux In", NULL, "AUXR"},
-};
-
-static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* TWL4030 not connected pins */
- snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
- snd_soc_dapm_nc_pin(dapm, "EARPIECE");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
- snd_soc_dapm_nc_pin(dapm, "CARKITL");
- snd_soc_dapm_nc_pin(dapm, "CARKITR");
-
- return 0;
-}
-
-static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- unsigned short reg;
-
- /* Enable voice interface */
- reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
- reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
- codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg);
-
- return 0;
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link zoom2_dai[] = {
- {
- .name = "TWL4030 I2S",
- .stream_name = "TWL4030 Audio",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = zoom2_twl4030_init,
- .ops = &zoom2_ops,
- },
- {
- .name = "TWL4030 PCM",
- .stream_name = "TWL4030 Voice",
- .cpu_dai_name = "omap-mcbsp.3",
- .codec_dai_name = "twl4030-voice",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = zoom2_twl4030_voice_init,
- .ops = &zoom2_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_zoom2 = {
- .name = "Zoom2",
- .owner = THIS_MODULE,
- .dai_link = zoom2_dai,
- .num_links = ARRAY_SIZE(zoom2_dai),
-
- .dapm_widgets = zoom2_twl4030_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(zoom2_twl4030_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *zoom2_snd_device;
-
-static int __init zoom2_soc_init(void)
-{
- int ret;
-
- if (!machine_is_omap_zoom2())
- return -ENODEV;
- printk(KERN_INFO "Zoom2 SoC init\n");
-
- zoom2_snd_device = platform_device_alloc("soc-audio", -1);
- if (!zoom2_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(zoom2_snd_device, &snd_soc_zoom2);
- ret = platform_device_add(zoom2_snd_device);
- if (ret)
- goto err1;
-
- BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
- gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
-
- BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
- gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(zoom2_snd_device);
-
- return ret;
-}
-module_init(zoom2_soc_init);
-
-static void __exit zoom2_soc_exit(void)
-{
- gpio_free(ZOOM2_HEADSET_MUX_GPIO);
- gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
-
- platform_device_unregister(zoom2_snd_device);
-}
-module_exit(zoom2_soc_exit);
-
-MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC Zoom2");
-MODULE_LICENSE("GPL");
-