January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 4/4] ASoC: mid-x86 add jack detect support
by Harsha Priya 19 Jan '11

19 Jan '11

This patch adds the jack detection in machine driver and adds the jack detect logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/sn95031.c | 82 +++++++++++++- sound/soc/codecs/sn95031.h | 20 ++++ sound/soc/codecs/sn95031_adc_control.h | 194 ++++++++++++++++++++++++++++++++ sound/soc/mid-x86/mfld_machine.c | 159 +++++++++++++++++++++----- 4 files changed, 424 insertions(+), 31 deletions(-) create mode 100644 sound/soc/codecs/sn95031_adc_control.h diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 5e8d091..c436311 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -27,6 +27,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/input.h> #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -35,13 +36,15 @@ #include <sound/initval.h> #include <sound/tlv.h> #include "sn95031.h" +#include "sn95031_adc_control.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +struct snd_soc_codec *sn95031_codec; + /* * todo: - * jack detection * PM functions */ @@ -113,6 +116,15 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, return 0; } +/* enables mic bias voltage */ +void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, + BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, + BIT(2), BIT(2)); +} + static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -310,6 +322,7 @@ static const struct snd_kcontrol_new sn95031_snd_controls[] = { static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { /* all end points mic, hs etc */ + SND_SOC_DAPM_HP("HeadPhones", NULL), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("EPOUT"), @@ -428,6 +441,8 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { static const struct snd_soc_dapm_route sn95031_audio_map[] = { /* headset and earpiece map */ + { "HeadPhones", NULL, "HPOUTR" }, + { "HeadPhones", NULL, "HPOUTL" }, { "HPOUTL", NULL, "Headset Left Playback" }, { "HPOUTR", NULL, "Headset Right Playback" }, { "EPOUT", NULL, "Earpiece Playback" }, @@ -658,6 +673,66 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +int sn95031_get_headset_state(void) +{ + unsigned int mic_bias = sn95031_get_mic_bias(sn95031_codec); + + if (mic_bias >= SN_HP_START && mic_bias < SN_HP_END) { + pr_debug("sst: Detected Headphone!!!\n"); + snd_soc_update_bits(sn95031_codec, AUDIO_GPIO_CTRL, + BIT(1), BIT(0)); + } else if (mic_bias >= SN_AM_HS_START && mic_bias < SN_AM_HS_END) { + pr_debug("sst: Detected American headset\n"); + snd_soc_update_bits(sn95031_codec, AUDIO_GPIO_CTRL, + BIT(1), BIT(1)); + } else if (mic_bias >= SN_HS_START && mic_bias < SN_HS_END) { + pr_debug("sst: Detected Headset!!!\n"); + snd_soc_update_bits(sn95031_codec, AUDIO_GPIO_CTRL, + BIT(1), BIT(0)); + enable_jack_btn(sn95031_codec); + return SND_JACK_HEADSET; + } else + pr_debug("sst: Detected Open Cable!!!\n"); + + return SND_JACK_HEADPHONE; +} + +void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg) +{ + unsigned int status, mask; + static int jack_state = SND_JACK_HEADSET; + + if (!sn95031_codec) { + pr_err("codec is null\n"); + return; + } + pr_debug("interrupt id read in sram = 0x%x\n", jack_msg->intr_id); + if (jack_msg->intr_id & 0x1) { + pr_debug("short_push detected\n"); + mask = status = SND_JACK_BTN_0; + } else if (jack_msg->intr_id & 0x2) { + pr_debug("long_push detected\n"); + mask = status = SND_JACK_BTN_1; + } else if (jack_msg->intr_id & 0x4) { + mask = status = sn95031_get_headset_state(); + jack_state = mask; + } else if (jack_msg->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + mask = jack_state; + disable_jack_btn(sn95031_codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(mfld_jack, status, mask); + /*button pressed and released */ + if (status == SND_JACK_BTN_0 || status == SND_JACK_BTN_1) + snd_soc_jack_report(mfld_jack, 0, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { @@ -667,6 +742,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.idle_bias_off = 1; + sn95031_codec = codec; /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots @@ -733,7 +809,7 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sn95031_codec = { +struct snd_soc_codec_driver sn95031_soc_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, .read = sn95031_read, @@ -744,7 +820,7 @@ struct snd_soc_codec_driver sn95031_codec = { static int __devinit sn95031_device_probe(struct platform_device *pdev) { pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); - return snd_soc_register_codec(&pdev->dev, &sn95031_codec, + return snd_soc_register_codec(&pdev->dev, &sn95031_soc_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d9..e79176d 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -25,6 +25,7 @@ */ #ifndef _SN95031_H #define _SN95031_H +#include <sound/jack.h> /*register map*/ #define SN95031_VAUD 0xDB @@ -96,4 +97,23 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +struct mfld_jack_msg_wq { + u8 intr_id; + struct work_struct wq; +}; + +static inline void disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +extern void sn95031_jack_detection(struct snd_soc_jack *mfld_jack, + struct mfld_jack_msg_wq *jack_msg); + #endif diff --git a/sound/soc/codecs/sn95031_adc_control.h b/sound/soc/codecs/sn95031_adc_control.h new file mode 100644 index 0000000..0971321 --- /dev/null +++ b/sound/soc/codecs/sn95031_adc_control.h @@ -0,0 +1,194 @@ +#ifndef __MFLD_MACHINE_ADC_CONTROL_H__ +#define __MFLD_MACHINE_ADC_CONTROL_H_ +/* + * sn95031_adc_control.h - Module for adc programming + * + * Copyright (C) 2008-10 Intel Corporation + * Authors: R Durgadadoss <r.durgadoss(a)intel.com> + * Dharageswari R <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 + +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 + +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 + +#define ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define ADC_LOOP_MAX (ADC_CHANLS_MAX - 1) + +/* ADC channel code values */ +#define AUDIO_DETECT_CODE 0x06 + +#define AUDIO_GPIO_CTRL 0x070 + +/* ADC base addresses */ +#define ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ + +/* multipier to convert to mV */ +#define ADC_ONE_LSB_MULTIPLIER 2346 + +enum MIC_BIAS_ZONES { + SN_HP_START = 0, + SN_HP_END = 400, + SN_AM_HS_START = 400, + SN_AM_HS_END = 650, + SN_HS_START = 650, + SN_HS_END = 2000, + SN_UNDEFINED, +}; + +void sn95031_enable_mic_bias(struct snd_soc_codec *codec); + +/** + * configure_adc - enables/disables the ADC for conversion + * @val: zero: disables the ADC non-zero:enables the ADC + * + * Enable/Disable the ADC depending on the argument + * + * Can sleep + */ +static inline void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + else + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + +/** + * find_free_channel - finds an empty channel for conversion + * + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * + */ +static inline int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/** + * sn95031_initialize_adc - initializing the ADC + * @dev: our device structure + * + * Initialize the ADC for reading thermistor values. Can sleep. + */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, AUDIO_DETECT_CODE | 0x10); + + chnl_addr = ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/** + * sn95031_get_mic_bias - get the mic bias in mV + * + * reads the ADC registers and gets the mic bias value in mV. + */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + /* convesion into mV */ + mic_bias = (ADC_ONE_LSB_MULTIPLIER * mic_bias) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +#endif + diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1cd49a2..7926151 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -27,6 +27,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -35,10 +36,32 @@ #define MID_MONO 1 #define MID_STEREO 2 #define MID_MAX_CAP 5 +#define IRQ_BASE 0xFFFF7FCD +#define IRQ_SIZE 1 static unsigned int hs_switch; static unsigned int lo_dac; +struct mfld_mc_private { + struct platform_device *socdev; + struct workqueue_struct *mfld_jack_wq; + void __iomem *int_base; +}; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "HeadPhones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -67,13 +90,11 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) { pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } snd_soc_dapm_sync(&codec->dapm); @@ -91,12 +112,10 @@ static void lo_enable_out_pins(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } } @@ -130,8 +149,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, case 1: pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "HeadPhones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); break; @@ -175,8 +193,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(dapm, "HeadPhones"); /* default is lineout NC, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); @@ -185,11 +202,31 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - /* we are using DMICs here so configure how the clock and out pin is set */ + /* we are using DMICs here so configure how the + * clock and out pin is set */ snd_soc_write(codec, SN95031_DMICBUF0123, 0x85); snd_soc_write(codec, SN95031_DMICBUF45, 0x02); snd_soc_update_bits(codec, SN95031_DMICMUX, 0x7, 0x07); - return snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + /* Headset jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + disable_jack_btn(codec); + return ret_val; + } struct snd_soc_dai_link mfld_msic_dailink[] = { @@ -238,37 +275,103 @@ static struct snd_soc_card snd_soc_card_mfld = { .num_links = ARRAY_SIZE(mfld_msic_dailink), }; +void snd_mfld_jack_detection(struct work_struct *work) +{ + struct mfld_jack_msg_wq *jack_msg = + container_of(work, struct mfld_jack_msg_wq, wq); + + sn95031_jack_detection(&mfld_jack, jack_msg); + kfree(jack_msg); +} + +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_jack_msg_wq *jack_msg; + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + jack_msg = kzalloc(sizeof(*jack_msg), GFP_ATOMIC); + memcpy_fromio(&jack_msg->intr_id, + ((void *)(mc_private->int_base)), + sizeof(u8)); + pr_debug("intr_handler called value %x\n", jack_msg->intr_id); + INIT_WORK(&jack_msg->wq, snd_mfld_jack_detection); + queue_work(mc_private->mfld_jack_wq, &jack_msg->wq); + + return IRQ_HANDLED; +} + static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) { - struct platform_device *socdev; - int ret_val = 0; + int ret_val = 0, irq; + struct mfld_mc_private *mc_private_data; pr_debug("snd_mfld_mc_probe called\n"); - socdev = platform_device_alloc("soc-audio", -1); - if (!socdev) { - pr_err("soc-audio device allocation failed\n"); + mc_private_data = kzalloc(sizeof(*mc_private_data), GFP_ATOMIC); + if (!mc_private_data) { + pr_err("mem alloation of mc_private failed\n"); return -ENOMEM; } - platform_set_drvdata(socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(socdev); + + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_private_data->int_base = ioremap_nocache(IRQ_BASE, IRQ_SIZE); + if (!mc_private_data->int_base) { + pr_err("Mapping of cache failed\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* create botoom half handler for interrupt */ + mc_private_data->mfld_jack_wq = create_workqueue("mfld_jack_wq"); + if (!mc_private_data->mfld_jack_wq) { + pr_err("cannot create jack workqueue\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* register for interrupt */ + ret_val = request_irq(irq, snd_mfld_jack_intr_handler, + IRQF_SHARED, pdev->dev.driver->name, + mc_private_data); + if (ret_val) { + pr_err("cannot register IRQ\n"); + goto destroy_wq; + } + /* create soc device */ + mc_private_data->socdev = platform_device_alloc("soc-audio", -1); + if (!mc_private_data->socdev) { + pr_err("soc-audio device allocation failed\n"); + ret_val = -ENOMEM; + goto destroy_wq; + } + platform_set_drvdata(mc_private_data->socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(mc_private_data->socdev); if (ret_val) { pr_err("Unable to add soc-audio device, err %d\n", ret_val); - platform_device_put(socdev); + goto unregister; } - - platform_set_drvdata(pdev, socdev); - + platform_set_drvdata(pdev, mc_private_data); pr_debug("successfully exited probe\n"); return ret_val; + +unregister: + platform_device_put(mc_private_data->socdev); +destroy_wq: + destroy_workqueue(mc_private_data->mfld_jack_wq); +unalloc: + kfree(mc_private_data); + return ret_val; } static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) { - struct platform_device *socdev = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); + struct mfld_mc_private *mc_private_data = platform_get_drvdata(pdev); - platform_device_unregister(socdev); + pr_debug("snd_mfld_mc_remove called\n"); + free_irq(platform_get_irq(pdev, 0), mc_private_data); + destroy_workqueue(mc_private_data->mfld_jack_wq); + platform_device_unregister(mc_private_data->socdev); platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.2.3

3 3

[alsa-devel] [PATCH] ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()
by Dimitris Papastamos 19 Jan '11

19 Jan '11

In the wm8995_set_tristate() function when updating the register bits use the value and not the register index as the value argument. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8995.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 7d56341..67eaaec 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1265,7 +1265,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } /* The size in bits of the FLL divide multiplied by 10 -- 1.7.3.5

3 2

Re: [alsa-devel] [PATCH] Asoc: WM8994: fix wrong value in tristate function
by Mark Brown 19 Jan '11

19 Jan '11

On Wed, Jan 19, 2011 at 07:10:47PM +0800, Qiao Zhou wrote: > fix wrong value in wm8994_set_tristate func. when updating reg bits, > it should use "value", not "reg". > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> Applied, thanks.

1 0

[alsa-devel] [PATCH 1/4] ASoC: sn95031: add capture support
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch adds the support for capture path in sn95031 codec. This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs are connected through a MUX to ADC. The TX paths can be assigned to any of the ADCs or DMICs. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/codecs/sn95031.c | 292 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 291 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 593632c..5e8d091 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -33,6 +33,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -40,7 +41,6 @@ /* * todo: - * capture paths * jack detection * PM functions */ @@ -145,6 +145,167 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, return 0; } +static int amic1_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int value = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("AMIC1 SND_SOC_DAPM_EVENT_ON doing\n"); + value = BIT(2); + } + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(2), value); + return 0; +} + +static int amic2_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int value = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("AMIC2 SND_SOC_DAPM_EVENT_ON doing\n"); + value = BIT(3); + } + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(3), value); + return 0; +} + +static int dmic12_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk = 0, out = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("DMIC12 SND_SOC_DAPM_EVENT_ON doing\n"); + ldo = BIT(5)|BIT(4); + clk = BIT(0); + out = BIT(3); + } + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + msleep(1); + /* enable/disable DMIC clock and o/p */ + snd_soc_update_bits(w->codec, SN95031_DMICLK, BIT(0), clk); + snd_soc_update_bits(w->codec, SN95031_DMICMUX, BIT(3), out); + return 0; +} + +static int dmic34_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk = 0, out = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("DMIC34 SND_SOC_DAPM_EVENT_ON doing\n"); + ldo = BIT(5)|BIT(4); + clk = BIT(1); + out = BIT(4); + } + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + msleep(1); + /* enable/disable DMIC clock */ + snd_soc_update_bits(w->codec, SN95031_DMICLK, BIT(1), clk); + snd_soc_update_bits(w->codec, SN95031_DMICMUX, BIT(4), out); + return 0; +} + +static int dmic56_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk = 0, out = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("DMIC56 SND_SOC_DAPM_EVENT_ON doing\n"); + ldo = BIT(7)|BIT(6); + clk = BIT(2); + out = BIT(5); + } + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); + msleep(1); + /* enable/disable DMIC clock */ + snd_soc_update_bits(w->codec, SN95031_DMICLK, BIT(2), clk); + snd_soc_update_bits(w->codec, SN95031_DMICMUX, BIT(5)|BIT(2), out); + return 0; +} +/* mux controls */ +static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; + +static const struct soc_enum sn95031_micl_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micl_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micl_enum); + +static const struct soc_enum sn95031_micr_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micr_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micr_enum); + +static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", + "DMIC4", "DMIC5", "DMIC6", + "ADC Left", "ADC Right" }; + +static const struct soc_enum sn95031_input1_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input1_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input1_enum); + +static const struct soc_enum sn95031_input2_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input2_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input2_enum); + +static const struct soc_enum sn95031_input3_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input3_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input3_enum); + +static const struct soc_enum sn95031_input4_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input4_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input4_enum); + +/* capture path controls */ + +/* 0dB to 30dB in 10dB steps */ +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); + +static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; + +static const struct soc_enum sn95031_micmode1_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); +static const struct soc_enum sn95031_micmode2_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); + +static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; + +static const struct soc_enum sn95031_dmic12_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic34_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic56_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); + +static const struct snd_kcontrol_new sn95031_snd_controls[] = { + SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), + SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum), + SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum), + SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum), + SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum), + SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1, + 2, 4, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2, + 2, 4, 0, mic_tlv), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { @@ -159,6 +320,20 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIB1OUT"), SND_SOC_DAPM_OUTPUT("VIB2OUT"), + SND_SOC_DAPM_MIC("AMIC1", amic1_mic_bias), /* headset mic */ + SND_SOC_DAPM_MIC("AMIC2", amic2_mic_bias), + SND_SOC_DAPM_MIC("DMIC1", dmic12_mic_bias), + SND_SOC_DAPM_MIC("DMIC2", dmic12_mic_bias), + SND_SOC_DAPM_MIC("DMIC3", dmic34_mic_bias), + SND_SOC_DAPM_MIC("DMIC4", dmic34_mic_bias), + SND_SOC_DAPM_MIC("DMIC5", dmic56_mic_bias), + SND_SOC_DAPM_MIC("DMIC6", dmic56_mic_bias), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + + SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, sn95031_vhs_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -209,6 +384,46 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SN95031_VIB1C5, 1, 0), SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", SN95031_VIB2C5, 1, 0), + + /* capture widgets */ + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "LineIn Enable Left", + SN95031_MICAMP1, 7, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "LineIn Enable Right", + SN95031_MICAMP2, 7, 1, 1, 0), + + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "MIC1 Enable", SN95031_MICAMP1, + 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "MIC2 Enable", SN95031_MICAMP2, + 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX1 Enable", SN95031_AUDIOTXEN, + 2, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX2 Enable", SN95031_AUDIOTXEN, + 3, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX3 Enable", SN95031_AUDIOTXEN, + 4, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_switch, "TX4 Enable", SN95031_AUDIOTXEN, + 5, 1, 1, 0), + + /* ADC have null stream as they will be turned ON by TX path */ + SND_SOC_DAPM_ADC("ADC Left", NULL, + SN95031_ADCCONFIG, 0, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, + SN95031_ADCCONFIG, 2, 0), + + SND_SOC_DAPM_MUX("Mic_InputL Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control), + SND_SOC_DAPM_MUX("Mic_InputR Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control), + + SND_SOC_DAPM_MUX("Txpath1 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control), + SND_SOC_DAPM_MUX("Txpath2 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control), + SND_SOC_DAPM_MUX("Txpath3 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control), + SND_SOC_DAPM_MUX("Txpath4 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control), + }; static const struct snd_soc_dapm_route sn95031_audio_map[] = { @@ -250,6 +465,69 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Lineout Right Playback", NULL, "Headset Right Filter"}, { "Lineout Right Playback", NULL, "Speaker Right Filter"}, { "Lineout Right Playback", NULL, "Vibra2 DAC"}, + + /* Headset (AMIC1) mic */ + { "MIC1 Enable", NULL, "AMIC1"}, + { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"}, + + /* AMIC2 */ + { "MIC2 Enable", NULL, "AMIC2"}, + { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"}, + + + /* Linein */ + { "LineIn Enable Left", NULL, "LINEINL"}, + { "LineIn Enable Right", NULL, "LINEINR"}, + { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"}, + { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"}, + + /* ADC connection */ + { "ADC Left", NULL, "Mic_InputL Capture Route"}, + { "ADC Right", NULL, "Mic_InputR Capture Route"}, + + /*TX path inputs*/ + { "Txpath1 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath2 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath3 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath4 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath1 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath1 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath2 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath3 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath4 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath1 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath2 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath3 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath4 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath1 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath2 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath3 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath4 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath1 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath2 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath3 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath4 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath1 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath2 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath3 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath4 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath1 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath2 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath3 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath4 Capture Route", "DMIC6", "DMIC6"}, + + /* tx path */ + { "TX1 Enable", NULL, "Txpath1 Capture Route"}, + { "TX2 Enable", NULL, "Txpath2 Capture Route"}, + { "TX3 Enable", NULL, "Txpath3 Capture Route"}, + { "TX4 Enable", NULL, "Txpath4 Capture Route"}, + { "PCM_Out", NULL, "TX1 Enable"}, + { "PCM_Out", NULL, "TX2 Enable"}, + { "PCM_Out", NULL, "TX3 Enable"}, + { "PCM_Out", NULL, "TX4 Enable"}, }; /* speaker and headset mutes, for audio pops and clicks */ @@ -339,6 +617,13 @@ struct snd_soc_dai_driver sn95031_dais[] = { .rates = SN95031_RATES, .formats = SN95031_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, .ops = &sn95031_headset_dai_ops, }, { .name = "SN95031 Speaker", @@ -390,6 +675,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32); /* pcm port setting * This sets the pcm port to slave and clock at 19.2Mhz which * can support 6slots, sampling rate set per stream in hw-params @@ -423,6 +710,9 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); + snd_soc_add_controls(codec, sn95031_snd_controls, + ARRAY_SIZE(sn95031_snd_controls)); + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, ARRAY_SIZE(sn95031_dapm_widgets)); if (ret) -- 1.7.2.3

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[alsa-devel] [PATCH 3/4] ASoC: mid-x86: Add support for capture in machine driver
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> This configures the capture unused pins and sets the DMIC cfg by default (also done by user) Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be..1cd49a2 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -182,6 +182,13 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + /* we are using DMICs here so configure how the clock and out pin is set */ + snd_soc_write(codec, SN95031_DMICBUF0123, 0x85); + snd_soc_write(codec, SN95031_DMICBUF45, 0x02); + snd_soc_update_bits(codec, SN95031_DMICMUX, 0x7, 0x07); return snd_soc_dapm_sync(dapm); } -- 1.7.2.3

3 3

[alsa-devel] [PATCH 0/4] ASoC: patch series to add support for capture and jack sense in mid-x86 drivers
by Harsha, Priya 19 Jan '11

19 Jan '11

Hi Mark, This patch set adds support for capture and jack sense in mid-x6 platform, machine and TI sn95031 codec drivers. [PATCH 1/4] ASoC: sn95031: add capture support This patch adds support for capture codec dais and widgets [PATCH 2/4] ASoC: sst_platform: add support for capture stream on headset dai This patch adds mid-x86 sst platform driver support for capture on headset dai [PATCH 3/4] ASoC: mid-x86: Add support for capture in machine driver This patch adds support for enabling capture in mfld machine driver [PATCH 4/4] ASoC: mid-x86 add jack detect support This patch adds support for jack detection and reporting support in sn95031 codec and mfld_machine driver -Harsha

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[alsa-devel] [PATCH 3/4] ASoC: mid-x86: Add support for capture in machine driver
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> This configures the capture unused pins and sets the DMIC cfg by default (also done by user) Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/mid-x86/mfld_machine.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be..1cd49a2 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -182,6 +182,13 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); + /* we are using DMICs here so configure how the clock and out pin is set */ + snd_soc_write(codec, SN95031_DMICBUF0123, 0x85); + snd_soc_write(codec, SN95031_DMICBUF45, 0x02); + snd_soc_update_bits(codec, SN95031_DMICMUX, 0x7, 0x07); return snd_soc_dapm_sync(dapm); } -- 1.7.2.3

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Re: [alsa-devel] [PATCH 0/3] Tegra: Add internal speaker support
by Liam Girdwood 19 Jan '11

19 Jan '11

On Tue, 2011-01-18 at 16:37 -0700, Stephen Warren wrote: > From: Stephen Warren <swarren(a)nvidia.com> > > Harmony has headers on the board for an internal speaker and internal mic. > I've started looking at enabling these, along with full jack detection etc. > This first set of patches is an RFC (Request For Comments) on whether it's > the correct direction, particularly with respect to: > > a) Is this the right way to expose GPIO APIs from a codec? The tlv320aic3 > seems to do it this way, but wm8962.c uses the generic GPIO API. I shied > away from the latter, since I wasn't sure how to name the WM8903's GPIOs > in the gpio_* calls. I suppose it'd need a bunch of GPIO driver platform > data to hook it all together and name GPIOs from a mach-tegra/include > header file, but if this is the way to go, I need to read up on that more. > > b) Is the WM8903 platform_data handling full fleshed out; simply by having > platform_data, WM8903_WSEQ_ENA is set, whereas without any pdata, it > isn't. Should this be conditional, or should I always have specified some > platform data for Tegra, and I'm just getting lucky that it works without > any? > > Thanks for any coments. > 2 & 3 look fine and Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH 2/4] ASoC: sst_platform: add support for capture stream on headset dai
by Harsha Priya 19 Jan '11

19 Jan '11

From: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/mid-x86/sst_platform.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index a4e3fa3..0d8c430 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -73,6 +73,12 @@ struct snd_soc_dai_driver sst_platform_dai[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, }, { .name = "Speaker-cpu-dai", -- 1.7.2.3

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Re: [alsa-devel] [PATCH] Asoc: WM8994: fix wrong value in tristate function
by Liam Girdwood 19 Jan '11

19 Jan '11

On Wed, 2011-01-19 at 19:10 +0800, Qiao Zhou wrote: > fix wrong value in wm8994_set_tristate func. when updating reg bits, > it should use "value", not "reg". > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

1 0