January 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/30] ASoC: hdac_hdmi: Add support to enable MST audio
by jeeja.kp＠intel.com 15 Mar '17

15 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> Display port 1.2 introduces new capability Multi-stream transport (MST) which will allow multiple stream to route to single display port where multiple monitors are connected. Multiple monitors will be connected by a MST Hub or a monitor capable of daisy-chaining. With MST support, a pin can support multiple ports and on each port a monitor can be connected. This patch series enables DP MST Audio by extending the current design from pin to port. o With MST, pin mux for CVT selection will be used to select CVT for pin-port instead of pin. So create the pin mux accordingly. o Pin is marked as MST capable when connect event reports pipe as valid. if pipe is -1, then pin is not MST capable and default to port 0. if pipe is valid, then pin is MST capable and port = pipe. o For MST capable pin, select the device entry before configuring the pin widget verbs controlling the sink device operations. o Add support to handle multiple Port mapped to same converter by creating as port list in pcm and report jack event based on the no of ports connected/disconnected . o i915 acomp APIs are now used to read the ELD info and support is extended for MST port as well. o Configure port/pin/audio infoframe in widget event handlers for dynamic routing when monitor is connected/disconnected instead of DAI ops. o Use set_tdm slot to set the stream tag instead of dai params and store it in pcm context and use this in configuring the converter in widget event handler. o Use pin widget to represent the output to enable/disable path when a device is connected/disconnected. Added API to create the pin widget and this will be called by the machine driver after jack creation. o Add channel map support for bxt machine. o Move from snd_jack to ASOC jack framework, create the jack in machine driver. o Create port pin widget to enable and disable the path dynamically when a jack is connected/disconnected. Note: This patch series has dependency on the series "ASoC: Intel: Skylake: Driver updates" Jeeja KP (29): ASoC: hdac_hdmi: Register widget event handlers ASoC: Intel: Skylake: Use set_tdm_slot to set the dma channel ASoC: hdac_hdmi: Move channel info from pin to PCM structure ASoC: Intel: bxt: add channel map support in rt298 machine ASoC: Intel: bxt: add channel map support in bxt_da7219_max98357a machine ASoC: hdac_hdmi: Begin to add support for DP Multi-stream audio ASoC: Intel: Skylake: Add route change to rt286 machine ASoC: Intel: Skylake: Add route change to nau88l25_max98357a machine ASoC: Intel: Skylake: Add route change to nau88l25_ssm4567 machine ASoC: Intel: bxt: Add route change to rt298 machine ASoC: Intel: bxt: Add route change to da7219_max98357a machine ASoC: hdac_hdmi: Add support to handle MST capable pin ASoC: hdac_hdmi: Add MST verb support ASoC: hdac_hdmi: Handle MST pin jack detection at boot/resume ASoc: hdac_hdmi: Configure pin verbs for MST ASoC: hdac_hdmi: Add support for multiple ports to a PCM ASoC: hdac_hdmi: Use ASoC jack instead of snd_jack ASoC: Intel: Skylake: Create ASoC jack for hdmi in rt286 machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in skl_nau88l25_max98357a machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in skl_nau88l25_ssm4567 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_rt298 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_da7219_max98357 machine ASoC: hdac_hdmi: Add machine pin widget for each port ASoC: Intel: Skylake: Add jack port initialize in rt286 machine ASoC: Intel: Skylake: Add jack port initialize in skl_nau88l25_max98357a machine ASoC: Intel: Skylake: Add jack port initialize in skl_nau88l25_ssm4567 machine ASoC: Intel: Skylake: Add jack port initialize in bxt_rt298 machine ASoC: Intel: Skylake: Add jack port initialize in bxt_da7219_max98357a machine ASoC: Intel: Skylake: Fix to delete DSP pipe after stopping pipe Sandeep Tayal (1): ASoC: hdac_hdmi: use audio component framework to read ELD sound/soc/codecs/hdac_hdmi.c | 1331 ++++++++++++++--------- sound/soc/codecs/hdac_hdmi.h | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 75 +- sound/soc/intel/boards/bxt_rt298.c | 78 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 33 +- sound/soc/intel/boards/skl_rt286.c | 30 +- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 10 +- 9 files changed, 1045 insertions(+), 553 deletions(-) -- 2.5.0

3 54

[alsa-devel] [PATCH] ASoC: rt286: fix headphone click/crack noise on Dell XPS 9343 I2S mode
by Kai-Heng Feng 13 Mar '17

13 Mar '17

HDA mode does not have these issues because necessary workarounds in linux/sound/pci/hda/patch_realtek.c are already applied. So we can leverage these workaournds into rt286. When jack is plugged, it rapidly generates I2C interrupts, which triggers rt286_irq() and rt286_jack_detect(), which causes the noise. alc_fixup_dell_xps13() in patch_realtek.c sets up a pin that can stop the frantic interrupts, hence fixes the click noise. When rt286 is under powersaving state, play a sound with headphone or plug a headphone in will produce a loud crack sound. alc_shutup_dell_xps13() patch_realtek.c mutes the headphone at plugging. Apply the same workaround to LDO2 power event can make the loud crack sound to a more tolerable pop sound. I found that unconditionally apply the workaround to all related power event can help a little further. Link: https://bugzilla.kernel.org/show_bug.cgi?id=112611 Link: https://bugzilla.redhat.com/show_bug.cgi?id=1313434 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/soc/codecs/rt286.c | 67 +++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 63 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 74c0e4e..d041937 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,6 +36,8 @@ #define RT286_VENDOR_ID 0x10ec0286 #define RT288_VENDOR_ID 0x10ec0288 +#define AMP_OUT_MUTE 0xb080 + struct rt286_priv { struct reg_default *index_cache; int index_cache_size; @@ -47,6 +49,7 @@ struct rt286_priv { struct delayed_work jack_detect_work; int sys_clk; int clk_id; + void (*mute_hpo_func)(struct snd_soc_codec *codec); }; static const struct reg_default rt286_index_def[] = { @@ -472,10 +475,51 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } +/* Add HV, VREF and LDO1 event functions to workaround headphone crack noise */ +static int rt286_hv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); + + return 0; +} + +static int rt286_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); + + return 0; +} + +static int rt286_ldo1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); + + return 0; +} + static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -516,16 +560,24 @@ static int rt286_mic1_event(struct snd_soc_dapm_widget *w, return 0; } +static void dell_dino_mute_hpo(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_MUTE); +} + static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1, - 12, 1, NULL, 0), + 12, 1, rt286_hv_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1, - 0, 1, NULL, 0), + 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2, - 2, 0, NULL, 0), + 2, 0, rt286_ldo1_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1, 13, 1, rt286_ldo2_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MCLK MODE", RT286_PLL_CTRL1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, @@ -1190,6 +1242,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { + /* Fix headphone click noise */ + if (dmi_check_system(dmi_dell_dino)) + regmap_write(rt286->regmap, + RT286_MIC1_DET_CTRL, 0x0020); + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0x5000); } @@ -1204,6 +1261,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + /* Power down LDO, VREF */ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0); regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); @@ -1222,6 +1280,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, RT286_SET_GPIO_DATA, 0x40, 0x40); regmap_update_bits(rt286->regmap, RT286_GPIO_CTRL, 0xc, 0x8); + rt286->mute_hpo_func = dell_dino_mute_hpo; } if (rt286->i2c->irq) { -- 2.10.0

3 7

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Driver updates on DSP firmware
by jeeja.kp＠intel.com 07 Mar '17

07 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides update on DSP firmware download by optimizing o updating the dsp register poll method to use accurate timeout. o optimizing ROM init retries. o saving the firmware/library context at boot and use this ctx for downloading the firmware to DSP memory. o cleanup to store the library manifest data o release the firmware in cleanup routine. Jeeja KP (6): ASoC: Intel: Common: Update dsp register poll implementation ASoC: Intel: bxtn: Use DSP poll API to poll FW status ASoC: Intel: Skylake: Clean up manifest info ASoC: Intel: Skylake: Release FW ctx in cleanup ASoC: Intel: bxtn: Fix to store the FW/Library context at boot ASoC: Intel: bxtn: optimize ROM init retries sound/soc/intel/common/sst-dsp.c | 52 +++++----- sound/soc/intel/skylake/bxt-sst.c | 142 ++++++++++++++------------- sound/soc/intel/skylake/skl-sst-dsp.h | 4 +- sound/soc/intel/skylake/skl-sst-ipc.h | 5 +- sound/soc/intel/skylake/skl-sst.c | 3 + sound/soc/intel/skylake/skl-topology.c | 41 ++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++ sound/soc/intel/skylake/skl-tplg-interface.h | 12 --- 8 files changed, 137 insertions(+), 135 deletions(-) -- 2.5.0

4 15

[alsa-devel] [PATCH] alsa-lib: Add htimestamp operation in plugin file
by sutar.mounesh＠gmail.com 27 Feb '17

27 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> PCM operation htimestamp is not implemented in plugin file. Calling snd_pcm_htimestamp() on a plugin file crashes. This scenario is considered now. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> --- --- a/src/pcm/pcm_file.c 2013-07-08 14:31:36.000000000 +0200 +++ b/src/pcm/pcm_file.c 2015-05-04 16:26:10.413615403 +0200 @@ -698,6 +698,7 @@ .readi = snd_pcm_file_readi, .readn = snd_pcm_file_readn, .avail_update = snd_pcm_generic_avail_update, + .htimestamp = snd_pcm_generic_htimestamp, .mmap_commit = snd_pcm_file_mmap_commit, .poll_descriptors_count = snd_pcm_generic_poll_descriptors_count, .poll_descriptors = snd_pcm_generic_poll_descriptors,

4 8

[alsa-devel] [PATCH] pcm_rate: Do not discard slave reported delay in status result.
by Alan Young 21 Feb '17

21 Feb '17

Similar to recent dshare patch.

3 31

[alsa-devel] New platform CherryTrail/ES8316 audio output is silent
by Daniel Drake 20 Feb '17

20 Feb '17

Hi, I'm trying to get audio working on the Weibu F3C miniPC which has a 3.5mm headphone jack and 3.5mm mic jack. The SoC is Intel Cherry Trail x5-Z8300. The ES8316 codec included is not supported by Linux at the moment. I'd like to fix that and upstream the driver but so far am I stuck with no audio output, only silence. I do have a working Windows setup that I can poke at for more info. I would appreciate any suggestions or ideas. Here is a description of my efforts so far, followed by the work in progress code: I started with the Cherry Trail ES8316 platform driver + ES8316 codec driver which I found in https://github.com/JideTechnology/remixos-kernel/tree/jide_chuwi_vi10_plus_… First I did the required updates to get it building on Linux 4.9 including adding the driver for the codec's MCLK input which is done with some patches currently under review: clk: x86: Add Atom PMC platform clocks arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 platform/x86: Enable Atom PMC platform clocks I confirmed that I am manipulating the right clock by disabling it under Windows and observing that audio output suddenly stops. I created an audio loop within the codec itself, by enabling MIC1 as an input to the Left HP Mux, enabling the LLIN Switch, and enabling Left HP Mux as an input to Left Hp Mixer. Now when I make noise in the microphone, I can hear it on the headphone. This verifies that most of the audio path through the codec is working fine, and also that it is clocked. I used asoc debugfs to trace the path through the DAI, platform and codec DAPM graphs. Everything is On and active as expected. I confirmed that it is basically the same as another Cherry Trail platform I have here (except for the codec which is rt5645 on the other platform), including mixer levels. I also dumped the codec registers over I2C from Windows and set them to the exact same values on Linux. Still not working. At this point I am suspecting that this is likely a problem with the audio interface and its linkage to the codec (SSP/I2S). I modified the platform driver to use ssp1 instead of ssp2, because the device schematics say that I2S_1 is being used. Then I ran into send_ssp_cmd() complaining that ssp1 is not supported, and found a point of curiosity here. This code is basically to set the value of sst_cmd_sba_hw_set_ssp.selection which is documented as: /* 0:SSP0(def), 1:SSP1, 2:SSP2 */ But other platforms like cht_bsw_rt5645 use ssp2, for which send_ssp_cmd() will use SSP_CODEC, which has value 3. So it looks like the above comment is wrong? Either way, I tried selection values 0, 1, 2, and 3, and still no audio output. Here's the code, apologies that it is a bit unclean, I'll continue improving it once I get something working. Any suggestions, or do I need to get a hardware scope to diganose further? --- drivers/base/regmap/regmap-debugfs.c | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8316.c | 1386 ++++++++++++++++++++++++++++ sound/soc/codecs/es8316.h | 124 +++ sound/soc/intel/Kconfig | 12 + sound/soc/intel/atom/sst-atom-controls.c | 5 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cht_es8316.c | 1462 ++++++++++++++++++++++++++++++ 10 files changed, 2999 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h create mode 100644 sound/soc/intel/boards/cht_es8316.c diff --git a/drivers/base/regmap/regmap-debugfs.c b/drivers/base/regmap/regmap-debugfs.c index 36ce351..6d3dc14 100644 --- a/drivers/base/regmap/regmap-debugfs.c +++ b/drivers/base/regmap/regmap-debugfs.c @@ -269,7 +269,7 @@ static ssize_t regmap_map_read_file(struct file *file, char __user *user_buf, count, ppos); } -#undef REGMAP_ALLOW_WRITE_DEBUGFS +#define REGMAP_ALLOW_WRITE_DEBUGFS #ifdef REGMAP_ALLOW_WRITE_DEBUGFS /* * This can be dangerous especially when we have clients such as diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..c74846c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_GTM601 @@ -512,6 +513,9 @@ config SND_SOC_HDMI_CODEC select SND_PCM_IEC958 select HDMI +config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 958cd49..63a9329 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -61,6 +61,7 @@ snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -286,6 +287,7 @@ obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 0000000..6566895 --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,1386 @@ +/* + * es8316.c -- es8316 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.com> + * + * Based on es8316.c + * 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. + */ +#define DEBUG +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/of_gpio.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <linux/proc_fs.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/regmap.h> +#include "es8316.h" + +#if 1 +#define DBG(x...) printk(x) +#else +#define DBG(x...) do { } while (0) +#endif +#define alsa_dbg DBG + +#define dmic_used 1 +#define amic_used 0 + +#define INVALID_GPIO -1 +int es8316_spk_con_gpio = INVALID_GPIO; +int es8316_hp_con_gpio = INVALID_GPIO; +int es8316_hp_det_gpio = INVALID_GPIO; +//static int HP_IRQ=0; +//static int mutex_status=0; +//static int hp_irq_flag = 0; +int es8316_init_reg = 0; + +#define GPIO_LOW 0 +#define GPIO_HIGH 1 +#ifndef es8316_DEF_VOL +#define es8316_DEF_VOL 0x1e +#endif + +struct snd_soc_codec *es8316_codec; +static int es8316_init_regs(struct snd_soc_codec *codec); +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); + +static const struct reg_default es8316_reg_defaults[] = { + {0x00, 0x03}, {0x01 ,0x03}, {0x02, 0x00}, {0x03, 0x20}, + {0x04, 0x11}, {0x05, 0x00}, {0x06, 0x11}, {0x07, 0x00}, + {0x08, 0x00}, {0x09, 0x01}, {0x0a, 0x00}, {0x0b, 0x00}, + {0x0c, 0xf8}, {0x0d, 0x3f}, {0x0e, 0x00}, {0x0f, 0x00}, + {0x10, 0x01}, {0x11, 0xfc}, {0x12, 0x28}, {0x13, 0x00}, + {0x14, 0x00}, {0x15, 0x33}, {0x16, 0x00}, {0x17, 0x00}, + {0x18, 0x88}, {0x19, 0x07}, {0x1a, 0x22}, {0x1b, 0x03}, + {0x1c, 0x0f}, {0x1d, 0x00}, {0x1e, 0x80}, {0x1f, 0x80}, + {0x20, 0x00}, {0x21, 0x00}, {0x22, 0xc0}, {0x23, 0x00}, + {0x24, 0x01}, {0x25, 0x08}, {0x26, 0x10}, {0x27, 0xc0}, + {0x28, 0x00}, {0x29, 0x1c}, {0x2a, 0x00}, {0x2b, 0xb0}, + {0x2c, 0x32}, {0x2d, 0x03}, {0x2e, 0x00}, {0x2f, 0x11}, + {0x30, 0x10}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0xc0}, + {0x34, 0xc0}, {0x35, 0x1f}, {0x36, 0xf7}, {0x37, 0xfd}, + {0x38, 0xff}, {0x39, 0x1f}, {0x3a, 0xf7}, {0x3b, 0xfd}, + {0x3c, 0xff}, {0x3d, 0x1f}, {0x3e, 0xf7}, {0x3f, 0xfd}, + {0x40, 0xff}, {0x41, 0x1f}, {0x42, 0xf7}, {0x43, 0xfd}, + {0x44, 0xff}, {0x45, 0x1f}, {0x46, 0xf7}, {0x47, 0xfd}, + {0x48, 0xff}, {0x49, 0x1f}, {0x4a, 0xf7}, {0x4b, 0xfd}, + {0x4c, 0xff}, {0x4d, 0x00}, {0x4e, 0x00}, {0x4f, 0xff}, + {0x50, 0x00}, {0x51, 0x00}, {0x52, 0x00}, {0x53, 0x00}, + }; + +static bool es8316_writeable(struct device *dev, unsigned int reg) +{ + if (reg <= 90) + return true; + else + return false; +} +static bool es8316_readable(struct device *dev, unsigned int reg) +{ + if (reg <= 90) + return true; + else + return false; +} +static bool es8316_volatile(struct device *dev, unsigned int reg) +{ + if (reg <= 90) + return true; + else + return false; +} +/* codec private data */ +struct es8316_priv { + struct regmap *regmap; + unsigned int dmic_amic; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct delayed_work pcm_pop_work; +}; +struct snd_soc_codec *tron_codec; +/* +* es8316_reset +* write value 0xff to reg0x00, the chip will be in reset mode +* then, writer 0x00 to reg0x00, unreset the chip +*/ +static int es8316_reset(struct snd_soc_codec *codec) +{ + /* snd_soc_write(codec, ES8316_RESET_REG00, 0x3F); */ + msleep(5); + /* return snd_soc_write(codec, ES8316_RESET_REG00, 0x03); */ + return 0; +} +/* +* es8316_hp_state_query +* Check the register 0x4F, return the register value +* User should update this code according the headphone jack configuratio +*/ + static int es8316_hp_state_query(struct snd_soc_codec *codec) + { + unsigned int retv,tmp; + + retv = snd_soc_read(codec, + 0); + tmp = retv & 0x06; + + return(tmp); + } +/* + * es8316S Controls + */ +//#define DECLARE_TLV_DB_SCALE(name, min, step, mute) +//static const DECLARE_TLV_DB_SCALE(hpout_vol_tlv, -4800, 1200, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(mic_bst_tlv, 0, 1200, 0); +//static const DECLARE_TLV_DB_SCALE(linin_pga_tlv, 0, 300, 0); +/* {0, +3, +6, +9, +12, +15, +18, +21, +24,+27,+30,+33} dB */ +static unsigned int linin_pga_tlv[] = { + TLV_DB_RANGE_HEAD(12), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(600, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(900, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(1200, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1500, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1800, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(2100, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(2400, 0, 0), +}; +static unsigned int hpout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 3, TLV_DB_SCALE_ITEM(-4800, 1200, 0), + +}; +static const char *alc_func_txt[] = {"Off", "On"}; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(ES8316_ADC_ALC1_REG29, 6, 2, alc_func_txt); + +static const char *ng_type_txt[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(ES8316_ADC_ALC6_REG2E, 6, 2, ng_type_txt); + +static const char *adcpol_txt[] = {"Normal", "Invert"}; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(ES8316_ADC_MUTE_REG26, 1, 2, adcpol_txt); +static const char *dacpol_txt[] = {"Normal", "R Invert", "L Invert", + "L + R Invert"}; +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8316_DAC_SET1_REG30, 0, 4, dacpol_txt); + +static const struct snd_kcontrol_new es8316_snd_controls[] = { + /* HP OUT VOLUME */ + SOC_DOUBLE_TLV("HP Playback Volume", ES8316_CPHP_ICAL_VOL_REG18, + 4, 0, 0, 1, hpout_vol_tlv), + /* HPMIXER VOLUME Control */ + SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL_REG16, + 0, 4, 7, 0, hpmixer_gain_tlv), + + /* DAC Digital controls */ + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL_REG33, + ES8316_DAC_VOLR_REG34, 0, 0xC0, 1, dac_vol_tlv), + + SOC_SINGLE("Enable DAC Soft Ramp", ES8316_DAC_SET1_REG30, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1_REG30, 2, 4, 0), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE("DAC Notch Filter", ES8316_DAC_SET2_REG31, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Mode", ES8316_DAC_SET2_REG31, 7, 1, 0), + SOC_SINGLE("DAC Volume Control-LeR", ES8316_DAC_SET2_REG31, 2, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3_REG32, 0, 7, 0), + + /* +20dB D2SE PGA Control */ + SOC_SINGLE_TLV("MIC Boost", ES8316_ADC_D2SEPGA_REG24, + 0, 1, 0, mic_bst_tlv), + /* 0-+24dB Lineinput PGA Control */ + SOC_SINGLE_TLV("Input PGA", ES8316_ADC_PGAGAIN_REG23, + 4, 8, 0, linin_pga_tlv), + + /* ADC Digital Control */ + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME_REG27, + 0, 0xC0, 1, adc_vol_tlv), + SOC_SINGLE("ADC Soft Ramp", ES8316_ADC_MUTE_REG26, 4, 1, 0), + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("ADC Double FS Mode", ES8316_ADC_DMIC_REG25, 4, 1, 0), + /* ADC ALC Control */ + SOC_SINGLE("ALC Capture Target Volume", ES8316_ADC_ALC3_REG2B, 4, 10, 0), + SOC_SINGLE("ALC Capture Max PGA", ES8316_ADC_ALC1_REG29, 0, 28, 0), + SOC_SINGLE("ALC Capture Min PGA", ES8316_ADC_ALC2_REG2A, 0, 28, 0), + SOC_ENUM("ALC Capture Function", alc_func), + SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3_REG2B, 0, 10, 0), + SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4_REG2C, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4_REG2C, 0, 10, 0), + SOC_SINGLE("ALC Capture NG Threshold", ES8316_ADC_ALC6_REG2E, 0, 31, 0), + SOC_ENUM("ALC Capture NG Type", ng_type), + SOC_SINGLE("ALC Capture NG Switch", ES8316_ADC_ALC6_REG2E, 5, 1, 0), +}; + +/* Analog Input MUX */ +static const char * const es8316_analog_in_txt[] = { + "lin1-rin1", + "lin2-rin2", + "lin1-rin1 with 20db Boost", + "lin2-rin2 with 20db Boost" + }; +static const unsigned int es8316_analog_in_values[] = { + 0,/*1,*/ + 1, + 2, + 3 + }; +static const struct soc_enum es8316_analog_input_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL_REG22, 4, 3, + ARRAY_SIZE(es8316_analog_in_txt), + es8316_analog_in_txt, + es8316_analog_in_values); +static const struct snd_kcontrol_new es8316_analog_in_mux_controls = + SOC_DAPM_ENUM("Route", es8316_analog_input_enum); + +/* Dmic MUX */ +static const char * const es8316_dmic_txt[] = { + "dmic disable", + "dmic data at high level", + "dmic data at low level", + }; +static const unsigned int es8316_dmic_values[] = { + 0,/*1,*/ + 1, + 2 + }; +static const struct soc_enum es8316_dmic_src_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC_REG25, 0, 3, + ARRAY_SIZE(es8316_dmic_txt), + es8316_dmic_txt, + es8316_dmic_values); +static const struct snd_kcontrol_new es8316_dmic_src_controls = + SOC_DAPM_ENUM("Route", es8316_dmic_src_enum); + +/* hp mixer mux */ +static const char *es8316_hpmux_texts[] = { + "lin1-rin1", + "lin2-rin2", + "lin-rin with Boost", + "lin-rin with Boost and PGA" + }; + +static const unsigned int es8316_hpmux_values[] = { + 0, 1, 2, 3}; + +static const struct soc_enum es8316_left_hpmux_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_HPMIX_SEL_REG13, 4, 7, + ARRAY_SIZE(es8316_hpmux_texts), + es8316_hpmux_texts, + es8316_hpmux_values); +static const struct snd_kcontrol_new es8316_left_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum); + +static const struct soc_enum es8316_right_hpmux_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_HPMIX_SEL_REG13, 0, 7, + ARRAY_SIZE(es8316_hpmux_texts), + es8316_hpmux_texts, + es8316_hpmux_values); +static const struct snd_kcontrol_new es8316_right_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum); + +/* headphone Output Mixer */ +static const struct snd_kcontrol_new es8316_out_left_mix[] = { + SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH_REG14, + 6, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", 0x30, + 5, 1, 0), +}; +static const struct snd_kcontrol_new es8316_out_right_mix[] = { + SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH_REG14, + 2, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", 0x30, + 5, 1, 0), +}; + +/* DAC data source mux */ +static const char *es8316_dacsrc_texts[] = { + "LDATA TO LDAC, RDATA TO RDAC", + "LDATA TO LDAC, LDATA TO RDAC", + "RDATA TO LDAC, RDATA TO RDAC", + "RDATA TO LDAC, LDATA TO RDAC", + }; + +static const unsigned int es8316_dacsrc_values[] = { + 0, 1, 2, 3}; + +static const struct soc_enum es8316_dacsrc_mux_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_DAC_SET1_REG30, 6, 4, + ARRAY_SIZE(es8316_dacsrc_texts), + es8316_dacsrc_texts, + es8316_dacsrc_values); +static const struct snd_kcontrol_new es8316_dacsrc_mux_controls = + SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum); + + +static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), +// SND_SOC_DAPM_MICBIAS("micbias", ES8316_SYS_PDN_REG0D, +// 5, 1), + + SND_SOC_DAPM_MICBIAS("micbias", SND_SOC_NOPM, + 0, 0), + /* Input MUX */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + SND_SOC_DAPM_PGA("Line input PGA", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* ADCs */ + //SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL_REG22, 6, 0), + SND_SOC_DAPM_ADC("Mono ADC", NULL, SND_SOC_NOPM, 0, 0), + + /* Dmic MUX */ + SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0, + &es8316_dmic_src_controls), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "Capture", 1, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "Playback", 0, + SND_SOC_NOPM, 0, 0), + + /* DACs DATA SRC MUX */ + SND_SOC_DAPM_MUX("DAC SRC Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + /* DACs */ + SND_SOC_DAPM_DAC("Right DAC", NULL, SND_SOC_NOPM, 0, 0), + + //SND_SOC_DAPM_DAC("Left DAC", NULL, ES8316_DAC_PDN_REG2F, 4, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, SND_SOC_NOPM, 0, 0), + + /* Headphone Output Side */ + /* hpmux for hp mixer */ + SND_SOC_DAPM_MUX("Left Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + /* Output mixer */ +/* SND_SOC_DAPM_MIXER("Left Hp mixer", ES8316_HPMIX_PDN_REG15, + 4, 1, &es8316_out_left_mix[0], ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", ES8316_HPMIX_PDN_REG15, + 0, 1, &es8316_out_right_mix[0], ARRAY_SIZE(es8316_out_right_mix)), */ + SND_SOC_DAPM_MIXER("Left Hp mixer", SND_SOC_NOPM, + 4, 1, &es8316_out_left_mix[0], ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", SND_SOC_NOPM, + 0, 1, &es8316_out_right_mix[0], ARRAY_SIZE(es8316_out_right_mix)), + + //SND_SOC_DAPM_MIXER("Left Hp mixer", 0x32, + // 5, 1, &es8316_out_left_mix[0], ARRAY_SIZE(es8316_out_left_mix)), + //SND_SOC_DAPM_MIXER("Right Hp mixer", 0x32, + // 4, 1, &es8316_out_right_mix[0], ARRAY_SIZE(es8316_out_right_mix)), + + /* Ouput charge pump */ +/* + SND_SOC_DAPM_PGA("HPCP L", 0x17, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPCP R", 0x17, + 2, 0, NULL, 0), +*/ + /* Ouput Driver */ + SND_SOC_DAPM_PGA("HPVOL L", SND_SOC_NOPM, + 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("HPVOL R", SND_SOC_NOPM, + 7, 1, NULL, 0), + /* Ouput Driver */ + SND_SOC_DAPM_PGA("SPKVOL L", SND_SOC_NOPM, + 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("SPKVOL R", SND_SOC_NOPM, + 7, 1, NULL, 0), + + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKOL"), + SND_SOC_DAPM_OUTPUT("SPKOR"), +}; + +static const struct snd_soc_dapm_route es8316_dapm_routes[] = { + /* + * record route map + */ + {"MIC1", NULL, "micbias"}, + {"MIC2", NULL, "micbias"}, + {"DMIC", NULL, "micbias"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "Line input PGA"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + {"Digital Mic Mux", "dmic data at high level", "DMIC"}, + {"Digital Mic Mux", "dmic data at low level", "DMIC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + /* + * playback route map + */ + {"DAC SRC Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + {"DAC SRC Mux", "LDATA TO LDAC, LDATA TO RDAC", "I2S IN"}, + {"DAC SRC Mux", "RDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + {"DAC SRC Mux", "RDATA TO LDAC, LDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC SRC Mux"}, + {"Right DAC", NULL, "DAC SRC Mux"}, + + + {"Left Hp mux", "lin1-rin1", "MIC1"}, + {"Left Hp mux", "lin2-rin2", "MIC2"}, + {"Left Hp mux", "lin-rin with Boost", "Differential Mux"}, + {"Left Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Right Hp mux", "lin1-rin1", "MIC1"}, + {"Right Hp mux", "lin2-rin2", "MIC2"}, + {"Right Hp mux", "lin-rin with Boost", "Differential Mux"}, + {"Right Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Hp mixer", "LLIN Switch", "Left Hp mux"}, + {"Left Hp mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Hp mixer", "RLIN Switch", "Right Hp mux"}, + {"Right Hp mixer", "Right DAC Switch", "Right DAC"}, + + {"HPVOL L", NULL, "Left Hp mixer"}, + {"HPVOL R", NULL, "Right Hp mixer"}, +/* + {"HPVOL L", NULL, "HPCP L"}, + {"HPVOL R", NULL, "HPCP R"}, +*/ + {"HPOL", NULL, "HPVOL L"}, + {"HPOR", NULL, "HPVOL R"}, + + {"SPKOL", NULL, "SPKVOL L"}, + {"SPKOR", NULL, "SPKVOL R"}, + + {"SPKVOL L", NULL, "Left Hp mixer"}, + {"SPKVOL R", NULL, "Right Hp mixer"}, +}; + +struct _coeff_div { + u32 mclk; //mclk frequency + u32 rate; //sample rate + u8 div; //adcclk and dacclk divider + u8 lrck_h; //adclrck divider and daclrck divider + u8 lrck_l; + u8 sr; //sclk divider + u8 osr; //adc osr +}; + + +/* codec hifi mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 8k */ + {12288000, 8000 , 6 , 0x06, 0x00, 21, 32}, + {11289600, 8000 , 6 , 0x05, 0x83, 20, 29}, + {18432000, 8000 , 9 , 0x09, 0x00, 27, 32}, + {16934400, 8000 , 8 , 0x08, 0x44, 25, 33}, + {12000000, 8000 , 7 , 0x05, 0xdc, 21, 25}, + {19200000, 8000 , 12, 0x09, 0x60, 27, 25}, + + /* 11.025k */ + {11289600, 11025, 4 , 0x04, 0x00, 16, 32}, + {16934400, 11025, 6 , 0x06, 0x00, 21, 32}, + {12000000, 11025, 4 , 0x04, 0x40, 17, 34}, + + /* 16k */ + {12288000, 16000, 3 , 0x03, 0x00, 12, 32}, + {18432000, 16000, 5 , 0x04, 0x80, 18, 25}, + {12000000, 16000, 3 , 0x02, 0xee, 12, 31}, + {19200000, 16000, 6 , 0x04, 0xb0, 18, 25}, + + /* 22.05k */ + {11289600, 22050, 2 , 0x02, 0x00, 8 , 32}, + {16934400, 22050, 3 , 0x03, 0x00, 12, 32}, + {12000000, 22050, 2 , 0x02, 0x20, 8 , 34}, + + /* 32k */ + {12288000, 32000, 1 , 0x01, 0x80, 6 , 48}, + {18432000, 32000, 2 , 0x02, 0x40, 9 , 32}, + {12000000, 32000, 1 , 0x01, 0x77, 6 , 31}, + {19200000, 32000, 3 , 0x02, 0x58, 10, 25}, + + /* 44.1k */ + {11289600, 44100, 1 , 0x01, 0x00, 4 , 32}, + {16934400, 44100, 1 , 0x01, 0x80, 6 , 32}, + {12000000, 44100, 1 , 0x01, 0x10, 4 , 34}, + + /* 48k */ + {12288000, 48000, 1 , 0x01, 0x00, 4 , 32}, + {18432000, 48000, 1 , 0x01, 0x80, 6 , 32}, + {12000000, 48000, 1 , 0x00, 0xfa, 4 , 31}, + {19200000, 48000, 2 , 0x01, 0x90, 6, 25}, + + /* 88.2k */ + {11289600, 88200, 1 , 0x00, 0x80, 2 , 32}, + {16934400, 88200, 1 , 0x00, 0xc0, 3 , 48}, + {12000000, 88200, 1 , 0x00, 0x88, 2 , 34}, + + /* 96k */ + {12288000, 96000, 1 , 0x00, 0x80, 2 , 32}, + {18432000, 96000, 1 , 0x00, 0xc0, 3 , 48}, + {12000000, 96000, 1 , 0x00, 0x7d, 1 , 31}, + {19200000, 96000, 1 , 0x00, 0xc8, 3 , 25}, +}; +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +/* The set of rates we can generate from the above for each SYSCLK */ + +static unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + +/* + * Note that this should be called from init rather than from hw_params. + */ +static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + + DBG("Enter::%s----%d, freq:%d\n",__FUNCTION__,__LINE__, freq); + + switch (freq) { + case 11289600: + case 18432000: + case 22579200: + case 36864000: + es8316->sysclk_constraints = &constraints_112896; + es8316->sysclk = freq; + return 0; + + case 12288000: + case 19200000: + case 16934400: + case 24576000: + case 33868800: + es8316->sysclk_constraints = &constraints_12288; + es8316->sysclk = freq; + return 0; + + case 12000000: + case 24000000: + es8316->sysclk_constraints = &constraints_12; + es8316->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface = 0; + u8 adciface = 0; + u8 daciface = 0; + + printk("%s----%d, fmt[%02x]\n",__FUNCTION__,__LINE__,fmt); + + iface = snd_soc_read(codec, ES8316_IFACE); + adciface = snd_soc_read(codec, ES8316_ADC_IFACE); + daciface = snd_soc_read(codec, ES8316_DAC_IFACE); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: // MASTER MODE + alsa_dbg("es8316 in master mode"); + iface |= 0x80; + break; + case SND_SOC_DAIFMT_CBS_CFS: // SLAVE MODE + alsa_dbg("es8316 in slave mode"); + iface &= 0x7F; + break; + default: + return -EINVAL; + } + + /* interface format */ + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adciface &= 0xFC; + daciface &= 0xFC; + break; + case SND_SOC_DAIFMT_RIGHT_J: + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + adciface &= 0xFC; + daciface &= 0xFC; + adciface |= 0x01; + daciface |= 0x01; + break; + case SND_SOC_DAIFMT_DSP_A: + adciface &= 0xDC; + daciface &= 0xDC; + adciface |= 0x03; + daciface |= 0x03; + break; + case SND_SOC_DAIFMT_DSP_B: + adciface &= 0xDC; + daciface &= 0xDC; + adciface |= 0x23; + daciface |= 0x23; + break; + default: + return -EINVAL; + } + + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface &= 0xDF; + adciface &= 0xDF; + daciface &= 0xDF; + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x20; + adciface |= 0x20; + daciface |= 0x20; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x20; + adciface &= 0xDF; + daciface &= 0xDF; + break; + case SND_SOC_DAIFMT_NB_IF: + iface &= 0xDF; + adciface |= 0x20; + daciface |= 0x20; + break; + default: + return -EINVAL; + } + snd_soc_write(codec, ES8316_IFACE, iface); + snd_soc_write(codec, ES8316_ADC_IFACE, adciface); + snd_soc_write(codec, ES8316_DAC_IFACE, daciface); + return 0; +} + +static void pcm_pop_work_events(struct work_struct *work) +{ + int ret; + //printk("es8316--------pcm_pop_work_events\n"); + snd_soc_write(tron_codec,ES8316_CPHP_OUTEN_REG17,0x66); + ret=snd_soc_read(tron_codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) { + gpio_set_value(368, 1); + //printk("2222 gpio_set_value 1\n"); + } + es8316_init_reg=1; +} + +static int es8316_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int ret; + //printk("es8316_pcm_startup Enter::%s----%d es8316->sysclk=%d\n",__FUNCTION__,__LINE__,es8316->sysclk); + if(playback) + { + snd_soc_write(codec, 0x50, 0xAA); + if(es8316_init_reg==0){ + //printk("==================>>>>>>>>es8316_pcm_startup es8316_init_reg=0\n"); + schedule_delayed_work(&es8316->pcm_pop_work,msecs_to_jiffies(100)); + } + else{ + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) + { + gpio_set_value(368, 1); + // printk("es8316_pcm_startup gpio_set_value 1\n"); + //msleep(250); + } + } + } + + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!es8316->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + //return -EINVAL; + } else { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, es8316->sysclk_constraints); +} + + return 0; +} + +static void es8316_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + struct snd_soc_codec *codec = dai->codec; + if(playback) + { + snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x30);//close �� + gpio_set_value(368, 0); + snd_soc_write(codec, 0x50, 0xA0); + //printk("es8316_pcm_shutdown----gpio_set_value 0\n"); + //printk("%s playback\n",__FUNCTION__); + } +} + + +static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + int retv; + int coeff; + u16 osrate = snd_soc_read(codec, ES8316_CLKMGR_ADCOSR_REG03) & 0xc0; + u16 mclkdiv = snd_soc_read(codec, ES8316_CLKMGR_CLKSW_REG01) & 0x7f; + u16 srate = snd_soc_read(codec, ES8316_SDP_MS_BCKDIV_REG09) & 0xE0; + u16 adciface = snd_soc_read(codec, ES8316_SDP_ADCFMT_REG0A) & 0xE3; + u16 daciface = snd_soc_read(codec, ES8316_SDP_DACFMT_REG0B) & 0xE3; + u16 adcdiv = snd_soc_read(codec, ES8316_CLKMGR_ADCDIV1_REG04); + u16 adclrckdiv_l = snd_soc_read(codec, ES8316_CLKMGR_ADCDIV2_REG05) & 0x00; + u16 dacdiv = snd_soc_read(codec, ES8316_CLKMGR_DACDIV1_REG06); + u16 daclrckdiv_l = snd_soc_read(codec, ES8316_CLKMGR_DACDIV2_REG07) & 0x00; + u16 adclrckdiv_h = adcdiv & 0xf0; + u16 daclrckdiv_h = dacdiv & 0xf0; + int ret; + adcdiv &= 0x0f; + dacdiv &= 0x0f; + + u16 tmp = snd_soc_read(codec, ES8316_CLKMGR_DACDIV2_REG07); + + coeff = get_coeff(es8316->sysclk, params_rate(params)); + if (coeff < 0) { + coeff = get_coeff(es8316->sysclk / 2, params_rate(params)); + mclkdiv |= 0x80; + } + if (coeff < 0) { + dev_err(codec->dev, + "Unable to configure sample rate %dHz with %dHz MCLK\n", + params_rate(params), es8316->sysclk); + return coeff; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + adciface |= 0x000C; + daciface |= 0x000C; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + adciface |= 0x0004; + daciface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + break; + case SNDRV_PCM_FORMAT_S32_LE: + adciface |= 0x0010; + daciface |= 0x0010; + break; + } + + /* set iface & srate*/ + snd_soc_update_bits(codec,ES8316_SDP_DACFMT_REG0B, 0xe3, daciface); + snd_soc_update_bits(codec, ES8316_SDP_ADCFMT_REG0A, 0xe3, adciface); + if (coeff >= 0) { + osrate = coeff_div[coeff].osr; + osrate &= 0x3f; + + srate |= coeff_div[coeff].sr; + srate &= 0x1f; + + adcdiv |= (coeff_div[coeff].div << 4); + adclrckdiv_h |= coeff_div[coeff].lrck_h; + adcdiv &= 0xf0; + adclrckdiv_h &= 0x0f; + adcdiv |= adclrckdiv_h; + adclrckdiv_l = coeff_div[coeff].lrck_l; + + dacdiv |= (coeff_div[coeff].div << 4); + daclrckdiv_h |= coeff_div[coeff].lrck_h; + dacdiv &= 0xf0; + daclrckdiv_h &= 0x0f; + dacdiv |= daclrckdiv_h; + daclrckdiv_l = coeff_div[coeff].lrck_l; + /* + * must do codec power on initialization at here becauses MCLK and I2S CLK always startup at here + */ + retv = snd_soc_read(codec, 0x14) ; + //pr_err("%s:%d, retv=%x \n", __func__, __LINE__, retv); + if(es8316_init_reg>0){ + //printk("===%s es8316_init_reg>0\n", __func__); + snd_soc_write(codec, 0X17, 0x66); + snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x10); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) { + if(es8316_init_reg>0){ + gpio_set_value(368, 1); + // printk("gpio_set_value 1\n"); + //msleep(250); + } + } + + }else{ + //printk("===%s es8316_init_reg=0\n", __func__); + snd_soc_write(codec, 0x01, 0x7F); + snd_soc_write(codec, 0x2F, 0x00); + snd_soc_write(codec, 0X18, 0x00); + snd_soc_write(codec, 0X1b, 0x30); + snd_soc_write(codec, 0X1a, 0x10); + snd_soc_write(codec, 0X19, 0x03); + //msleep(10); + snd_soc_write(codec, 0X14, 0x88); + snd_soc_write(codec, 0X15, 0x00); + snd_soc_write(codec, 0X16, 0xbb); + snd_soc_write(codec, 0x00, 0xC0); //CHARGE PUMP DIVIDER + msleep(50); + snd_soc_write(codec, 0X17, 0x66); + snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x00); + snd_soc_write(codec, ES8316_GPIO_SEL_REG4D, 0x00); //set gpio2 to GM SHORT + snd_soc_write(codec, ES8316_GPIO_DEBUNCE_INT_REG4E, 0xf3); //maximum debance time, enable interrupt, low active + es8316_set_bias_level(codec, SND_SOC_BIAS_ON); + //es8316_init_reg = 1; + } + //pr_err("%s:%d, retv=%x \n", __func__, __LINE__, retv); + } + + //snd_soc_write(codec,ES8316_DAC_SET2_REG31, 0x20); + retv = snd_soc_read(codec, ES8316_GPIO_FLAG); + //pr_err("%s:%d, retv=%x \n", __func__, __LINE__, retv); + return 0; +} + +static int es8316_mute(struct snd_soc_dai *dai, int mute) +{ + //printk("===enter %s\n", __func__); + #if 0 + struct snd_soc_codec *codec = dai->codec; + int ret; + dev_dbg(codec->dev, "%s %d\n", __func__, mute); + if (mute){ + //snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x20);//close �� + gpio_set_value(368, 0); + }else{ + //snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x00); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) { //remove + gpio_set_value(368, 1); + } + } + #endif + return 0; +} + +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dev_dbg(codec->dev, "%s on\n", __func__); + break; + case SND_SOC_BIAS_PREPARE: + dev_dbg(codec->dev, "%s prepare\n", __func__); + if(es8316_init_reg>0) + { + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x7F); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x88); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x00); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0xBB); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x10); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x03); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x00); + snd_soc_write(codec, ES8316_RESET_REG00, 0xC0); + //snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x66); + } + break; + case SND_SOC_BIAS_STANDBY: + dev_dbg(codec->dev, "%s standby\n", __func__); + if(es8316_init_reg>0) // if codec intialized, + { + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x7F); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x88); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x00); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0xBB); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x10); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x03); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x00); + snd_soc_write(codec, ES8316_RESET_REG00, 0xC0); + //snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x66); + } + break; + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "%s off\n", __func__); + /*if(es8316_init_reg>0){ + // snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x33); + snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x07); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_write(codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x11); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x3F); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + snd_soc_write(codec, ES8316_RESET_REG00, 0x7F); + }*/ + break; + } + + return 0; +} + +#define es8316_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops es8316_ops = { + .startup = es8316_pcm_startup, + .hw_params = es8316_pcm_hw_params, + .set_fmt = es8316_set_dai_fmt, + .set_sysclk = es8316_set_dai_sysclk, + .digital_mute = es8316_mute, + .shutdown = es8316_pcm_shutdown, +}; + +static struct snd_soc_dai_driver es8316_dai = { + .name = "ES8316 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8316_RATES, + .formats = es8316_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8316_RATES, + .formats = es8316_FORMATS, + }, + .ops = &es8316_ops, + .symmetric_rates = 1, +}; + + +static int es8316_init_regs(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s\n", __func__); + snd_soc_write(codec,0x01, 0x7f); //setup adc volume + snd_soc_write(codec,0x02, 0x00); //disable DAC ds and notch mode + snd_soc_write(codec, 0x03,0x19); //CHARGE PUMP DIVIDER + snd_soc_write(codec, 0x04,0x21); + snd_soc_write(codec, 0x05,0x90); + snd_soc_write(codec, 0x06,0x11); + snd_soc_write(codec, 0x07,0x90); + snd_soc_write(codec, 0x08,0x00); + snd_soc_write(codec, 0x09,0x01); + + snd_soc_write(codec, 0x0A, 0x0C);//ADC FORMAT , i2s-16bit, + snd_soc_write(codec, 0x0B, 0x0C);//DAC FORMAT, i2s-16bit + snd_soc_write(codec, 0x0D,0x00); //CLK CASE + snd_soc_write(codec, 0x22,0x20); //CHARGE PUMP DIVIDER + snd_soc_write(codec, 0x2F,0x00); + snd_soc_write(codec, 0x10,0x11); + snd_soc_write(codec, 0x12,0x28); + snd_soc_write(codec, 0x1C,0x0F); + snd_soc_write(codec, 0x1D,0x0C); + snd_soc_write(codec, 0x00,0xC0); //CHARGE PUMP DIVIDER + msleep(50); + + snd_soc_write(codec, 0x23,0x10); //CLK ON + snd_soc_write(codec, 0x24,0x01); //VMID SEQUENCE + snd_soc_write(codec, 0x26,0x00); //ADC,DAC VREF SET, micbias pup + snd_soc_write(codec, 0x27,0x00); //CSM POWER UP + snd_soc_write(codec, 0x30, 0x10); //INPUT SELECT INPUT1:0X20,INPUT2:0X30, ain and adc power down + snd_soc_write(codec, 0x33, 0x00); //POWER UP DACPOWER + snd_soc_write(codec, 0x34, 0x00); //L,R MIXER SET + snd_soc_write(codec, 0x14,0x88); //MIXER + snd_soc_write(codec, 0x15,0x00); //MIXER + snd_soc_write(codec, 0x16,0xAA); //MIXER + snd_soc_write(codec, 0X18,0x00); //CHARGE PUMP + snd_soc_write(codec, 0X31, 0x00); //CHARGE PUMP + snd_soc_write(codec, 0X29, 0xd2); //CHARGE PUMP + snd_soc_write(codec, 0X2A, 0x08); //vmidlow = 10'b + snd_soc_write(codec, 0X2B,0xA0); //HPOUT SET + + snd_soc_write(codec, 0X2C, 0x05); + snd_soc_write(codec, 0X2D, 0x06); + snd_soc_write(codec, 0X2E, 0x61); + /*alc set*/ + snd_soc_write(codec, 0X4D, 0x00); + snd_soc_write(codec, 0X4E, 0xF3); + snd_soc_write(codec, 0X50, 0xA0); //adc ds mode, HPF enable + snd_soc_write(codec, 0X51, 0x00); //ALC ON, + snd_soc_write(codec, 0X52, 0x00); + //msleep(50); + + snd_soc_write(codec, 0X18, 0x00); + snd_soc_write(codec, 0X1b, 0x30); + snd_soc_write(codec, 0X1a, 0x10); + snd_soc_write(codec, 0X19, 0x03); + snd_soc_write(codec, 0X14, 0x88); + snd_soc_write(codec, 0X15, 0x00); + snd_soc_write(codec, 0X16, 0xbb); + snd_soc_write(codec, 0X17, 0x66); + return 0; +} + +//static int es8316_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int es8316_suspend(struct device *dev) +{ + //printk("%s suspend\n", __func__); + gpio_set_value(368, 0); + // snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x33); + if(es8316_init_reg>0){ + /*if Codec has been initialized before, then enter into suspend directly */ + //printk("===%s es8316_init_reg > 0\n", __func__); + snd_soc_write(tron_codec, ES8316_CPHP_ICAL_VOL_REG18, 0x11); + mdelay(5); + snd_soc_write(tron_codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(tron_codec, 0x2f, 0x11); + snd_soc_write(tron_codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(tron_codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(tron_codec, ES8316_CPHP_PDN1_REG19, 0x06); + snd_soc_write(tron_codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(tron_codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(tron_codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_update_bits(tron_codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0, 0xc0); + snd_soc_write(tron_codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + //snd_soc_write(tron_codec, 0x00,0x7f); //CHARGE PUMP DIVIDER + } else { + #if 0 + /*if Codec has not been initialized before, enter into normal mode firstly, then enter into suspend at last */ + //snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0, 0x00); + //printk("===%s es8316_init_reg = 0\n", __func__); + snd_soc_write(tron_codec, 0x2F, 0x00); + snd_soc_write(tron_codec, 0X18, 0x11); + snd_soc_write(tron_codec, 0X1b, 0x30); + snd_soc_write(tron_codec, 0X1a, 0x10); + snd_soc_write(tron_codec, 0X19, 0x03); + snd_soc_write(tron_codec, 0X14, 0x88); + snd_soc_write(tron_codec, 0X15, 0x00); + snd_soc_write(tron_codec, 0X16, 0xbb); + snd_soc_write(tron_codec, 0x00,0xC0); //CHARGE PUMP DIVIDER + //msleep(50); + //snd_soc_write(tron_codec, 0X17, 0x66); + msleep(200); + snd_soc_write(tron_codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(tron_codec, 0x2f, 0x11); + snd_soc_write(tron_codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(tron_codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(tron_codec, ES8316_CPHP_PDN1_REG19, 0x06); + snd_soc_write(tron_codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(tron_codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(tron_codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_update_bits(tron_codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0, 0xc0); + snd_soc_write(tron_codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + //es8316_init_reg = 1; + #endif + } + //snd_soc_write(tron_codec, 0X4E, 0xF0); + return 0; +} + +//static int es8316_resume(struct snd_soc_codec *codec) +static int es8316_resume(struct device *dev) +{ + //printk("===%s resume\n", __func__); + if(es8316_init_reg>0){ + //printk("es8316 resume\n"); + snd_soc_write(tron_codec, ES8316_CLKMGR_CLKSW_REG01, 0x7f); + snd_soc_write(tron_codec, ES8316_CPHP_ICAL_VOL_REG18, 0x11); + snd_soc_write(tron_codec, ES8316_CPHP_LDOCTL_REG1B, 0x30); + snd_soc_write(tron_codec, ES8316_CPHP_PDN2_REG1A, 0x10); + snd_soc_write(tron_codec, ES8316_CPHP_PDN1_REG19, 0x02); + snd_soc_write(tron_codec, ES8316_HPMIX_SWITCH_REG14, 0x88); + snd_soc_write(tron_codec, ES8316_HPMIX_PDN_REG15, 0x00); + snd_soc_write(tron_codec, ES8316_HPMIX_VOL_REG16, 0xbb); + snd_soc_update_bits(tron_codec, ES8316_ADC_PDN_LINSEL_REG22, 0xc0, 0x00); + snd_soc_write(tron_codec, 0x2f, 0x00); + snd_soc_write(tron_codec, ES8316_CPHP_OUTEN_REG17, 0x66); + } + //snd_soc_write(tron_codec, 0X4E, 0xF3); + return 0; +} + +static int es8316_probe(struct snd_soc_codec *codec) +{ + int ret = 0; +// int ReadBuffer,ReadBuffer1,ReadBuffer2; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + pr_debug("---%s--start--\n",__FUNCTION__); + tron_codec = codec; + es8316_init_reg = 0; + //gpio_set_value(139, 0); + snd_soc_write(codec, 0x00, 0xc2); + //msleep(100); + //retv = snd_soc_read(codec, ES8316_CLKMGR_ADCDIV2_REG05) ; + snd_soc_write(codec, 0x01, 0x7f); //setup adc volume + snd_soc_write(codec, 0x02, 0x00); //disable DAC ds and notch mode + snd_soc_write(codec, 0x03,0x19); //CHARGE PUMP DIVIDER + snd_soc_write(codec, 0x04,0x21); + snd_soc_write(codec, 0x05,0x90); + snd_soc_write(codec, 0x06,0x11); + snd_soc_write(codec, 0x07,0x90); + snd_soc_write(codec, 0x08,0x00); + snd_soc_write(codec, 0x09,0x01); + + snd_soc_write(codec, 0x0A, 0x0C);//ADC FORMAT , i2s-16bit, + snd_soc_write(codec, 0x0B, 0x0C);//DAC FORMAT, i2s-16bit + snd_soc_write(codec, 0x0D,0x00); //CLK CASE + + snd_soc_write(codec, 0x10, 0x11); + snd_soc_write(codec, 0x12, 0x28); + snd_soc_write(codec, 0x1C, 0x0F); + snd_soc_write(codec, 0x1D, 0x0C); + msleep(50); + + snd_soc_write(codec, 0x23, 0x50); //CLK ON + snd_soc_write(codec, 0x24, 0x00); //VMID SEQUENCE + if(es8316->dmic_amic == dmic_used){ + snd_soc_write(codec, 0x25, 0x0A); //dmic channel + }else { + snd_soc_write(codec, 0x25, 0x08); //amic channel + } + snd_soc_write(codec, 0x26, 0x00); //ADC,DAC VREF SET, micbias pup + snd_soc_write(codec, 0x27, 0x00); //CSM POWER UP + snd_soc_write(codec, 0x30, 0x10); //INPUT SELECT INPUT1:0X20,INPUT2:0X30, ain and adc power down + snd_soc_write(codec, 0x33, 0x00); //POWER UP DACPOWER + snd_soc_write(codec, 0x34, 0x00); //L,R MIXER SET + snd_soc_write(codec, 0X31, 0x00); //CHARGE PUMP + if(es8316->dmic_amic == dmic_used){ + snd_soc_write(codec, 0X29, 0xDC); //CHARGE PUMP + snd_soc_write(codec, 0X2A, 0x1C); //vmidlow = 10'b + }else { + snd_soc_write(codec, 0X29, 0xD0); //CHARGE PUMP + snd_soc_write(codec, 0X2A, 0x08); //vmidlow = 10'b + } + snd_soc_write(codec, 0X2B, 0xA0); //HPOUT SET + + snd_soc_write(codec, 0X2C, 0x05); + snd_soc_write(codec, 0X2D, 0x06); + snd_soc_write(codec, 0X2E, 0x21); + /*alc set*/ + snd_soc_write(codec, 0X4D, 0x00); + snd_soc_write(codec, 0X4E, 0xF0); + snd_soc_write(codec, 0X50, 0xA0); //adc ds mode, HPF enable + snd_soc_write(codec, 0X51, 0x00); //ALC ON, + snd_soc_write(codec, 0X52, 0x00); +#if defined(HS_IRQ) + det_initalize(); +#elif defined(HS_TIMER) + hsdet_init(); +#endif + INIT_DELAYED_WORK(&es8316->pcm_pop_work, pcm_pop_work_events); + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + //.suspend = es8316_suspend, + //.resume = es8316_resume, + .set_bias_level = es8316_set_bias_level, + //.idle_bias_off = true, + + .component_driver = { + .controls = es8316_snd_controls, + .num_controls = ARRAY_SIZE(es8316_snd_controls), + .dapm_widgets = es8316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets), + .dapm_routes = es8316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), + }, +}; + +static const struct regmap_config es8316_regmap = { + .reg_bits = 8, + .val_bits = 8, + .volatile_reg = es8316_volatile, + .readable_reg = es8316_readable, + .max_register = ARRAY_SIZE(es8316_reg_defaults) + 1, + .reg_defaults = es8316_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(es8316_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static void es8316_i2c_shutdown(struct i2c_client *i2c) +{ + struct snd_soc_codec *codec; + + if (!tron_codec) + goto err; + codec = tron_codec; + if(es8316_init_reg>0){ + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x00); + snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x06); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_write(codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x11); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x3F); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + snd_soc_write(codec, ES8316_RESET_REG00, 0x7F); + } +err: + return; +} + +static int es8316_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct es8316_priv *es8316; + struct regmap *regmap; + int ret = -1; + + es8316 = kzalloc(sizeof(*es8316), GFP_KERNEL); + if (es8316 == NULL){ + return -ENOMEM; + } + es8316->dmic_amic = dmic_used; //if internal mic is amic + + regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + i2c_set_clientdata(i2c_client, es8316); + ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316, + &es8316_dai, 1); + if (ret < 0) { + kfree(es8316); + return ret; + } + + gpio_request(368,"SPK_OUT_SHUTDOWN"); + gpio_direction_output(368, 0); + return ret; +} + +static int es8316_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const unsigned short normal_i2c[] = {0x11, I2C_CLIENT_END}; +static const struct i2c_device_id es8316_i2c_id[] = { + {"es8316", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8316_acpi_match[] = { + { "ESSX8316", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); +#endif + +static struct dev_pm_ops es8316_pm_ops = { + .suspend = es8316_suspend, + .resume = es8316_resume, +}; + + +static struct i2c_driver es8316_i2c_driver = { + .driver = { + .name = "es8316", + .pm = &es8316_pm_ops, + .acpi_match_table = ACPI_PTR(es8316_acpi_match), + }, + .shutdown = es8316_i2c_shutdown, + .probe = es8316_i2c_probe, + .remove = es8316_i2c_remove, + .id_table = es8316_i2c_id, +}; +module_i2c_driver(es8316_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8316 driver"); +MODULE_AUTHOR("Will <will(a)everset-semi.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 0000000..cb5d035 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,124 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.com> + * + * Based on ES8323.h + * + * 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 _ES8316_H +#define _ES8316_H + +/* ES8316 register space */ +/* +* RESET Control +*/ +#define ES8316_RESET_REG00 0x00 +/* +* Clock Managerment +*/ +#define ES8316_CLKMGR_CLKSW_REG01 0x01 +#define ES8316_CLKMGR_CLKSEL_REG02 0x02 +#define ES8316_CLKMGR_ADCOSR_REG03 0x03 +#define ES8316_CLKMGR_ADCDIV1_REG04 0x04 +#define ES8316_CLKMGR_ADCDIV2_REG05 0x05 +#define ES8316_CLKMGR_DACDIV1_REG06 0x06 +#define ES8316_CLKMGR_DACDIV2_REG07 0x07 +#define ES8316_CLKMGR_CPDIV_REG08 0x08 +/* +* SDP Control +*/ +#define ES8316_SDP_MS_BCKDIV_REG09 0x09 +#define ES8316_SDP_ADCFMT_REG0A 0x0a +#define ES8316_SDP_DACFMT_REG0B 0x0b +/* +* System Control +*/ +#define ES8316_SYS_VMIDSEL_REG0C 0x0c +#define ES8316_SYS_PDN_REG0D 0x0d +#define ES8316_SYS_LP1_REG0E 0x0e +#define ES8316_SYS_LP2_REG0F 0x0f +#define ES8316_SYS_VMIDLOW_REG10 0x10 +#define ES8316_SYS_VSEL_REG11 0x11 +#define ES8316_SYS_REF_REG12 0x12 +/* +* HP Mixer +*/ +#define ES8316_HPMIX_SEL_REG13 0x13 +#define ES8316_HPMIX_SWITCH_REG14 0x14 +#define ES8316_HPMIX_PDN_REG15 0x15 +#define ES8316_HPMIX_VOL_REG16 0x16 +/* +* Charge Pump Headphone driver +*/ +#define ES8316_CPHP_OUTEN_REG17 0x17 +#define ES8316_CPHP_ICAL_VOL_REG18 0x18 +#define ES8316_CPHP_PDN1_REG19 0x19 +#define ES8316_CPHP_PDN2_REG1A 0x1a +#define ES8316_CPHP_LDOCTL_REG1B 0x1b +/* +* Calibration +*/ +#define ES8316_CAL_TYPE_REG1C 0x1c +#define ES8316_CAL_SET_REG1D 0x1d +#define ES8316_CAL_HPLIV_REG1E 0x1e +#define ES8316_CAL_HPRIV_REG1F 0x1f +#define ES8316_CAL_HPLMV_REG20 0x20 +#define ES8316_CAL_HPRMV_REG21 0x21 +/* +* ADC Control +*/ +#define ES8316_ADC_PDN_LINSEL_REG22 0x22 +#define ES8316_ADC_PGAGAIN_REG23 0x23 +#define ES8316_ADC_D2SEPGA_REG24 0x24 +#define ES8316_ADC_DMIC_REG25 0x25 +#define ES8316_ADC_MUTE_REG26 0x26 +#define ES8316_ADC_VOLUME_REG27 0x27 +#define ES8316_ADC_ALC1_REG29 0x29 +#define ES8316_ADC_ALC2_REG2A 0x2a +#define ES8316_ADC_ALC3_REG2B 0x2b +#define ES8316_ADC_ALC4_REG2C 0x2c +#define ES8316_ADC_ALC5_REG2D 0x2d +#define ES8316_ADC_ALC6_REG2E 0x2e +/* +* DAC Control +*/ +#define ES8316_DAC_PDN_REG2F 0x2f +#define ES8316_DAC_SET1_REG30 0x30 +#define ES8316_DAC_SET2_REG31 0x31 +#define ES8316_DAC_SET3_REG32 0x32 +#define ES8316_DAC_VOLL_REG33 0x33 +#define ES8316_DAC_VOLR_REG34 0x34 +/* +* GPIO +*/ +#define ES8316_GPIO_SEL_REG4D 0x4D +#define ES8316_GPIO_DEBUNCE_INT_REG4E 0x4E +#define ES8316_GPIO_FLAG 0x4F +/* +* TEST MODE +*/ +#define ES8316_TESTMODE_REG50 0x50 +#define ES8316_TEST1_REG51 0x51 +#define ES8316_TEST2_REG52 0x52 +#define ES8316_TEST3_REG53 0x53 +/* +* es8396 System clock derived from MCLK or BCLK +*/ +#define ES8316_CLKID_MCLK 0 +#define ES8316_CLKID_BCLK 1 +#define ES8316_CLKID_PLLO 2 + + +#define ES8316_IFACE ES8316_SDP_MS_BCKDIV_REG09 +#define ES8316_ADC_IFACE ES8316_SDP_ADCFMT_REG0A +#define ES8316_DAC_IFACE ES8316_SDP_DACFMT_REG0B + +#define ES8316_REGNUM 84 + +#endif diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index fd5d1e0..b3023db 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -210,6 +210,18 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH platforms with MAX98090 audio codec it also can support TI jack chip as aux device. If unsure select "N". +config SND_SOC_INTEL_CHT_ES8316_MACH + tristate "ASoC Audio driver for Intel Cherrytrail with ES8316 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ES8316 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail + platforms with ES8316 audio codec. + If unsure select "N". + config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0838478..25ea25c 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -21,6 +21,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt +#define DEBUG #include <linux/slab.h> #include <sound/soc.h> @@ -894,7 +895,7 @@ int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt) * this can be overwritten by set_dai_xxx APIs */ static const struct sst_ssp_config sst_ssp_configs = { - .ssp_id = SSP_CODEC, + .ssp_id = SSP_BT, .bits_per_slot = 24, .slots = 4, .ssp_mode = SSP_MODE_MASTER, @@ -943,6 +944,8 @@ int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) ssp_id = SSP_MODEM; else if (strcmp(id, "ssp2-port") == 0) ssp_id = SSP_CODEC; + else if (strcmp(id, "ssp1-port") == 0) + ssp_id = SSP_BT; // FIXME what is the right ID? else { dev_dbg(dai->dev, "port %s is not supported\n", id); return -1; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 0a88537..75997e2 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -458,6 +458,8 @@ static struct sst_acpi_mach sst_acpi_chv[] = { &chv_platform_data }, {"193C9890", "cht-bsw-max98090", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, &chv_platform_data }, + {"ESSX8316", "cht-es8316", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, + &chv_platform_data }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", cht_quirk, &chv_platform_data }, diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 5639f10..2124aaa 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -10,6 +10,7 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-sst-cht-es8316-objs := cht_es8316.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_ES8316_MACH) += snd-soc-sst-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o diff --git a/sound/soc/intel/boards/cht_es8316.c b/sound/soc/intel/boards/cht_es8316.c new file mode 100644 index 0000000..a6e105d --- /dev/null +++ b/sound/soc/intel/boards/cht_es8316.c @@ -0,0 +1,1462 @@ +/* + * cht_cr_es8316.c - ASoc DPCM Machine driver + * for Intel CherryTrail MID platform + * + * Copyright (C) 2014 Intel Corp + * Author: Praveen Diwakar <praveen.diwakar(a)intel.com> + * Bhakte, GurudattaX <gurudattax.bhakte(a)intel.com> + * This file is modified from cht_bl_dpcm_rt5672.c for cherrytrail CR + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define DEBUG +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/input.h> +#include <linux/gpio.h> +#include <linux/acpi.h> +#include <linux/slab.h> +#include <linux/input.h> +#include <linux/mutex.h> +#include <linux/clk.h> +#include <asm/intel-mid.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/es8316.h" +#include "../atom/sst-atom-controls.h" + +struct clk *mclk; + +#define CHT_PLAT_CLK_3_HZ 19200000 + +#define CHT_INTR_DEBOUNCE 0 //0 /* updated by everest-david 15-3-4*/ +#define CHT_HS_DEBOUNCE_DELAY 500//500 /* added by everest-david */ +#define CHT_HS_INSERT_DET_DELAY 400//600 +#define CHT_HS_REMOVE_DET_DELAY 40 //100 /* updated by everest-david 15-3-4*/ +#define CHT_BUTTON_PRESS_DELAY 50 +#define CHT_BUTTON_RELEASE_DELAY 50 +#define CHT_HS_DET_POLL_INTRVL 50 +#define CHT_BUTTON_EN_DELAY 0 + +#define CHT_HS_DET_RETRY_COUNT 8 +#define CHT_HS_DEBOUNCE_RETRY_COUNT 3 /* added by everest-david */ +#define CHT_HS_BUTTON_PRESS_CHK_COUNT 3 /* added by everest-david */ +#define CHT_HS_BUTTON_RELEASE_CHK_COUNT 3 /* added by everest-david */ +#define CHT_HS_REMOVE_RETRY_COUNT 1 /* added by everest-david 15-3-4*/ + +#define VLV2_PLAT_CLK_AUDIO 3 +#define PLAT_CLK_FORCE_ON 1 +#define PLAT_CLK_FORCE_OFF 2 + +#define MAINMIC_DMIC_USED 1 +//#define MAINMIC_AMIC_USED 1 + + +struct cht_mc_private { + struct snd_soc_codec *codec; + struct snd_soc_jack jack; + struct delayed_work hs_insert_work; + struct delayed_work hs_remove_work; + struct delayed_work hs_button_press_work; + struct delayed_work hs_button_release_work; + struct delayed_work hs_debounce_work; /* added by everest-david */ + struct delayed_work hs_poll_work; + struct mutex jack_mlock; + struct mutex poll_mlock; + /* To enable button press interrupts after a delay after HS detection. + * This is to avoid spurious button press events during slow + * HS insertion + */ + struct delayed_work hs_button_en_work; + int intr_debounce; + int hs_debounce_delay; /* added by everest-david */ + int hs_debounce_retry; /* added by everest-david */ + int hs_insert_det_delay; + int hs_remove_det_delay; + int hs_remove_retry; /* added by everest-david 15-3-4*/ + int button_press_delay; + int button_press_chk_count; /* added by everest-david */ + int button_release_delay; + int button_release_chk_count; /* added by everest-david */ + int button_en_delay; + int hs_det_poll_intrvl; + int hs_det_retry; + + int same_reg_cnt; + int hs_reg_v; + int poll_cnt; + int poll_status; + int poll_same_cnt; + bool process_button_events; +}; + +static int cht_hs_detection(void *); + +static struct snd_soc_jack_gpio hs_gpio = { + .name = "cht-codec-int", + .report = SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0, + .debounce_time = CHT_INTR_DEBOUNCE, + .jack_status_check = cht_hs_detection, +}; +static inline void cht_force_enable_pin(struct snd_soc_codec *codec, + const char *bias_widget, bool enable) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + pr_debug("%s %s\n", enable ? "enable" : "disable", bias_widget); + if (enable) + snd_soc_dapm_force_enable_pin(dapm, bias_widget); + else + snd_soc_dapm_disable_pin(dapm, bias_widget); +} + +static inline void cht_set_mic_bias(struct snd_soc_codec *codec, bool enable) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + if (enable) + cht_force_enable_pin(codec, "micbias", true); + else + cht_force_enable_pin(codec, "micbias", false); + snd_soc_dapm_sync(dapm); +} + +static inline void cht_set_codec_power(struct snd_soc_codec *codec, + int jack_type) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + switch (jack_type) { + case SND_JACK_HEADSET: + cht_force_enable_pin(codec, "micbias2", true); + cht_force_enable_pin(codec, "JD Power", true); + cht_force_enable_pin(codec, "Mic Det Power", true); + break; + case SND_JACK_HEADPHONE: + cht_force_enable_pin(codec, "JD Power", true); + cht_force_enable_pin(codec, "Mic Det Power", false); + cht_force_enable_pin(codec, "micbias2", false); + break; + case 0: + cht_force_enable_pin(codec, "JD Power", false); + cht_force_enable_pin(codec, "Mic Det Power", false); + cht_force_enable_pin(codec, "micbias2", false); + break; + default: + return; + } + snd_soc_dapm_sync(dapm); +} +static int es8316_headset_detect(struct snd_soc_codec *codec) +{ + + int value; + int status; + + value = snd_soc_read(codec, ES8316_GPIO_FLAG); + //printk("=== es8316_headset_detect: 0x%x\n",value); + switch (value) { + //case 0x24: + case 0x20: + status = SND_JACK_HEADPHONE; + break; + case 0x22: + status = SND_JACK_HEADSET; + break; + default: + status = 0; + break; + } + + return status; +} + +/* Identify the jack type as Headset/Headphone/None */ +static int cht_check_jack_type(struct snd_soc_jack *jack, + struct snd_soc_codec *codec) +{ + int status, jack_type = 0; + struct cht_mc_private *ctx = container_of(jack, + struct cht_mc_private, jack); + + status = es8316_headset_detect(codec); + /* jd status high indicates some accessory has been connected */ + if (status) { + pr_debug("Jack insert intr"); + snd_soc_write(codec,0x30,0x30); + gpio_direction_output(368,0); + + pr_debug("Jack insert intr"); + /* Do not process button events until accessory is + detected as headset*/ + ctx->process_button_events = false; + cht_set_mic_bias(codec, true); + jack_type = es8316_headset_detect(codec); //read flag register to check headphone / headset type / or none + pr_info("jack_type:%d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) { + ctx->process_button_events = true; +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin2rin2 + snd_soc_write(codec, 0x25,0x08); //record path switch to headset mic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin1rin1 +#endif +#endif + + /* If headset is detected, enable button + interrupts after a delay */ + schedule_delayed_work(&ctx->hs_button_en_work, + msecs_to_jiffies(ctx->button_en_delay)); + } + if (jack_type != SND_JACK_HEADSET){ + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + + } + } else { + jack_type = 0; + } + pr_info("Jack type detected:%d\n", jack_type); + return jack_type; +} + +/* Work function invoked by the Jack Infrastructure. + * Other delayed works for jack detection/removal/button + * press are scheduled from this function + */ +static int cht_hs_detection(void *data) +{ + int status, jack_type = 0; + int ret,value; + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(jack, + struct cht_mc_private, jack); + struct snd_soc_codec *codec = ctx->codec; + + pr_info("Enter:%s", __func__); + /* Ack interrupt first */ + mutex_lock(&ctx->jack_mlock); + + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + + /* Initialize jack status with previous status. + The delayed work will confirm the event and + send updated status later */ + + jack_type = jack->status; + value = snd_soc_read(codec, ES8316_GPIO_FLAG); + //printk("=== cht_hs_detection gpio flag: 0x%x\n",value); + if (!jack->status) { + //printk("=== jack->status = 0\n"); + snd_soc_write(codec,0x30,0x30); //mute dac when hp inserted + //snd_soc_write(codec,0x22,0x30); + ctx->hs_det_retry = CHT_HS_DET_RETRY_COUNT; + ctx->hs_debounce_retry = CHT_HS_DEBOUNCE_RETRY_COUNT; + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + ctx->hs_remove_retry = CHT_HS_REMOVE_RETRY_COUNT; //UPDATED BY DAVID,15-3-4 + + ret = schedule_delayed_work(&ctx->hs_insert_work, + msecs_to_jiffies(ctx->hs_insert_det_delay)); + if (!ret){ + pr_info("byt_check_hs_insert_status already queued"); + } else { + pr_info("%s:Check hs insertion after %d msec", + __func__, ctx->hs_insert_det_delay); + } + } else { + /* First check for accessory removal; If not removed, + check for button events*/ + status = es8316_headset_detect(codec); + /* jd status low indicates accessory has been disconnected. + However, confirm the removal in the delayed work */ + if (!status) { + printk("=== >status = 0\n"); + /* Do not process button events while we make sure + accessory is disconnected*/ + ctx->process_button_events = false; + snd_soc_write(codec,0x30,0x20); + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + if (!ret) { + pr_info("byt_check_hs_remove_status already queued"); + }else +{ + pr_info("%s:Check hs removal after %d msec", __func__, + ctx->hs_remove_det_delay); + } + } else { /* Must be button event. + * Confirm the event in delayed work */ + printk("=== >status = 1\n"); + if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) && + ctx->process_button_events) { + ret = schedule_delayed_work(&ctx->hs_button_press_work, + msecs_to_jiffies(ctx->button_press_delay)); + if (!ret) + { + pr_info("byt_check_hs_button_press_status already queued"); + } + else + pr_info("%s:check BP/BR after %d msec", + __func__, + ctx->button_press_delay); + } + else + { + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + if (!ret) { + pr_info("byt_check_hs_remove_status already queued"); + }else { + pr_info("%s:Check hs removal after %d msec", __func__, + ctx->hs_remove_det_delay); + } + } + } + } + + pr_info("Exit:%s", __func__); + mutex_unlock(&ctx->jack_mlock); + + return jack_type; +} +/* Checks jack insertion and identifies the jack type. + * Retries the detection if necessary + */ +static void cht_check_hs_insert_status(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, + struct cht_mc_private, hs_insert_work.work); + int jack_type = 0,ret; + struct snd_soc_codec *codec = ctx->codec; + mutex_lock(&ctx->jack_mlock); + pr_debug("Enter:%s", __func__); + + //jack_type = cht_check_jack_type(jack, codec); + + /* Report jack immediately only if jack is headset. + * If headphone or no jack was detected, dont report it + * until the last HS det try. + * This is to avoid reporting any temporary jack removal or + * accessory change (eg, HP to HS) during the detection tries. + * This provides additional debounce that will help in the case + * of slow insertion. + * This also avoids the pause in audio due to accessory change + * from HP to HS + */ + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if(ctx->hs_reg_v == ret) { + ctx->same_reg_cnt++; + } + else { + ctx->same_reg_cnt = 0; + } + ctx->hs_reg_v = ret; + //if (ctx->hs_det_retry <= 0) /* end of retries; report the status */{ + if (ctx->same_reg_cnt == CHT_HS_DET_RETRY_COUNT) /* end of retries; report the status */{ + jack_type = cht_check_jack_type(jack, codec); + pr_info("%d Jack type sent is %d\n", __LINE__, jack_type); + if(jack_type == 0) { /* if no hp inserted */ + cht_set_mic_bias(codec, true); + ctx->process_button_events = false; + snd_soc_jack_report(jack, jack_type, gpio->report); //report Jack status to Frame-work +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + } else { /* if headset or headphone inserted*/ + snd_soc_jack_report(jack, jack_type, gpio->report); //report Jack status to Frame-work + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); + gpio_direction_output(368, 0); + + } + ctx->same_reg_cnt = 0; + pr_info("jack report:%s,%d\n", __func__, __LINE__); + /* now, schedule one debounce work-quene, to cancel the noise at remove or insert */ + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } else { + /* Schedule another detection try if headphone or + * no jack is detected. + * During slow insertion of headset, first a headphone + * may be detected. + * Hence retry until headset is detected + */ + //ctx->hs_det_retry--; + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + schedule_delayed_work(&ctx->hs_insert_work, + msecs_to_jiffies(ctx->hs_det_poll_intrvl)); + pr_info("%s:re-try hs detection after %d msec", + __func__, ctx->hs_det_poll_intrvl); + } + + mutex_unlock(&ctx->jack_mlock); +} +/* Checks jack removal. */ +static void cht_check_hs_remove_status(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, + struct cht_mc_private, hs_remove_work.work); + struct snd_soc_codec *codec = ctx->codec; + int status = 0, jack_type = 0; + + int ret; + /* Cancel any pending insertion detection. There + could be pending insertion detection in the + case of very slow insertion or insertion and + immediate removal.*/ + + cancel_delayed_work_sync(&ctx->hs_insert_work); + mutex_lock(&ctx->jack_mlock); + pr_info("Enter:%s", __func__); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if(ctx->hs_reg_v == ret) { + ctx->same_reg_cnt++; + } + else { + ctx->same_reg_cnt = 0; + } + ctx->hs_reg_v = ret; + //if(ctx->hs_remove_retry <= 0){ //Updated by david, 15-3-4 + if (ctx->same_reg_cnt == CHT_HS_REMOVE_RETRY_COUNT) /* end of retries; report the status */{ + + ctx->same_reg_cnt = 0; + /* Initialize jack_type with previous status. + If the event was an invalid one, we return the preious state*/ + jack_type = jack->status; + ret = snd_soc_read(codec, 0x50); + if(ret == 0xaa) { + gpio_direction_output(368, 1); //1 + } + + msleep(100); + if (jack->status) { + /* jack is in connected state; look for removal event */ + + status = es8316_headset_detect(codec); + if (!status) { + pr_info("Jack remove event"); + snd_soc_write(codec, 0x30,0x30); + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + //cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + jack_type = 0; + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + + } else if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) && + !ctx->process_button_events) { + /* Jack is still connected. We may come here if + there was a spurious jack removal event. + No state change is done until removal is confirmed + by the check_jd_status above.i.e. jack status + remains Headset or headphone. But as soon as + the interrupt thread(byt_hs_detection) detected a jack + removal, button processing gets disabled. + Hence re-enable button processing in the case of + headset */ + pr_info(" spurious Jack remove event for headset \ + re-enable button events"); + ctx->process_button_events = true; +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin2rin2 + snd_soc_write(codec, 0x25,0x08); //record path switch to headset mic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin1rin1 +#endif +#endif + } + } + snd_soc_jack_report(jack, jack_type, gpio->report); + + pr_info("%d Jack type sent is %d\n", __LINE__, jack_type); + if(jack_type == 0) + { + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + //cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + jack_type = 0; + snd_soc_jack_report(jack, jack_type, gpio->report); + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + snd_soc_write(codec,0x30,0x30); + /*Now, to schedule a debounce work-quenue to cancel the noise at remove moment*/ + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } else + { + snd_soc_jack_report(jack, jack_type, gpio->report); + } + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); + } + else + { + //ctx->hs_remove_retry--; + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + } + pr_info("Exit:%s", __func__); + mutex_unlock(&ctx->jack_mlock); +} +static int es8316_query_btn_press(struct snd_soc_codec *codec) +{ + + int btnstate = -1; + u32 value; + //bool status; + value = snd_soc_read(codec, ES8316_GPIO_FLAG); + + switch (value & 0x6) { + case 0x0: + btnstate = SND_JACK_BTN_0; + break; + case 0x2: + btnstate = 0; + break; + default: + btnstate = 0; + break; + } + return btnstate; +} +/* Check for button press/release */ +static void cht_check_hs_button_press_status(struct work_struct *work) +{ + + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = + container_of(work, struct cht_mc_private, + hs_button_press_work.work); + struct snd_soc_codec *codec = ctx->codec; + int status = 0, jack_type = 0; + int ret; + mutex_lock(&ctx->jack_mlock); + pr_info("Enter:%s\n", __func__); + jack_type = jack->status; + + status = es8316_headset_detect(codec); + if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) + && ctx->process_button_events) { + + status = es8316_headset_detect(codec); + if (status) { /* confirm jack is connected */ + status = es8316_query_btn_press(codec); + if (status & SND_JACK_BTN_0) { + jack_type = SND_JACK_HEADSET | SND_JACK_BTN_0; + pr_info("%d Jack type sent is %d\n", + __LINE__, jack_type); + snd_soc_jack_report(jack, jack_type, gpio->report); + /* Since there is not button_relese interrupt + schedule delayed work to poll for button + release status + */ + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + ret = schedule_delayed_work(&ctx->hs_button_release_work, + msecs_to_jiffies(ctx->button_release_delay)); + } else { + if(ctx->button_press_chk_count >0){ + ctx->button_press_chk_count--; + ret = schedule_delayed_work(&ctx->hs_button_press_work, + msecs_to_jiffies(ctx->button_press_delay)); + } else { + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + } + } + } + else + { + status = es8316_headset_detect(codec); + if(!status){ + ctx->process_button_events = false; + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + + } else { + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + ret = schedule_delayed_work(&ctx->hs_button_press_work, + msecs_to_jiffies(ctx->button_press_delay)); + } + } + } + if(!jack_type){ + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + jack_type = 0; + cht_set_mic_bias(codec, true); + #ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic + #else + #ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 + #endif + #endif + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } + pr_info("Exit:%s\n", __func__); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + mutex_unlock(&ctx->jack_mlock); +} + + +/* Check for button release */ +static void cht_check_hs_button_release_status(struct work_struct *work) +{ + + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_release_work.work); + struct snd_soc_codec *codec = ctx->codec; + int status = 0, jack_type = 0; + int ret; + mutex_lock(&ctx->jack_mlock); + pr_info("Enter:%s\n", __func__); + jack_type = jack->status; + + if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) + && ctx->process_button_events) { + + status = es8316_headset_detect(codec); + if (status) { /* confirm jack is connected */ + + status = es8316_query_btn_press(codec); + if (!(status & SND_JACK_BTN_0)) { + jack_type = SND_JACK_HEADSET; + pr_info("%d Jack type sent is %d\n", + __LINE__, jack_type); + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + snd_soc_jack_report(jack, jack_type, gpio->report); + } else { + /* Schedule again */ + if(ctx->button_release_chk_count > 0){ + ctx->button_release_chk_count--; + ret = schedule_delayed_work(&ctx->hs_button_release_work, + msecs_to_jiffies(ctx->button_release_delay)); + } else { + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + } + } + + }else { + status = es8316_headset_detect(codec); + if(!status){ + ctx->process_button_events = false; + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + } else { + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + ret = schedule_delayed_work(&ctx->hs_button_release_work, + msecs_to_jiffies(ctx->button_release_delay)); + } + + } + } + if(!jack_type){ + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + jack_type = 0; + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } + + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + pr_info("Exit:%s\n", __func__); + mutex_unlock(&ctx->jack_mlock); +} +/* +* added by everest-david +*Used for remove/button debounce +*/ +static void cht_hs_debounce_events(struct work_struct *work) +{ + + int ret; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_release_work.work); + struct snd_soc_codec *codec = ctx->codec; + pr_info("Enter:%s\n", __func__); + + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x06)==0x06){ //remove + ret = snd_soc_read(codec, 0x50); + if(ret == 0xaa) { + gpio_direction_output(368, 1); //1 + } + printk("byt_hs_debounce_remove\n"); + } + else { //inserted + gpio_direction_output(368, 0); + printk("byt_hs_debounce_insert\n"); + } + snd_soc_write(codec,0x30,0x10); +} + +static void cht_hs_poll_events(struct work_struct *work) +{ + + int ret,status; + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_release_work.work); + struct snd_soc_codec *codec = ctx->codec; + //pr_info("Enter:%s\n", __func__); + + mutex_lock(&ctx->poll_mlock); + status = es8316_headset_detect(codec); + if(ctx->poll_status == status){ + ctx->poll_same_cnt++; + } else { + ctx->poll_same_cnt = 0; + } + if(!status){ + //ret = schedule_delayed_work(&ctx->hs_remove_work, + // msecs_to_jiffies(0)); + if(ctx->poll_same_cnt > 2) { + snd_soc_jack_report(jack, status, gpio->report); + ctx->poll_same_cnt = 0; + } else { + if(ctx->poll_cnt < 10){ + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); ctx->poll_cnt++; + } else { + ctx->poll_cnt = 0; + } + } + }else { + if(jack->status != status){ + if(ctx->poll_same_cnt>2) { + snd_soc_jack_report(jack, status, gpio->report); + } + //ret = schedule_delayed_work(&ctx->hs_insert_work, + // msecs_to_jiffies(0)); + } + if(ctx->poll_cnt<10){ + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); + ctx->poll_cnt++; + } else { + ctx->poll_cnt = 0; + } + } + mutex_unlock(&ctx->poll_mlock); + //ret = schedule_delayed_work(&ctx->hs_poll_work, + // msecs_to_jiffies(1000)); + +} + +/* Delayed work for enabling the overcurrent detection circuit + * and interrupt for generating button events */ +static void cht_enable_hs_button_events(struct work_struct *work) +{ +#if 0 + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct snd_soc_codec *codec = jack->codec; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_en_work.work); + + int status = aic31xx_query_jack_status(codec); + if (status == SND_JACK_HEADSET) + ctx->process_button_events = true; + else + ctx->process_button_events = false; + aic31xx_btn_press_intr_enable(codec, ctx->process_button_events); +#endif +} + +static inline struct snd_soc_codec *cht_get_codec(struct snd_soc_card *card) +{ + bool found = false; + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + printk(KERN_ERR " codec->name = %s\n", codec->component.name); + if (!strstr(codec->component.name, "i2c-ESSX8316:00")) { // "i2c-10ES8316:00")) { + pr_debug("codec was %s", codec->component.name); + continue; + } else { + found = true; + break; + } + } + if (found == false) { + pr_err("%s: cant find codec", __func__); + return NULL; + } + return codec; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_codec *codec; + + codec = cht_get_codec(card); + if (!codec) { + pr_err("Codec not found; Unable to set platform clock\n"); + return -EIO; + } + if (SND_SOC_DAPM_EVENT_ON(event)) { + //vlv2_plat_configure_clock(VLV2_PLAT_CLK_AUDIO, + // PLAT_CLK_FORCE_ON); +clk_prepare_enable(mclk); + pr_info("Platform clk turned ON\n"); + } else { + /* Set codec clock source to internal clock before + * turning off the platform clock. Codec needs clock + * for Jack detection and button press + */ + //vlv2_plat_configure_clock(VLV2_PLAT_CLK_AUDIO, + // PLAT_CLK_FORCE_ON); + clk_disable_unprepare(mclk); + pr_info("Platform clk turned OFF\n"); + } + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_MIC("D-MIC", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU| + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { +#if 0 + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKOUTL"}, + {"Ext Spk", NULL, "SPKOUTR"}, + /* AMIC */ + /* + Don't use RVP resource + for HP detection issue on RVP + */ + {"Headset AMIC", NULL, "MIC Bias"}, + {"MONOINP", NULL, "Headset AMIC"}, + {"MONOINN", NULL, "Headset AMIC"}, + + {"AMIC", NULL, "MIC Bias"}, + {"MIC", NULL, "AMIC"}, +#else + {"Ext Spk", NULL, "SPKOL"}, + {"Ext Spk", NULL, "SPKOR"}, + + {"D-MIC", NULL, "micbias"}, + {"Headset Mic", NULL, "micbias"}, + {"Internal Mic", NULL, "micbias"}, + /* Headset Mic: Headset Mic with bias */ + {"MIC1", NULL, "Headset Mic"}, + {"MIC2", NULL, "Internal Mic"}, + {"DMIC", NULL, "D-MIC"}, + + /* Headset Stereophone(Headphone): HSOL, HSOR */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"D-MIC", NULL, "Platform Clock"}, + {"Internal Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +#endif + {"Playback", NULL, "ssp1 Tx"}, + {"ssp1 Tx", NULL, "codec_out0"}, + {"ssp1 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp1 Rx"}, + {"codec_in1", NULL, "ssp1 Rx"}, + {"ssp1 Rx", NULL, "Capture"}, + {"ssp0 Tx", NULL, "modem_out"}, + {"Playback", NULL, "Platform Clock"}, + {"Capture", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("D-MIC"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + + +static int cht_aif1_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; + unsigned int fmt; + int ret; + + pr_debug("Enter:%s", __func__); + + //add for voip call no sound by zm 1211 + if (strncmp(codec_dai->name, "ES8316 HiFi", 11)) + return 0; + + /* I2S Slave Mode*/ + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err("can't set codec DAI configuration %d\n", ret); + return ret; + } + +#if 0 + ret = snd_soc_dai_set_pll(codec_dai, 0, ES8316_PLL_SRC_FRM_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 256); + if (ret < 0) { + pr_err("can't set codec pll***********: %d\n", ret); + return ret; + } + + + if (codec_dai->driver && codec_dai->driver->ops->hw_params) + codec_dai->driver->ops->hw_params(substream, params, codec_dai); + + snd_soc_dai_set_sysclk(codec_dai, ES8316_CLKID_PLLO, CHT_PLAT_CLK_3_HZ, 0); +#endif +#if 0 + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("can't set codec sysclk: %d\n", ret); + return ret; + } +#endif + + snd_soc_dai_set_sysclk(codec_dai, ES8316_CLKID_PLLO, CHT_PLAT_CLK_3_HZ, 0); + + return 0; +} + +static int cht_compr_set_params(struct snd_compr_stream *cstream) +{ + return 0; +} + +static const struct snd_soc_pcm_stream cht_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_compr_ops cht_compr_ops = { + .set_params = cht_compr_set_params, +}; + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + pr_debug("Invoked %s for dailink %s\n", __func__, rtd->dai_link->name); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 4;//2 + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int cht_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + card->dapm.bias_level = level; + pr_debug("card(%s)->bias_level %u\n", card->name, + card->dapm.bias_level); + break; + default: + pr_err("%s: Invalid bias level=%d\n", __func__, level); + ret = -EINVAL; + } + + return ret; +} + +static int cht_audio_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + int ReadBuffer,ReadBuffer1,ReadBuffer2; + struct snd_soc_codec *codec; + struct snd_soc_card *card = runtime->card; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct gpio_desc *desc; + pr_debug("Enter:%s", __func__); + + codec = cht_get_codec(card); + if (!codec) { + pr_err("Codec not found: %s:failed\n", __func__); + return -EIO; + } + ctx->codec = codec; + desc = devm_gpiod_get_index(codec->dev, NULL, 1, 0); + if (!IS_ERR(desc)) { + hs_gpio.gpio = 307;//desc_to_gpio(desc); + devm_gpiod_put(codec->dev, desc); + pr_info("cht-cr GPIOs - JD/BP-int: %d\n", hs_gpio.gpio); + } else { + hs_gpio.gpio = 307; + pr_err("Failed to get gpio desc for Jack det\n"); + } + pr_err("hs codec gpio %d\n", hs_gpio.gpio); + + /* Set codec bias level */ + cht_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + /* Headset jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &ctx->jack, NULL, 0); + if (ret) { + pr_err("Jack creation failed\n"); +// todo return ret; + } + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + + ret = snd_soc_jack_add_gpios(&ctx->jack, 1, &hs_gpio); + if (ret) { + pr_err("Adding jack GPIO failed with error %d\n", ret); +//todo return ret; + } + + + ret = snd_soc_add_card_controls(card, cht_mc_controls, + ARRAY_SIZE(cht_mc_controls)); + if (ret) { + pr_err("unable to add card controls\n"); + return ret; + } + + ret = snd_soc_dapm_sync(&card->dapm); + if (ret) { + pr_err("unable to sync dapm\n"); + return ret; + } + + ReadBuffer = snd_soc_read(codec, ES8316_GPIO_DEBUNCE_INT_REG4E); + // printk("===probe ReadBuffer: 0x%x\n",ReadBuffer); + ReadBuffer1=ReadBuffer | 0xFE; + ReadBuffer2=ReadBuffer1 & 0x01; + if (ReadBuffer2 & 0x01) + { + // printk("===11111111 ReadBuffer2: 0x%x\n",ReadBuffer2); + snd_soc_write(codec, 0X4E, 0xF2); + } + else + { + //printk("===22222222 ReadBuffer2: 0x%x\n",ReadBuffer2); + snd_soc_write(codec, 0X4E, 0xF3); + } + mdelay(10); + snd_soc_write(codec, 0X4E, 0xF3); + ret = snd_soc_read(codec, 0x4F); + //printk("===read 0x4F reg: 0x%x\n", ret); + + + return ret; +} + +static unsigned int rates_8000_16000[] = { + 8000, + 16000, +}; + +static struct snd_pcm_hw_constraint_list constraints_8000_16000 = { + .count = ARRAY_SIZE(rates_8000_16000), + .list = rates_8000_16000, +}; + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static int cht_8k_16k_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_8000_16000); +} + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + .init = cht_audio_init, + .ignore_suspend = 1, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &cht_aif1_ops, + .ignore_suspend = 1, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .compr_ops = &cht_compr_ops, + }, + /* Back ends */ + { + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "ssp1-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "ES8316 HiFi", + .codec_name = "i2c-ESSX8316:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = cht_codec_fixup, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +#ifdef CONFIG_PM_SLEEP +static int snd_cht_prepare(struct device *dev) +{ + pr_debug("In %s device name\n", __func__); + return snd_soc_suspend(dev); +} + +static void snd_cht_complete(struct device *dev) +{ + pr_debug("In %s\n", __func__); + snd_soc_resume(dev); +} + +static int snd_cht_poweroff(struct device *dev) +{ + pr_debug("In %s\n", __func__); + return snd_soc_poweroff(dev); +} +#else +#define snd_cht_prepare NULL +#define snd_cht_complete NULL +#define snd_cht_poweroff NULL +#endif + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailaud", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .set_bias_level = cht_set_bias_level, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + pr_debug("Entry %s\n", __func__); + + /* Audio Platform clock is on by default. The machine driver requests + * this clock to be turned ON and OFF on playing any stream. But + * until any stream is played the clock remains ON. Hence request the + * clock to be turned OFF initially. + */ + //vlv2_plat_configure_clock(VLV2_PLAT_CLK_AUDIO, PLAT_CLK_FORCE_ON); + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + //ret_val = gpio_request(341, "I2S_RESET_GPIO"); + if (ret_val) + { + pr_debug("request gpio fail\n"); + } + + drv->intr_debounce = CHT_INTR_DEBOUNCE; + drv->hs_insert_det_delay = CHT_HS_INSERT_DET_DELAY; + drv->hs_remove_det_delay = CHT_HS_REMOVE_DET_DELAY; + drv->button_press_delay = CHT_BUTTON_PRESS_DELAY; + drv->button_release_delay = CHT_BUTTON_RELEASE_DELAY; + drv->hs_det_poll_intrvl = CHT_HS_DET_POLL_INTRVL; + drv->hs_det_retry = CHT_HS_DET_RETRY_COUNT; + drv->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + drv->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + drv->button_en_delay = CHT_BUTTON_EN_DELAY; + drv->hs_debounce_delay = CHT_HS_DEBOUNCE_DELAY; /*added by everest-david*/ + drv->hs_debounce_retry = CHT_HS_DEBOUNCE_RETRY_COUNT; /*added by everest-david*/ + drv->process_button_events = false; + drv->poll_cnt = 0; + drv->poll_status = 0; + drv->poll_same_cnt = 0; + INIT_DELAYED_WORK(&drv->hs_insert_work, cht_check_hs_insert_status); + INIT_DELAYED_WORK(&drv->hs_remove_work, cht_check_hs_remove_status); + INIT_DELAYED_WORK(&drv->hs_button_press_work, + cht_check_hs_button_press_status); + INIT_DELAYED_WORK(&drv->hs_button_release_work, + cht_check_hs_button_release_status); + INIT_DELAYED_WORK(&drv->hs_button_en_work, cht_enable_hs_button_events); + INIT_DELAYED_WORK(&drv->hs_debounce_work, cht_hs_debounce_events); /*added by everest-david*/ + INIT_DELAYED_WORK(&drv->hs_poll_work, cht_hs_poll_events); /*added by everest-david for hampoo, 16-3-3*/ + mutex_init(&drv->jack_mlock); + mutex_init(&drv->poll_mlock); + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + pr_err("snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + pr_info("%s successful\n", __func__); + mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + return ret_val; +} + +static void snd_cht_unregister_jack(struct cht_mc_private *ctx) +{ + /* Set process button events to false so that the button + delayed work will not be scheduled.*/ + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + cancel_delayed_work_sync(&ctx->hs_poll_work); + snd_soc_jack_free_gpios(&ctx->jack, 1, &hs_gpio); +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct cht_mc_private *drv = snd_soc_card_get_drvdata(soc_card); + + pr_debug("In %s\n", __func__); + snd_cht_unregister_jack(drv); + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static void snd_cht_mc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct cht_mc_private *drv = snd_soc_card_get_drvdata(soc_card); + struct snd_soc_codec *codec = drv->codec; + snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x00); + + pr_debug("In %s\n", __func__); + snd_cht_unregister_jack(drv); +} + +const struct dev_pm_ops snd_cht_cr_mc_pm_ops = { + .prepare = snd_cht_prepare, + .complete = snd_cht_complete, + .poweroff = snd_cht_poweroff, +}; + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-es8316", + .pm = &snd_cht_cr_mc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, + .shutdown = snd_cht_mc_shutdown, +}; + +static int __init snd_cht_driver_init(void) +{ + int ret; + ret = platform_driver_register(&snd_cht_mc_driver); + if (ret) + pr_err("Fail to register Cherrytrail Machine driver cht_es8316\n"); + else + pr_info("Cherrytrail Machine Driver cht_es8316 registerd\n"); + return ret; +} +late_initcall(snd_cht_driver_init); + +static void __exit snd_cht_driver_exit(void) +{ + pr_debug("In %s\n", __func__); + platform_driver_unregister(&snd_cht_mc_driver); +} +module_exit(snd_cht_driver_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) Cherrytrail CR Machine driver"); +MODULE_AUTHOR("Praveen Diwakar <praveen.diwakar(a)intel.com>"); +MODULE_AUTHOR("Bhakte, GurudattaX <gurudattax.bhakte(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht_es8316"); -- 2.9.3

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[alsa-devel] [PATCH v2 0/6] Move dell-led to drivers/platform/x86
by Michał Kępień 16 Feb '17

16 Feb '17

This patch series moves the dell-led driver from the LED subsystem to the x86 platform driver subsystem. The original motivation behind this effort was to move all code using the dell-smbios module to the x86 platform driver subsystem. While I was investigating the possibilities to do that, it quickly emerged that dell-led can and in fact should be moved to the x86 platform driver subsystem in its entirety. dell-led consists of two major parts: - the part exposing a microphone mute LED interface, introduced in db6d8cc00773 ("dell-led: add mic mute led interface"); this interface is used by sound/pci/hda/dell_wmi_helper.c; while the original implementation used a WMI interface, it was changed to use dell-smbios in cf0d7ea33596 ("dell-led: use dell_smbios_find_token() for finding mic DMI tokens") and 0c41a08e131d ("dell-led: use dell_smbios_send_request() for performing SMBIOS calls"), - the part handling an activity LED present in Dell Latitude 2100 netbooks, introduced in 72dcd8d08aca ("leds: Add Dell Business Class Netbook LED driver"); it binds to a specific WMI GUID and then registers a LED device which is controlled using WMI (i.e. it is essentially a WMI driver). Patches 1 and 2 clean up the microphone mute LED interface to minimize the amount of code moved around. Patch 3 updates a variable name in sound/pci/hda/dell_wmi_helper.c so that it better matches that variable's role. Patch 4 moves the microphone mute LED interface to drivers/platform/x86/dell-laptop.c, effectively causing sound/pci/hda/dell_wmi_helper.c to depend on CONFIG_DELL_LAPTOP instead of CONFIG_LEDS_DELL_NETBOOKS. Patch 5 reverts dell-led to the state it was in after its initial commit 72dcd8d08aca ("leds: Add Dell Business Class Netbook LED driver") by removing all remnants of the microphone mute LED handling code. Patch 6 moves all that is left of dell-led (i.e. the activity LED part, as originally implemented), to a new module which is placed in drivers/platform/x86/dell-wmi-led.c. As all patches except patch 3 in this series affect the LED subsystem, the series is based on linux-leds/for-4.11. Anthony, I would be grateful if you could test this patch series on the Dell machines you have access to that were previously supported by dell-led as Jacek needs a Tested-by from someone to sign off on these changes. Please note the Kconfig option rename done by the last patch. Thanks! Changes from v1: - Squash patches 2-4 from v1 into a single patch (#2 in v2). - Add patch 3. - Fix subject pattern in patch 4. - Slight commit message adjustments, including fixing a typo ("COFIG_LEDS_DELL_NETBOOKS") in patch 6. - Remove the name of the module's source file from the header comment in drivers/platform/x86/dell-wmi-led.c to avoid the need to update it in the future. drivers/leds/Kconfig | 9 --- drivers/leds/Makefile | 1 - drivers/platform/x86/Kconfig | 8 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/dell-laptop.c | 28 ++++++++ .../dell-led.c => platform/x86/dell-wmi-led.c} | 75 ++-------------------- include/linux/dell-led.h | 6 +- sound/pci/hda/dell_wmi_helper.c | 30 ++++----- 8 files changed, 60 insertions(+), 98 deletions(-) rename drivers/{leds/dell-led.c => platform/x86/dell-wmi-led.c} (73%) -- 2.11.0

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[alsa-devel] [PATCH] ALSA: hda/patch_sigmatel: Add AmigaOne X1000 pin configs
by Darren Stevens 14 Feb '17

14 Feb '17

The AmigaOne X1000 has a Sigmatel STAC92HD700 attached to the HD Audio on an ATI SB600. Add the required settings to enable sound. Signed-off-by: Darren Stevens <darren(a)stevens-zone.net> ---

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[alsa-devel] [PATCH v7 0/4] DMA Engine: switch PL330 driver to non-irq-safe runtime PM
by Marek Szyprowski 09 Feb '17

09 Feb '17

Hello, This patchset changes the way the runtime PM is implemented in the PL330 DMA engine driver. The main goal of such change is to add support for the audio power domain to Exynos5 SoCs (5250, 542x, 5433, probably others) and let it to be properly turned off, when no audio is being used. Switching to non-irq-safe runtime PM is required to properly let power domain to be turned off (irq-safe runtime PM keeps power domain turned on all the time) and to integrate with clock controller's runtime PM (this cannot be workarounded any other way, PL330 uses clocks from the controller, which belongs to the same power domain). For more details of the proposed change to the PL330 driver see patch #4. Audio power domain on Exynos5 SoCs contains following hardware modules: 1. clock controller 2. pin controller 3. PL330 DMA controller 4. I2S audio controller Patches for adding or fixing runtime PM for each of the above devices is handled separately. Runtime PM patches for clock controllers is possible and has been proposed in the following thread (pending review): "[PATCH v4 0/4] Add runtime PM support for clocks (on Exynos SoC example)", http://www.spinics.net/lists/arm-kernel/msg550747.html Runtime PM support for Exynos pin controller has been posted in the following thread: "[PATCH 0/9] Runtime PM for Exynos pin controller driver", http://www.spinics.net/lists/arm-kernel/msg550161.html Exynos I2S driver supports runtime PM, but some fixes were needed for it and they are already queued to linux-next. This patchset is based on linux-next from 25th January 2017 with "dmaengine: pl330: fix double lock" patch applied. Best regards Marek Szyprowski Samsung R&D Institute Poland Changelog: v7: - added missing of_dma_request_slave_channel API change to sound/soc/sh/rcar driver - extended commit message with information about drawbacks of irq-safe runtime pm - added Ulf's reviewed-by tags v6: https://www.spinics.net/lists/arm-kernel/msg557377.html - fixed pl330 system sleep suspend/resume callbacks, previous implementation incorrectly tried to unprepare clocks unconditionally - after a fix pl330 suspend/resume callbacks can be simply replaced by generic pm_runtime_force_{suspend,resume} helpers, what simplifies code even more v5: https://www.spinics.net/lists/arm-kernel/msg555001.html - added Acks - additional mutex is indeed not needed, rely on dma_list_mutex in dmaengine core, added comment about locking v4: http://www.spinics.net/lists/dmaengine/msg12329.html - rebased onto "dmaengine: pl330: fix double lock" patch: http://www.spinics.net/lists/dmaengine/msg12289.html - added a mutex to protect runtime PM links creation/removal to avoid races - moved mem2mem channel case handing to pl330_{add,del}_slave_pm_link functions to simplify code and error paths v3: http://www.spinics.net/lists/dmaengine/msg12245.html - removed pl330_filter function as suggested by Arnd Bergmann - removed pl330.h from arch/arm/plat-samsung/devs.c - fixes some minor style issues pointed by Krzysztof Kozlowski v2: https://www.spinics.net/lists/arm-kernel/msg552772.html - rebased onto linux next-20170109 - improved patch description - separated patch #3 from #4 (storing a pointer to slave device for each DMA channel) as requested by Krzysztof Kozlowski v1: https://www.spinics.net/lists/arm-kernel/msg550008.html - initial version Patch summary: Marek Szyprowski (4): dmaengine: pl330: remove pdata based initialization dmaengine: Forward slave device pointer to of_xlate callback dmaengine: pl330: Store pointer to slave device dmaengine: pl330: Don't require irq-safe runtime PM arch/arm/plat-samsung/devs.c | 1 - drivers/dma/amba-pl08x.c | 2 +- drivers/dma/at_hdmac.c | 4 +- drivers/dma/at_xdmac.c | 2 +- drivers/dma/bcm2835-dma.c | 2 +- drivers/dma/coh901318.c | 2 +- drivers/dma/cppi41.c | 2 +- drivers/dma/dma-jz4780.c | 2 +- drivers/dma/dmaengine.c | 2 +- drivers/dma/dw/platform.c | 2 +- drivers/dma/edma.c | 4 +- drivers/dma/fsl-edma.c | 2 +- drivers/dma/img-mdc-dma.c | 2 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/imx-sdma.c | 2 +- drivers/dma/k3dma.c | 2 +- drivers/dma/lpc18xx-dmamux.c | 2 +- drivers/dma/mmp_pdma.c | 2 +- drivers/dma/mmp_tdma.c | 2 +- drivers/dma/moxart-dma.c | 2 +- drivers/dma/mxs-dma.c | 2 +- drivers/dma/nbpfaxi.c | 2 +- drivers/dma/of-dma.c | 19 ++-- drivers/dma/pl330.c | 220 ++++++++++++++++------------------------ drivers/dma/pxa_dma.c | 2 +- drivers/dma/qcom/bam_dma.c | 2 +- drivers/dma/sh/rcar-dmac.c | 2 +- drivers/dma/sh/shdma-of.c | 2 +- drivers/dma/sh/usb-dmac.c | 2 +- drivers/dma/sirf-dma.c | 2 +- drivers/dma/st_fdma.c | 2 +- drivers/dma/ste_dma40.c | 2 +- drivers/dma/stm32-dma.c | 2 +- drivers/dma/sun4i-dma.c | 2 +- drivers/dma/sun6i-dma.c | 2 +- drivers/dma/tegra20-apb-dma.c | 2 +- drivers/dma/tegra210-adma.c | 2 +- drivers/dma/xilinx/xilinx_dma.c | 2 +- drivers/dma/xilinx/zynqmp_dma.c | 2 +- drivers/dma/zx_dma.c | 2 +- include/linux/amba/pl330.h | 35 ------- include/linux/of_dma.h | 19 ++-- sound/soc/sh/rcar/dma.c | 5 +- sound/soc/sh/rcar/dvc.c | 3 +- sound/soc/sh/rcar/rsnd.h | 3 +- sound/soc/sh/rcar/src.c | 3 +- sound/soc/sh/rcar/ssi.c | 3 +- 47 files changed, 158 insertions(+), 231 deletions(-) delete mode 100644 include/linux/amba/pl330.h -- 1.9.1

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[alsa-devel] [PATCH v1 1/4] ASoC: Intel: common: Replace custom implementation of readq / writeq
by Andy Shevchenko 09 Feb '17

09 Feb '17

The readq() and writeq() helpers are available in the linux/io-64-nonatomic-hi-lo.h and linux/io-64-nonatomic-lo-hi.h headers. Replace custom implementation by the generic helpers. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/common/sst-dsp.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index 11c0805393ff..748f1f5c02df 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -22,6 +22,8 @@ #include <linux/io.h> #include <linux/delay.h> +#include <linux/io-64-nonatomic-lo-hi.h> + #include "sst-dsp.h" #include "sst-dsp-priv.h" @@ -43,16 +45,13 @@ EXPORT_SYMBOL_GPL(sst_shim32_read); void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + lo_hi_writeq(value, addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_write64); u64 sst_shim32_read64(void __iomem *addr, u32 offset) { - u64 val; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return lo_hi_readq(addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_read64); -- 2.11.0

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