January 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] LAC 2018 : Call for Papers / Posters / Workshops/ Music Performances / Multimedia Installations
by Robin Gareus 12 Jan '18

12 Jan '18

[Apologies for cross-postings] [Please distribute] Conference date: 7th - 10th June 2018 The Linux Audio Conference 2018 will be hosted at c-base, Berlin - in partnership with the Electronic Music Studio (TU Berlin) and Spektrum. https://lac.linuxaudio.org/2018 Deadline for all submissions: Friday, February 28th, 2018 (23:59 UTC) - - - 1: Call for Papers / Posters / Workshops LAC 2018 invites submissions of papers, posters and workshops addressing all areas of audio processing based on Linux and open source software. All submissions and presentations are in English. All submissions are peer reviewed by a committee of experts from different disciplines. Submissions can focus on technical, artistic or scientific issues and can target developers or users. For more details see the website: https://lac.linuxaudio.org/2018/pages/cfp/ *Papers* Papers must be written and presented in English. The length of papers is 4 to 8 pages, with up to 5 keywords, and an abstract of up to 200 words. The poster presentations are based on paper submissions of 2-4 pages, with up to 5 keywords and an abstract of up to 150 words. *Workshops* The workshop presentation (max duration of 2h) should be 1-4 pages, with up to 5 keywords, and an abstract of up to 150 words to be published on the conference website. Submit a brief description of the workshop including URL (if available). How to submit? * Use the OpenConf online submission tool at https://lac.linuxaudio.org/2018/openconf * Select the relevant submission type ('_ PAPER _' / '_ POSTER _' / '_ WORKSHOP _') * The required file format is PDF, formatted for A4 paper. Authors are required to use the templates for paper formatting available as download on the conference website. * Please notify us if you need a special technical setup for your presentation. - - - 2: Call for Music Performances / Multimedia Installations LAC 2018 also invites submissions of Electroacoustic Works and Multimedia Installations. A jury will select the compositions and installations to be included in the conference program according to artistic merit and technical feasibility. Please be prepared to perform your work yourself and make sure that you have all resources needed to perform your piece at your disposal (e.g. instruments, other performers, etc.). Unfortunately, LAC 2018 cannot pay a fee neither for you nor for any additional performers, and LAC 2018 cannot organize or pay for travel or accommodation neither for you nor for any additional performers. All submissions are peer reviewed by a committee of experts in music and arts. Electroacoustic Works and Multimedia Installation can address all areas of digital audio and audiovisual art. Available Technical Setups For concerts, the LAC will provide the following equipment: * 8-channel - speaker setup * Stereo setup + video projection * Digital mixing desk Additional requests can not be guaranteed. How to submit? * Use the OpenConf online submission tool at https://lac.linuxaudio.org/2018/openconf * Select the submission type '_ PERFORMANCE _' * The required file format is PDF, formatted for A4 paper. It includes: * Description of the project program notes * Link to video or audio demonstration of the project * Technical rider of the work Important Dates * Deadline for all submissions: Friday, February 28th, 2018 (23:59 UTC) * Acceptance notification: March 31st, 2018 * Final deadline for 'camera ready' paper: April 15th, 2018 * Author registration deadline: April 15th, 2018 * Final program: May 1st, 2018 Up-to-date information regarding the conference can be found on its website: https://lac.linuxaudio.org/2018 Looking forward to seeing you in Berlin! The Linux Audio Conference 2018 team

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[alsa-devel] [rfc] Sound support for n9
by Pavel Machek 12 Jan '18

12 Jan '18

Hi! Binding documentation is pending, and code will need to be updated to match it. I guess burst_bclkdiv should be handled as int, not as u8? SPDX is modern these days. Anything else that needs to be fixed? Best regards, Pavel diff --git a/include/sound/n9.h b/include/sound/n9.h new file mode 100644 index 0000000..b14ddf0 --- /dev/null +++ b/include/sound/n9.h @@ -0,0 +1,17 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (C) 2009 Nokia + +#ifndef _N9_H_ +#define _N9_H_ + +struct dfl61audio_hsmic_event { + void *private; + void (*event)(void *priv, bool on); +}; + +void dfl61_jack_report(int status); +int dfl61_request_hsmicbias(bool enable); +void dfl61_register_hsmic_event_cb(struct dfl61audio_hsmic_event *event); +int dfl61_request_hp_enable(bool enable); +#endif diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5b94a15..3823bcc 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +32,7 @@ #include <linux/interrupt.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1484,16 +1487,11 @@ static struct snd_soc_dai_driver dac33_dai = { static int dac33_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_platform_data *pdata = client->dev.platform_data; struct tlv320dac33_priv *dac33; + struct device_node *np = client->dev.of_node; int ret, i; - if (client->dev.platform_data == NULL) { - dev_err(&client->dev, "Platform data not set\n"); - return -ENODEV; - } - pdata = client->dev.platform_data; - dac33 = devm_kzalloc(&client->dev, sizeof(struct tlv320dac33_priv), GFP_KERNEL); if (dac33 == NULL) @@ -1505,10 +1503,26 @@ static int dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); - dac33->power_gpio = pdata->power_gpio; - dac33->burst_bclkdiv = pdata->burst_bclkdiv; - dac33->keep_bclk = pdata->keep_bclk; - dac33->mode1_latency = pdata->mode1_latency; + if (pdata) { + dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; + dac33->keep_bclk = pdata->keep_bclk; + dac33->mode1_latency = pdata->mode1_latency; + } else if (np) { + ret = of_get_named_gpio(np, "power-gpio", 0); + if (ret >= 0) + dac33->power_gpio = ret; + else + dac33->power_gpio = -1; + + if (of_property_read_bool(np, "keep-bclk")) + dac33->keep_bclk = true; + + of_property_read_u8(np, "burst-bclkdiv", &dac33->burst_bclkdiv); + } else { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } if (!dac33->mode1_latency) dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; @@ -1574,9 +1588,16 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); +static const struct of_device_id tlv320dac33_of_match[] = { + { .compatible = "ti,tlv320dac33", }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_of_match); + static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", + .of_match_table = of_match_ptr(tlv320dac33_of_match), }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index f5451c7..2772414 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -47,6 +47,18 @@ config SND_OMAP_SOC_RX51 Say Y if you want to add support for SoC audio on Nokia N900 cellphone. +config SND_OMAP_SOC_N9 + tristate "SoC Audio support for Nokia N9/N950" + depends on SND_OMAP_SOC && ARM && I2C + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + select SND_SOC_TLV320DAC33 + select SND_SOC_TPA6130A2 + select SND_SOC_WL1273 if MFD_WL1273_CORE + depends on GPIOLIB + help + Say Y if you want to add support for SoC audio on Nokia N9/N950. + config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA && TTY diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index a6785dc..07efd21 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -15,6 +15,7 @@ obj-$(CONFIG_SND_OMAP_SOC_HDMI_AUDIO) += snd-soc-omap-hdmi-audio.o # OMAP Machine Support snd-soc-n810-objs := n810.o snd-soc-rx51-objs := rx51.o +snd-soc-n9-objs := n9.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-am3517evm-objs := am3517evm.o @@ -24,6 +25,7 @@ snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o +obj-$(CONFIG_SND_OMAP_SOC_N9) += snd-soc-n9.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o diff --git a/sound/soc/omap/n9.c b/sound/soc/omap/n9.c new file mode 100644 index 0000000..db61463 --- /dev/null +++ b/sound/soc/omap/n9.c @@ -0,0 +1,748 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// n9.c -- SoC audio for Nokia N9/N950 +// +// Copyright (C) 2008 - 2009 Nokia Corporation +// +// Contact: Peter Ujfalusi <peter.ujfalusi(a)ti.com> +// Eduardo Valentin <eduardo.valentin(a)nokia.com> +// Jarkko Nikula <jarkko.nikula(a)bitmer.com> +// + +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <linux/platform_data/asoc-ti-mcbsp.h> +#include <linux/mfd/twl4030-audio.h> +#include "../codecs/tlv320dac33.h" +#include "../codecs/tpa6130a2.h" +#include "../codecs/wl1273.h" + +#include <asm/mach-types.h> + +#include <sound/n9.h> +#include "omap-mcbsp.h" +#include "mcbsp.h" + +#define JACK_REPORT_MASK (SND_JACK_MECHANICAL | SND_JACK_AVOUT | \ + SND_JACK_HEADSET) + +struct dfl61wl1273_audio_pdata { + struct gpio_desc *power_gpio; +}; + +struct dfl61twl_audio_pdata { + struct gpio_desc *speaker_amp_gpio; +}; + +static int dfl61dac33_interconnect_enable(int); +static struct snd_soc_card dfl61dac33_sound_card; +static struct snd_soc_card dfl61twl_sound_card; +static struct snd_soc_jack dfl61_jack; +static struct dfl61audio_hsmic_event *hsmic_event; + +static struct snd_soc_component *find_component(struct snd_soc_card *card) { + struct snd_soc_component *component; + + if (list_empty(&card->component_dev_list)) { + pr_err("Can't find codec for %s\n", card->name); + return NULL; + } + + component = list_entry(card->component_dev_list.next, + struct snd_soc_component, card_list); + + return component; +} + +/* TWL4030 */ +void dfl61_jack_report(int status) +{ + if (dfl61_jack.card) + snd_soc_jack_report(&dfl61_jack, status, JACK_REPORT_MASK); + else + pr_err("twl4030: Cannot report jack status"); +} +EXPORT_SYMBOL_GPL(dfl61_jack_report); + +int dfl61_request_hsmicbias(bool enable) +{ + struct snd_soc_component *component; + struct snd_soc_dapm_context *dapm; + bool lock = false; + int ret; + + if (!dfl61twl_sound_card.instantiated) { + pr_warn("twl4030: sound card not instantiated yet"); + return -EPROBE_DEFER; + } + + component = find_component(&dfl61twl_sound_card); + if (!component) + return -ENODEV; + + dapm = snd_soc_component_get_dapm(component); + if (!dapm) { + pr_err("twl4030: Cannot set hsmicbias yet"); + return -ENODEV; + } + + mutex_lock(&dfl61twl_sound_card.dapm_mutex); + lock = true; + + if (enable) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Headset Mic Bias"); + else + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic Bias"); + + ret = snd_soc_dapm_sync_unlocked(dapm); + + if (lock) + mutex_unlock(&dfl61twl_sound_card.dapm_mutex); + + return ret; +} +EXPORT_SYMBOL(dfl61_request_hsmicbias); + +void dfl61_register_hsmic_event_cb(struct dfl61audio_hsmic_event *event) +{ + hsmic_event = event; +} +EXPORT_SYMBOL(dfl61_register_hsmic_event_cb); + +static int dfl61twl_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int fmt; + int r; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + /* Set codec DAI configuration */ + r = snd_soc_dai_set_fmt(rtd->codec_dai, fmt); + if (r < 0) { + pr_err("Can't set codec DAI configuration for twl4030: %d\n", r); + return r; + } + + /* Set cpu DAI configuration */ + r = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (r < 0) { + pr_err("Can't set cpu DAI configuration for twl4030: %d\n", r); + return r; + } + + return 0; +} + +static int dfl61twl_spk_event(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 dfl61twl_audio_pdata *pdata = snd_soc_card_get_drvdata(card); + + gpiod_set_raw_value_cansleep(pdata->speaker_amp_gpio, + !!SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static int dfl61twl_tlv320dac33_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int r; + + if (SND_SOC_DAPM_EVENT_ON(event)) + r = dfl61dac33_interconnect_enable(1); + else + r = dfl61dac33_interconnect_enable(0); + + return r; +} + +static int dfl61twl_hsmic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (!hsmic_event || !hsmic_event->event) + return 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + hsmic_event->event(hsmic_event->private, 1); + else + hsmic_event->event(hsmic_event->private, 0); + + return 0; +} + +/* DAPM widgets and routing for TWL4030 */ +static const struct snd_soc_dapm_widget dfl61twl_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", dfl61twl_spk_event), + SND_SOC_DAPM_SPK("Earpiece", NULL), + SND_SOC_DAPM_SPK("HAC", NULL), + SND_SOC_DAPM_SPK("Vibra", NULL), + SND_SOC_DAPM_SPK("DAC33 interconnect", dfl61twl_tlv320dac33_event), + + SND_SOC_DAPM_MIC("Digital Mic", NULL), + SND_SOC_DAPM_MIC("Headset Mic", dfl61twl_hsmic_event), + + SND_SOC_DAPM_LINE("FMRX Left Line-in", NULL), + SND_SOC_DAPM_LINE("FMRX Right Line-in", NULL), +}; + +static const struct snd_soc_dapm_route dfl61twl_audio_map[] = { + {"Ext Spk", NULL, "PREDRIVER"}, + {"Earpiece", NULL, "EARPIECE"}, + {"HAC", NULL, "HFL"}, + {"Vibra", NULL, "VIBRA"}, + {"DAC33 interconnect", NULL, "PREDRIVEL"}, + + {"DIGIMIC0", NULL, "Digital Mic"}, + {"Digital Mic", NULL, "Mic Bias 1"}, + + {"HSMIC", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "Headset Mic Bias"}, + + {"AUXL", NULL, "FMRX Left Line-in"}, + {"AUXR", NULL, "FMRX Right Line-in"}, +}; + +/* Pre DAC routings for the twl4030 codec */ +static const char *twl4030_predacl1_texts[] = { + "SDRL1", "SDRM1", "SDRL2", "SDRM2", +}; +static const char *twl4030_predacr1_texts[] = { + "SDRR1", "SDRM1", "SDRR2", "SDRM2" +}; +static const char *twl4030_predacl2_texts[] = {"SDRL2", "SDRM2"}; +static const char *twl4030_predacr2_texts[] = {"SDRR2", "SDRM2"}; + +static const struct soc_enum twl4030_predacl1_enum = + SOC_ENUM_SINGLE(TWL4030_REG_RX_PATH_SEL, 2, + ARRAY_SIZE(twl4030_predacl1_texts), + twl4030_predacl1_texts); + +static const struct soc_enum twl4030_predacr1_enum = + SOC_ENUM_SINGLE(TWL4030_REG_RX_PATH_SEL, 0, + ARRAY_SIZE(twl4030_predacr1_texts), + twl4030_predacr1_texts); + +static const struct soc_enum twl4030_predacl2_enum = + SOC_ENUM_SINGLE(TWL4030_REG_RX_PATH_SEL, 5, + ARRAY_SIZE(twl4030_predacl2_texts), + twl4030_predacl2_texts); + +static const struct soc_enum twl4030_predacr2_enum = + SOC_ENUM_SINGLE(TWL4030_REG_RX_PATH_SEL, 4, + ARRAY_SIZE(twl4030_predacr2_texts), + twl4030_predacr2_texts); + +static const struct snd_kcontrol_new dfl61twl_controls[] = { + /* Mux controls before the DACs */ + SOC_ENUM("DACL1 Playback Mux", twl4030_predacl1_enum), + SOC_ENUM("DACR1 Playback Mux", twl4030_predacr1_enum), + SOC_ENUM("DACL2 Playback Mux", twl4030_predacl2_enum), + SOC_ENUM("DACR2 Playback Mux", twl4030_predacr2_enum), +}; + +static int dfl61twl_init(struct snd_soc_pcm_runtime *rtd) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int r; + + /* Create jack for accessory reporting */ + r = snd_soc_card_jack_new(&dfl61twl_sound_card, "Jack", + JACK_REPORT_MASK , &dfl61_jack, NULL, 0); + if (r) { + pr_err("Failed to add Jack\n"); + return r; + } + + snd_soc_add_codec_controls(rtd->codec, dfl61twl_controls, + ARRAY_SIZE(dfl61twl_controls)); + + if (omap_mcbsp_st_add_controls(rtd, 3)) + dev_dbg(rtd->codec->dev, "Unable to set Sidetone for McBSP3\n"); + + mcbsp->dma_op_mode = MCBSP_DMA_MODE_THRESHOLD; + + return 0; +} + +static struct snd_soc_ops dfl61twl_ops = { + .hw_params = dfl61twl_hw_params, +}; + +/* TLV320DAC33 */ +static int dfl61dac33_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int r; + + /* Set codec DAI configuration */ + r = snd_soc_dai_set_fmt(rtd->codec_dai, + SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (r < 0) { + pr_err("Can't set codec DAI configuration for tlv320dac33: %d\n", r); + return r; + } + + /* Set cpu DAI configuration */ + r = snd_soc_dai_set_fmt(rtd->cpu_dai, + SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (r < 0) { + pr_err("Can't set cpu DAI configuration for tlv320dac33: %d\n", r); + return r; + } + + /* Set the codec system clock for DAC and ADC */ + r = snd_soc_dai_set_sysclk(rtd->codec_dai, TLV320DAC33_SLEEPCLK, 32768, + SND_SOC_CLOCK_IN); + if (r < 0) { + pr_err("Can't set codec system clock for tlv320dac33\n"); + return r; + } + + return 0; +} + +static int dfl61dac33_interconnect_enable(int enable) +{ + struct snd_soc_component *component = + find_component(&dfl61dac33_sound_card); + struct snd_soc_dapm_context *dapm; + bool lock = false; + + if (!component) + return -ENODEV; + + dapm = snd_soc_component_get_dapm(component); + + mutex_lock(&dapm->card->dapm_mutex); + lock = true; + + if (enable) + snd_soc_dapm_enable_pin_unlocked(dapm, "twl4030 interconnect"); + else + snd_soc_dapm_disable_pin_unlocked(dapm, "twl4030 interconnect"); + + snd_soc_dapm_sync_unlocked(dapm); + + if (lock) + mutex_unlock(&dapm->card->dapm_mutex); + + return 0; +} + +static void dfl61dac33_hp_enable(struct snd_soc_component *component, int enable) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + + if (enable) { + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone"); + } else { + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone"); + } + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +int dfl61_request_hp_enable(bool enable) +{ + struct snd_soc_component *component = + find_component(&dfl61dac33_sound_card); + + if (!component) { + pr_err("dfl61-request_hp_enable"); + return -ENODEV; + } + + dfl61dac33_hp_enable(component, enable); + + return 0; +} +EXPORT_SYMBOL(dfl61_request_hp_enable); + +static const struct snd_kcontrol_new dfl61dac33_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), +}; + +static const struct snd_soc_dapm_widget dfl61dac33_dapm_widgets[] = { + /* Outputs */ + SND_SOC_DAPM_LINE("FMTX_L Line Out", NULL), + SND_SOC_DAPM_LINE("FMTX_R Line Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + /* Inputs */ + SND_SOC_DAPM_LINE("twl4030 interconnect", NULL), +}; + +static const struct snd_soc_dapm_route dfl61dac33_audio_map[] = { + {"Headphone", NULL, "TPA6140A2 HPLEFT"}, + {"Headphone", NULL, "TPA6140A2 HPRIGHT"}, + {"TPA6140A2 HPLEFT", NULL, "LEFT_LO"}, + {"TPA6140A2 HPRIGHT", NULL, "RIGHT_LO"}, + + {"FMTX_L Line Out", NULL, "LEFT_LO"}, + {"FMTX_R Line Out", NULL, "RIGHT_LO"}, + + {"LINER", NULL, "twl4030 interconnect"}, + {"LINEL", NULL, "twl4030 interconnect"}, +}; + +static int dfl61dac33_init(struct snd_soc_pcm_runtime *rtd) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(rtd->codec); + + snd_soc_limit_volume(rtd->card, "TPA6140A2 Headphone Playback Volume", 21); + + snd_soc_dapm_disable_pin(dapm, "twl4030 interconnect"); + + if (omap_mcbsp_st_add_controls(rtd, 2)) + dev_dbg(rtd->codec->dev, "Unable to set Sidetone for McBSP2\n"); + + mcbsp->dma_op_mode = MCBSP_DMA_MODE_THRESHOLD; + + return 0; +} + +static struct snd_soc_ops dfl61dac33_ops = { + .hw_params = dfl61dac33_hw_params, +}; + +/* WL1273 */ +static int dfl61wl1273_init(struct snd_soc_pcm_runtime *rtd) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(rtd->cpu_dai); + mcbsp->dma_op_mode = MCBSP_DMA_MODE_THRESHOLD; + + return 0; +} + +static int dfl61wl1273_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int fmt; + int r; + + r = wl1273_get_format(rtd->codec, &fmt); + if (r < 0) { + pr_err("Can't get fmt for wl1273: %d\n", r); + return r; + } + + /* Set cpu DAI configuration */ + r = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (r < 0) { + pr_err("Can't set cpu DAI configuration for wl1273: %d\n", r); + return r; + } + return 0; +} + +static struct snd_soc_ops dfl61wl1273_ops = { + .startup = dfl61wl1273_startup, +}; + +static int dfl61wl1273_card_probe(struct snd_soc_card *card) +{ + struct dfl61wl1273_audio_pdata *pdata = snd_soc_card_get_drvdata(card); + + gpiod_set_value(pdata->power_gpio, 1); + return 0; +} + + +static int dfl61wl1273_card_remove(struct snd_soc_card *card) +{ + struct dfl61wl1273_audio_pdata *pdata = snd_soc_card_get_drvdata(card); + + gpiod_set_value(pdata->power_gpio, 0); + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link dfl61twl_dai[] = { + { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai_name = "omap-mcbsp.3", + .codec_dai_name = "twl4030-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "twl4030-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .init = dfl61twl_init, + .ops = &dfl61twl_ops, + }, +}; + +static struct snd_soc_dai_link dfl61dac33_dai[] = { + { + .name = "TLV320DAC33", + .stream_name = "DAC33", + .cpu_dai_name = "omap-mcbsp.2", + .codec_dai_name = "tlv320dac33-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "tlv320dac33-codec.1-0019", + .init = dfl61dac33_init, + .ops = &dfl61dac33_ops, + }, +}; + +static struct snd_soc_aux_dev dfl61dac33_aux_dev[] = { + { + .name = "TPA6140A2", + .codec_name = "tpa6130a2.1-0060", + }, +}; + +static struct snd_soc_codec_conf dfl61dac33_codec_conf[] = { + { + .dev_name = "tpa6130a2.2-0060", + .name_prefix = "TPA6140A2", + }, +}; + +static struct snd_soc_dai_link dfl61wl1273_dai[] = { + { + .name = "BT/FM PCM", + .stream_name = "BT/FM Stream", + .cpu_dai_name = "omap-mcbsp.4", + .codec_dai_name = "wl1273-fm", + .platform_name = "omap-pcm-audio", + .codec_name = "wl1273-codec", + .init = dfl61wl1273_init, + .ops = &dfl61wl1273_ops, + }, +}; + +/* Audio cards */ +static struct snd_soc_card dfl61twl_sound_card = { + .name = "dfl61-twl4030", + .owner = THIS_MODULE, + .dai_link = dfl61twl_dai, + .num_links = ARRAY_SIZE(dfl61twl_dai), + .fully_routed = true, + .dapm_widgets = dfl61twl_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dfl61twl_dapm_widgets), + .dapm_routes = dfl61twl_audio_map, + .num_dapm_routes = ARRAY_SIZE(dfl61twl_audio_map), +}; + +static struct snd_soc_card dfl61dac33_sound_card = { + .name = "dfl61-dac33", + .owner = THIS_MODULE, + .dai_link = dfl61dac33_dai, + .num_links = ARRAY_SIZE(dfl61dac33_dai), + .aux_dev = dfl61dac33_aux_dev, + .num_aux_devs = ARRAY_SIZE(dfl61dac33_aux_dev), + .codec_conf = dfl61dac33_codec_conf, + .num_configs = ARRAY_SIZE(dfl61dac33_codec_conf), + .fully_routed = true, + .controls = dfl61dac33_controls, + .num_controls = ARRAY_SIZE(dfl61dac33_controls), + .dapm_widgets = dfl61dac33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dfl61dac33_dapm_widgets), + .dapm_routes = dfl61dac33_audio_map, + .num_dapm_routes = ARRAY_SIZE(dfl61dac33_audio_map), +}; + +static struct snd_soc_card dfl61wl1273_sound_card = { + .name = "dfl61-wl1273", + .owner = THIS_MODULE, + .probe = dfl61wl1273_card_probe, + .remove = dfl61wl1273_card_remove, + .dai_link = dfl61wl1273_dai, + .num_links = ARRAY_SIZE(dfl61wl1273_dai), + .fully_routed = true, +}; + +static int n9_soc_probe(struct platform_device *pdev) +{ + struct dfl61twl_audio_pdata *pdata_twl; + struct dfl61wl1273_audio_pdata *pdata_wl1273; + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card_twl = &dfl61twl_sound_card; + struct snd_soc_card *card_dac33 = &dfl61dac33_sound_card; + struct snd_soc_card *card_wl1273 = &dfl61wl1273_sound_card; + int err; + + if (!(machine_is_nokia_rm696() || machine_is_nokia_rm680()) + && !(of_machine_is_compatible("nokia,omap3-n9") + || of_machine_is_compatible("nokia,omap3-n950"))) + return -ENODEV; + + card_twl->dev = &pdev->dev; + card_dac33->dev = &pdev->dev; + card_wl1273->dev = &pdev->dev; + + if (np) { + struct device_node *dai_node; + + dai_node = of_parse_phandle(np, "nokia,twl4030-cpu-dai", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McBSP node for TWL4030 is not provided\n"); + return -EINVAL; + } + dfl61twl_dai[0].cpu_dai_name = NULL; + dfl61twl_dai[0].platform_name = NULL; + dfl61twl_dai[0].cpu_of_node = dai_node; + dfl61twl_dai[0].platform_of_node = dai_node; + + dai_node = of_parse_phandle(np, "nokia,tlv320dac33-cpu-dai", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McBSP node for TLV320DAC33 is not provided\n"); + return -EINVAL; + } + dfl61dac33_dai[0].cpu_dai_name = NULL; + dfl61dac33_dai[0].platform_name = NULL; + dfl61dac33_dai[0].cpu_of_node = dai_node; + dfl61dac33_dai[0].platform_of_node = dai_node; + + dai_node = of_parse_phandle(np, "nokia,wl1273-cpu-dai", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McBSP node for WL1273 is not provided\n"); + return -EINVAL; + } + dfl61wl1273_dai[0].cpu_dai_name = NULL; + dfl61wl1273_dai[0].platform_name = NULL; + dfl61wl1273_dai[0].cpu_of_node = dai_node; + dfl61wl1273_dai[0].platform_of_node = dai_node; + + dai_node = of_parse_phandle(np, "nokia,twl4030-codec", 0); + if (!dai_node) { + dev_err(&pdev->dev, "TWL4030 codec node is not provided\n"); + return -EINVAL; + } + dfl61twl_dai[0].codec_name = NULL; + dfl61twl_dai[0].codec_of_node = dai_node; + + dai_node = of_parse_phandle(np, "nokia,tlv320dac33-codec", 0); + if (!dai_node) { + dev_err(&pdev->dev, "TLV320DAC33 codec node is not provided\n"); + return -EINVAL; + } + dfl61dac33_dai[0].codec_name = NULL; + dfl61dac33_dai[0].codec_of_node = dai_node; + + dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0); + if (!dai_node) { + dev_err(&pdev->dev, "Headphone amplifier node is not provided\n"); + return -EINVAL; + } + dfl61dac33_aux_dev[0].codec_name = NULL; + dfl61dac33_aux_dev[0].codec_of_node = dai_node; + dfl61dac33_codec_conf[0].dev_name = NULL; + dfl61dac33_codec_conf[0].of_node = dai_node; + + dai_node = of_parse_phandle(np, "nokia,wl1273-codec", 0); + if (!dai_node) { + dev_err(&pdev->dev, "WL1273 codec node is not provided\n"); + return -EINVAL; + } + dfl61wl1273_dai[0].codec_name = NULL; + dfl61wl1273_dai[0].codec_of_node = dai_node; + } + + pdata_twl = devm_kzalloc(&pdev->dev, sizeof(*pdata_twl), GFP_KERNEL); + if (pdata_twl == NULL) { + dev_err(card_twl->dev, "failed to create private data for twl4030\n"); + return -ENOMEM; + } + snd_soc_card_set_drvdata(card_twl, pdata_twl); + + pdata_twl->speaker_amp_gpio = devm_gpiod_get(card_twl->dev, + "speaker-amplifier", + GPIOD_OUT_LOW); + if (IS_ERR(pdata_twl->speaker_amp_gpio)) { + dev_err(card_twl->dev, "could not get speaker enable gpio\n"); + return PTR_ERR(pdata_twl->speaker_amp_gpio); + } + + pdata_wl1273 = devm_kzalloc(&pdev->dev, sizeof(*pdata_wl1273), GFP_KERNEL); + if (pdata_wl1273 == NULL) { + dev_err(card_wl1273->dev, "failed to create private data for wl1273\n"); + return -ENOMEM; + } + snd_soc_card_set_drvdata(card_wl1273, pdata_wl1273); + + pdata_wl1273->power_gpio = devm_gpiod_get(card_wl1273->dev, + "wl1273-power", + GPIOD_OUT_LOW); + if (IS_ERR(pdata_wl1273->power_gpio)) { + dev_err(card_wl1273->dev, "could not get wl1273 enable gpio\n"); + return PTR_ERR(pdata_wl1273->power_gpio); + } + + err = devm_snd_soc_register_card(&pdev->dev, card_twl); + if (err < 0) { + dev_err(card_twl->dev, "failed to register twl4030 card: %d\n", err); + return err; + } + + err = devm_snd_soc_register_card(&pdev->dev, card_dac33); + if (err < 0) { + dev_err(card_dac33->dev, "failed to register tlv320dac33 card: %d\n", err); + return err; + } + + + + err = devm_snd_soc_register_card(&pdev->dev, card_wl1273); + if (err < 0) { + dev_err(card_wl1273->dev, "failed to register wl1273 card\n"); + return err; + } + + return 0; +} + +static const struct of_device_id n9_audio_of_match[] = { + { .compatible = "nokia,n9-audio", }, + {}, +}; +MODULE_DEVICE_TABLE(of, n9_audio_of_match); + +static struct platform_driver n9_soc_driver = { + .driver = { + .name = "n9-audio", + .of_match_table = of_match_ptr(n9_audio_of_match), + }, + .probe = n9_soc_probe, +}; + +module_platform_driver(n9_soc_driver); + +MODULE_AUTHOR("Nokia Corporation"); +MODULE_DESCRIPTION("ALSA SoC Nokia N9/N950"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:n9-audio"); -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [PATCH] ASoC: rt5645: add fallback case for jack detection support
by Pierre-Louis Bossart 12 Jan '18

12 Jan '18

Commit 78f5605c0329 ("ASoC: rt5645: cleanup DMI matching code") did a lot of useful cleanups, but the logic is a bit flawed: if there is no DMI match on an ACPI machine, the code reads non-existent properties. Re-add previous code that checked for the existence of those properties and a fallback mode to enable jd-mode3 if there is no pdata, device property or quirk. This will help avoid having to add new DMI quirks when only the jd-mode is needed for headset playback. Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt5645.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index ebc4f5970f24..476020445f65 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3687,6 +3687,17 @@ static const struct dmi_system_id dmi_platform_data[] = { { } }; +static bool rt5645_check_dp(struct device *dev) +{ + if (device_property_present(dev, "realtek,in2-differential") || + device_property_present(dev, "realtek,dmic1-data-pin") || + device_property_present(dev, "realtek,dmic2-data-pin") || + device_property_present(dev, "realtek,jd-mode")) + return true; + + return false; +} + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3727,8 +3738,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else + else if (rt5645_check_dp(&i2c->dev)) rt5645_parse_dt(rt5645, &i2c->dev); + else + rt5645->pdata = jd_mode3_platform_data; if (quirk != -1) { rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); -- 2.14.1

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[alsa-devel] [PATCH v3 00/27] kill devm_ioremap_nocache
by Yisheng Xie 12 Jan '18

12 Jan '18

Hi all, When I tried to use devm_ioremap function and review related code, I found devm_ioremap and devm_ioremap_nocache is almost the same with each other, except one use ioremap while the other use ioremap_nocache. While ioremap's default function is ioremap_nocache, so devm_ioremap_nocache also have the same function with devm_ioremap, which can just be killed to reduce the size of devres.o(from 20304 bytes to 18992 bytes in my compile environment). I have posted two versions, which use macro instead of function for devm_ioremap_nocache[1] or devm_ioremap[2]. And Greg suggest me to kill devm_ioremap_nocache for no need to keep a macro around for the duplicate thing. So here comes v3 and please help to review. Thanks so much! Yisheng Xie [1] https://lkml.org/lkml/2017/11/20/135 [2] https://lkml.org/lkml/2017/11/25/21 Yisheng Xie (27): ASOC: replace devm_ioremap_nocache with devm_ioremap spi: replace devm_ioremap_nocache with devm_ioremap staging: replace devm_ioremap_nocache with devm_ioremap ipack: replace devm_ioremap_nocache with devm_ioremap media: replace devm_ioremap_nocache with devm_ioremap gpio: replace devm_ioremap_nocache with devm_ioremap mmc: replace devm_ioremap_nocache with devm_ioremap PCI: replace devm_ioremap_nocache with devm_ioremap platform/x86: replace devm_ioremap_nocache with devm_ioremap tty: replace devm_ioremap_nocache with devm_ioremap video: replace devm_ioremap_nocache with devm_ioremap rtc: replace devm_ioremap_nocache with devm_ioremap char: replace devm_ioremap_nocache with devm_ioremap mtd: nand: replace devm_ioremap_nocache with devm_ioremap dmaengine: replace devm_ioremap_nocache with devm_ioremap ata: replace devm_ioremap_nocache with devm_ioremap irqchip: replace devm_ioremap_nocache with devm_ioremap pinctrl: replace devm_ioremap_nocache with devm_ioremap drm: replace devm_ioremap_nocache with devm_ioremap regulator: replace devm_ioremap_nocache with devm_ioremap watchdog: replace devm_ioremap_nocache with devm_ioremap tools/testing/nvdimm: replace devm_ioremap_nocache with devm_ioremap MIPS: pci: replace devm_ioremap_nocache with devm_ioremap can: replace devm_ioremap_nocache with devm_ioremap wireless: replace devm_ioremap_nocache with devm_ioremap ethernet: replace devm_ioremap_nocache with devm_ioremap devres: kill devm_ioremap_nocache Documentation/driver-model/devres.txt | 1 - arch/mips/pci/pci-ar2315.c | 3 +-- drivers/ata/pata_arasan_cf.c | 3 +-- drivers/ata/pata_octeon_cf.c | 9 ++++---- drivers/ata/pata_rb532_cf.c | 2 +- drivers/char/hw_random/bcm63xx-rng.c | 3 +-- drivers/char/hw_random/octeon-rng.c | 10 ++++----- drivers/dma/altera-msgdma.c | 3 +-- drivers/dma/sprd-dma.c | 4 ++-- drivers/gpio/gpio-ath79.c | 3 +-- drivers/gpio/gpio-em.c | 6 ++--- drivers/gpio/gpio-htc-egpio.c | 4 ++-- drivers/gpio/gpio-xgene.c | 3 +-- drivers/gpu/drm/hisilicon/hibmc/hibmc_drm_drv.c | 2 +- drivers/gpu/drm/msm/msm_drv.c | 2 +- drivers/gpu/drm/sti/sti_dvo.c | 3 +-- drivers/gpu/drm/sti/sti_hda.c | 4 ++-- drivers/gpu/drm/sti/sti_hdmi.c | 2 +- drivers/gpu/drm/sti/sti_tvout.c | 2 +- drivers/gpu/drm/sti/sti_vtg.c | 2 +- drivers/ipack/devices/ipoctal.c | 13 +++++------ drivers/irqchip/irq-renesas-intc-irqpin.c | 4 ++-- drivers/media/platform/tegra-cec/tegra_cec.c | 4 ++-- drivers/mmc/host/sdhci-acpi.c | 3 +-- drivers/mtd/nand/fsl_upm.c | 4 ++-- drivers/net/can/sja1000/sja1000_platform.c | 4 ++-- drivers/net/ethernet/altera/altera_tse_main.c | 3 +-- drivers/net/ethernet/ethoc.c | 8 +++---- drivers/net/ethernet/lantiq_etop.c | 4 ++-- drivers/net/ethernet/ti/netcp_core.c | 2 +- drivers/net/wireless/ath/ath9k/ahb.c | 2 +- drivers/pci/dwc/pci-dra7xx.c | 2 +- drivers/pinctrl/bcm/pinctrl-ns2-mux.c | 2 +- drivers/pinctrl/bcm/pinctrl-nsp-mux.c | 4 ++-- drivers/pinctrl/freescale/pinctrl-imx1-core.c | 2 +- drivers/pinctrl/pinctrl-amd.c | 4 ++-- drivers/platform/x86/intel_pmc_core.c | 5 ++--- drivers/regulator/ti-abb-regulator.c | 6 ++--- drivers/rtc/rtc-sh.c | 4 ++-- drivers/spi/spi-jcore.c | 3 +-- drivers/staging/fsl-mc/bus/mc-io.c | 8 +++---- drivers/tty/mips_ejtag_fdc.c | 4 ++-- drivers/tty/serial/8250/8250_omap.c | 3 +-- drivers/tty/serial/lantiq.c | 3 +-- drivers/tty/serial/meson_uart.c | 3 +-- drivers/tty/serial/owl-uart.c | 2 +- drivers/tty/serial/pic32_uart.c | 4 ++-- drivers/video/fbdev/mbx/mbxfb.c | 9 ++++---- drivers/video/fbdev/mmp/hw/mmp_ctrl.c | 2 +- drivers/video/fbdev/pxa168fb.c | 4 ++-- drivers/watchdog/bcm63xx_wdt.c | 4 ++-- drivers/watchdog/rc32434_wdt.c | 4 ++-- include/linux/io.h | 2 -- lib/devres.c | 29 ------------------------- scripts/coccinelle/free/devm_free.cocci | 2 -- sound/soc/au1x/ac97c.c | 4 ++-- sound/soc/au1x/i2sc.c | 4 ++-- sound/soc/intel/atom/sst/sst_acpi.c | 20 ++++++++--------- sound/soc/intel/boards/mfld_machine.c | 4 ++-- sound/soc/sh/fsi.c | 4 ++-- tools/testing/nvdimm/Kbuild | 2 +- tools/testing/nvdimm/test/iomap.c | 2 +- 62 files changed, 109 insertions(+), 168 deletions(-) -- 1.8.3.1

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[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: Ensuring proper setting of output widget power state
by abhijeet.kumar＠intel.com 12 Jan '18

12 Jan '18

From: Abhijeet Kumar <abhijeet.kumar(a)intel.com> In usecases like hot plug-unplug DP panel or modeset during a playback, sometimes we observe no audio after codec resets. During no audio condition, we have noticed that the power state of the pin or the connector is D3. Optimzing the way we set the power could mitigate the issue. With this changes the verb is sent to set the power state and response is received. Thus ensuring power state is set. Signed-off-by: Abhijeet Kumar <abhijeet.kumar(a)intel.com> --- Changes in v2: - update commit message sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f3b4f4dfae6a..e24caecf0a4f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -718,7 +718,7 @@ static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, { if (get_wcaps(&edev->hdac, nid) & AC_WCAP_POWER) { if (!snd_hdac_check_power_state(&edev->hdac, nid, pwr_state)) - snd_hdac_codec_write(&edev->hdac, nid, 0, + snd_hdac_codec_read(&edev->hdac, nid, 0, AC_VERB_SET_POWER_STATE, pwr_state); } } -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: rt5645: add fallback case for jack detection support
by Pierre-Louis Bossart 11 Jan '18

11 Jan '18

Commit 78f5605c0329 ("ASoC: rt5645: cleanup DMI matching code") did a lot of useful cleanups. This patch adds a default case to enable jack detection if there is no pdata, device property or quirk. The chosen jd-mode3 is the most common and should limit the addition of new DMI-based quirks. Existing DMI quirks which only set this mode are left as is and not removed. Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: no functional change, only rewrite of the commit message for clarity sound/soc/codecs/rt5645.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 789346cb30b9..8f140c8b93ac 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3739,6 +3739,17 @@ static const struct dmi_system_id dmi_platform_data[] = { { } }; +static bool rt5645_check_dp(struct device *dev) +{ + if (device_property_present(dev, "realtek,in2-differential") || + device_property_present(dev, "realtek,dmic1-data-pin") || + device_property_present(dev, "realtek,dmic2-data-pin") || + device_property_present(dev, "realtek,jd-mode")) + return true; + + return false; +} + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3779,8 +3790,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else + else if (rt5645_check_dp(&i2c->dev)) rt5645_parse_dt(rt5645, &i2c->dev); + else + rt5645->pdata = jd_mode3_platform_data; if (quirk != -1) { rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); -- 2.14.1

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[alsa-devel] [PATCH] ASoC: sgtl5000: fix powerup/down of micbias-voltage
by fischerc＠swissphone.de 11 Jan '18

11 Jan '18

From: Christian Fischer <fischerc(a)swissphone.de> The mic-bias-voltage should get powered up/down automatically. Added missing DAPM_PRE and DAPM_POST widgets to fire up the mic_bias_event function. Signed-off-by: Christian Fischer <fischerc(a)swissphone.de> --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f2bb4fe..4973365 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -260,6 +260,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), + + SND_SOC_DAPM_PRE("MIC_BIAS_PRE", mic_bias_event), + SND_SOC_DAPM_POST("MIC_BIAS_POST", mic_bias_event), }; /* routes for sgtl5000 */ -- 2.7.4

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[alsa-devel] [PATCH 1/2] sound: Remove unused register_sound_midi() and co
by Takashi Iwai 11 Jan '18

11 Jan '18

These helpers are no longer used after the removal of the legacy OSS drivers. Let's clean up. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/linux/sound.h | 2 -- sound/sound_core.c | 35 ----------------------------------- 2 files changed, 37 deletions(-) diff --git a/include/linux/sound.h b/include/linux/sound.h index 3c6d393c7f29..ec85b7a1f8d1 100644 --- a/include/linux/sound.h +++ b/include/linux/sound.h @@ -12,11 +12,9 @@ struct device; extern int register_sound_special(const struct file_operations *fops, int unit); extern int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev); extern int register_sound_mixer(const struct file_operations *fops, int dev); -extern int register_sound_midi(const struct file_operations *fops, int dev); extern int register_sound_dsp(const struct file_operations *fops, int dev); extern void unregister_sound_special(int unit); extern void unregister_sound_mixer(int unit); -extern void unregister_sound_midi(int unit); extern void unregister_sound_dsp(int unit); #endif /* _LINUX_SOUND_H */ diff --git a/sound/sound_core.c b/sound/sound_core.c index 99b73c675743..8b026b66cf18 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -452,26 +452,6 @@ int register_sound_mixer(const struct file_operations *fops, int dev) EXPORT_SYMBOL(register_sound_mixer); -/** - * register_sound_midi - register a midi device - * @fops: File operations for the driver - * @dev: Unit number to allocate - * - * Allocate a midi device. Unit is the number of the midi device requested. - * Pass -1 to request the next free midi unit. - * - * Return: On success, the allocated number is returned. On failure, - * a negative error code is returned. - */ - -int register_sound_midi(const struct file_operations *fops, int dev) -{ - return sound_insert_unit(&chains[2], fops, dev, 2, 130, - "midi", S_IRUSR | S_IWUSR, NULL); -} - -EXPORT_SYMBOL(register_sound_midi); - /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. @@ -532,21 +512,6 @@ void unregister_sound_mixer(int unit) EXPORT_SYMBOL(unregister_sound_mixer); -/** - * unregister_sound_midi - unregister a midi device - * @unit: unit number to allocate - * - * Release a sound device that was allocated with register_sound_midi(). - * The unit passed is the return value from the register function. - */ - -void unregister_sound_midi(int unit) -{ - sound_remove_unit(&chains[2], unit); -} - -EXPORT_SYMBOL(unregister_sound_midi); - /** * unregister_sound_dsp - unregister a DSP device * @unit: unit number to allocate -- 2.15.1

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[alsa-devel] [PATCH] ALSA: seq: Make ioctls race-free
by Takashi Iwai 11 Jan '18

11 Jan '18

The ALSA sequencer ioctls have no protection against racy calls while the concurrent operations may lead to interfere with each other. As reported recently, for example, the concurrent calls of setting client pool with a combination of write calls may lead to either the unkillable dead-lock or UAF. As a slightly big hammer solution, this patch introduces the mutex to make each ioctl exclusive. Although this may reduce performance via parallel ioctl calls, usually it's not demanded for sequencer usages, hence it should be negligible. Reported-by: Luo Quan <a4651386(a)163.com> Reviewed-by: Kees Cook <keescook(a)chromium.org> Reviewed-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 3 +++ sound/core/seq/seq_clientmgr.h | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6e22eea72654..d01913404581 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -221,6 +221,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) rwlock_init(&client->ports_lock); mutex_init(&client->ports_mutex); INIT_LIST_HEAD(&client->ports_list_head); + mutex_init(&client->ioctl_mutex); /* find free slot in the client table */ spin_lock_irqsave(&clients_lock, flags); @@ -2130,7 +2131,9 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, return -EFAULT; } + mutex_lock(&client->ioctl_mutex); err = handler->func(client, &buf); + mutex_unlock(&client->ioctl_mutex); if (err >= 0) { /* Some commands includes a bug in 'dir' field. */ if (handler->cmd == SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT || diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index c6614254ef8a..0611e1e0ed5b 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -61,6 +61,7 @@ struct snd_seq_client { struct list_head ports_list_head; rwlock_t ports_lock; struct mutex ports_mutex; + struct mutex ioctl_mutex; int convert32; /* convert 32->64bit */ /* output pool */ -- 2.15.1

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[alsa-devel] [PATCH -next] ASoC: stm32: fix a typo in stm32_adfsdm_probe()
by Wei Yongjun 11 Jan '18

11 Jan '18

Fix a typo, we should return PTR_ERR(priv->iio_cb) instead of PTR_ERR(priv->iio_ch). Fixes: 55da094824c4 ("ASoC: stm32: add DFSDM DAI support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index af50891..7306e3e 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -320,7 +320,7 @@ static int stm32_adfsdm_probe(struct platform_device *pdev) priv->iio_cb = iio_channel_get_all_cb(&pdev->dev, NULL, NULL); if (IS_ERR(priv->iio_cb)) - return PTR_ERR(priv->iio_ch); + return PTR_ERR(priv->iio_cb); ret = devm_snd_soc_register_platform(&pdev->dev, &stm32_adfsdm_soc_platform);

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