September 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ES938 support for ES18xx driver
by Ondrej Zary 29 Sep '14

29 Sep '14

Add support for ES938 3-D Audio Effects Processor found on some ES18xx (and possibly other) sound cards, doing bass/treble and 3D control. ES938 is controlled by MIDI SysEx commands sent through card's MPU401 port. The code opens/closes the MIDI device everytime a mixer control value is changed so userspace apps can still use the MIDI port. Changing the mixer controls will fail when the MIDI port is open by an application. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- Changes in v2: - debug message when ES938 detected - reworked sysex messages to use structs - ktime instead of jiffies for timeout --- sound/isa/Kconfig | 4 + sound/isa/Makefile | 2 + sound/isa/es18xx.c | 13 +++- sound/isa/es938.c | 215 ++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/es938.h | 48 ++++++++++++ 5 files changed, 281 insertions(+), 1 deletion(-) create mode 100644 sound/isa/es938.c create mode 100644 sound/isa/es938.h diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..db0b678 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -17,6 +17,9 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON +config SND_ES938 + tristate + menuconfig SND_ISA bool "ISA sound devices" depends on ISA && ISA_DMA_API @@ -183,6 +186,7 @@ config SND_ES18XX select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_ES938 help Say Y here to include support for ESS AudioDrive ES18xx chips. diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..d59e0bf 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -8,6 +8,7 @@ snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o +snd-es938-objs := es938.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,6 +20,7 @@ obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o +obj-$(CONFIG_SND_ES938) += snd-es938.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..e174e8c 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -96,6 +96,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> +#include "es938.h" #define PFX "es18xx: " @@ -122,6 +123,7 @@ struct snd_es18xx { struct snd_pcm_substream *playback_b_substream; struct snd_rawmidi *rmidi; + struct snd_es938 es938; struct snd_kcontrol *hw_volume; struct snd_kcontrol *hw_switch; @@ -2167,7 +2169,16 @@ static int snd_audiodrive_probe(struct snd_card *card, int dev) return err; } - return snd_card_register(card); + err = snd_card_register(card); + if (err < 0) + return err; + + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) + /* no error if this fails because ES938 is optional */ + if (snd_es938_init(&chip->es938, card, 0, 0) == 0) + snd_printk(KERN_DEBUG "found ES938 audio processor\n"); + + return 0; } static int snd_es18xx_isa_match(struct device *pdev, unsigned int dev) diff --git a/sound/isa/es938.c b/sound/isa/es938.c new file mode 100644 index 0000000..07f6ccb --- /dev/null +++ b/sound/isa/es938.c @@ -0,0 +1,215 @@ +/* + * Driver for ESS ES938 3-D Audio Effects Processor + * Copyright (c) 2014 Ondrej Zary + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/module.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/rawmidi.h> +#include "es938.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("ESS ES938"); +MODULE_LICENSE("GPL"); + +static int snd_es938_read_reg(struct snd_es938 *chip, u8 reg, u8 *out) +{ + int i = 0, res; + struct snd_es938_sysex_reg req = { + .midi_cmd = MIDI_CMD_COMMON_SYSEX, + .id = ES938_ID, + .cmd = ES938_CMD_REG_R, + .reg = reg, + .midi_end = MIDI_CMD_COMMON_SYSEX_END, + }; + struct snd_es938_sysex_regval reply = { }; + u8 *p = (void *)&reply; + ktime_t end_time; + + snd_rawmidi_kernel_write(chip->rfile.output, (void *)&req, sizeof(req)); + + end_time = ktime_add_ms(ktime_get(), 100); + while (i < sizeof(reply)) { + res = snd_rawmidi_kernel_read(chip->rfile.input, p + i, + sizeof(reply) - i); + if (res > 0) + i += res; + if (ktime_after(ktime_get(), end_time)) + return -1; + } + + /* check if the reply is our and has SYSEX_END at the end */ + if (memcmp(&reply, &req, sizeof(req) - 1) || + reply.midi_end != MIDI_CMD_COMMON_SYSEX_END) + return -1; + + if (out) + *out = reply.val; + + return 0; +} + +static void snd_es938_write_reg(struct snd_es938 *chip, u8 reg, u8 val) +{ + struct snd_es938_sysex_regval req = { + .midi_cmd = MIDI_CMD_COMMON_SYSEX, + .id = ES938_ID, + .cmd = ES938_CMD_REG_W, + .reg = reg, + .val = val, + .midi_end = MIDI_CMD_COMMON_SYSEX_END, + }; + + snd_rawmidi_kernel_write(chip->rfile.output, (void *)&req, sizeof(req)); + chip->regs[reg] = val; +} + +#define ES938_MIXER(xname, xindex, reg, shift, mask) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ +.info = snd_es938_info_mixer, \ +.get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ +.private_value = reg | (shift << 8) | (mask << 16) } + +#define ES938_MIXER_TLV(xname, xindex, reg, shift, mask, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ +.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ +.info = snd_es938_info_mixer, \ +.get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ +.private_value = reg | (shift << 8) | (mask << 16), \ +.tlv = { .p = (xtlv) } } + +static const DECLARE_TLV_DB_SCALE(db_scale_tone, -900, 300, 0); + +static struct snd_kcontrol_new snd_es938_controls[] = { +ES938_MIXER_TLV("Tone Control - Bass", 0, ES938_REG_TONE, 0, 7, db_scale_tone), +ES938_MIXER_TLV("Tone Control - Treble", 0, ES938_REG_TONE, 4, 7, db_scale_tone), +ES938_MIXER("3D Control - Level", 0, ES938_REG_SPATIAL, 1, 63), +ES938_MIXER("3D Control - Switch", 0, ES938_REG_SPATIAL_EN, 0, 1), +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice) +{ + int ret, i; + + ret = snd_rawmidi_kernel_open(card, device, subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile); + if (ret < 0) { + snd_printk(KERN_WARNING "es938: unable to open MIDI device\n"); + return ret; + } + + /* try to read a register to detect chip presence */ + if (snd_es938_read_reg(chip, ES938_REG_MISC, NULL) < 0) { + ret = -ENODEV; + goto err; + } + + /* write default values (there's no reset) */ + snd_es938_write_reg(chip, ES938_REG_MISC, 0x49); + snd_es938_write_reg(chip, ES938_REG_TONE, 0x33); + /* reserved bit 0 must be set for 3D to work */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL, 0x01); + /* datasheet specifies invalid value 0x00 as default */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL_EN, 0x02); + snd_es938_write_reg(chip, ES938_REG_POWER, 0x0e); + + strlcat(card->mixername, " + ES938", sizeof(card->mixername)); + for (i = 0; i < ARRAY_SIZE(snd_es938_controls); i++) { + ret = snd_ctl_add(card, + snd_ctl_new1(&snd_es938_controls[i], chip)); + if (ret < 0) + goto err; + } + + chip->card = card; + chip->device = device; + chip->subdevice = subdevice; +err: + snd_rawmidi_kernel_release(&chip->rfile); + + return ret; +} +EXPORT_SYMBOL(snd_es938_init); + +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_info_mixer); + +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = chip->regs[reg]; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_get_mixer); + +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = ucontrol->value.integer.value[0] & mask; + u8 oldval = chip->regs[reg]; + u8 newval; + int err; + + mask <<= shift; + val <<= shift; + + newval = (oldval & ~mask) | val; + if (newval == oldval) + return 0; + + /* this will fail if the MIDI port is used by an application */ + err = snd_rawmidi_kernel_open(chip->card, chip->device, + chip->subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile); + if (err < 0) + return err; + + snd_es938_write_reg(chip, reg, newval); + + snd_rawmidi_kernel_release(&chip->rfile); + + return 1; +} +EXPORT_SYMBOL(snd_es938_put_mixer); diff --git a/sound/isa/es938.h b/sound/isa/es938.h new file mode 100644 index 0000000..08148ff --- /dev/null +++ b/sound/isa/es938.h @@ -0,0 +1,48 @@ +#include <sound/tlv.h> + +#define ES938_ID 0x7b + +#define ES938_CMD_REG_R 0x7e +#define ES938_CMD_REG_W 0x7f + +#define ES938_REG_MISC 0 +#define ES938_REG_TONE 1 +#define ES938_REG_SPATIAL 5 +#define ES938_REG_SPATIAL_EN 6 +#define ES938_REG_POWER 7 + +struct snd_es938 { + u8 regs[8]; + struct snd_card *card; + int device; + int subdevice; + struct snd_rawmidi_file rfile; +}; + +struct snd_es938_sysex_reg { + u8 midi_cmd; + u8 zeros[2]; + u8 id; + u8 cmd; + u8 reg; + u8 midi_end; +}; + +struct snd_es938_sysex_regval { + u8 midi_cmd; + u8 zeros[2]; + u8 id; + u8 cmd; + u8 reg; + u8 val; + u8 midi_end; +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice); +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); -- Ondrej Zary

2 1

[alsa-devel] [PATCH v2 2/3] ASoC: ssm2602: add device tree bindings
by Stefan Kristiansson 29 Sep '14

29 Sep '14

Allow the ssm2602/ssm2603/ssm2604 codec driver to be instantiated from the device tree. Also, add Kconfig prompts to allow manual selection of both the I2C and SPI configuration versions of the driver. Signed-off-by: Stefan Kristiansson <stefan.kristiansson(a)saunalahti.fi> --- Changes in v2: - Remove "ssm2603" and "ssm2604" from the spi compatible list --- .../devicetree/bindings/sound/adi,ssm2602.txt | 19 +++++++++++++++++++ sound/soc/codecs/Kconfig | 8 ++++++-- sound/soc/codecs/ssm2602-i2c.c | 9 +++++++++ sound/soc/codecs/ssm2602-spi.c | 7 +++++++ 4 files changed, 41 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2602.txt diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt new file mode 100644 index 0000000..3b3302f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt @@ -0,0 +1,19 @@ +Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices + +SSM2602 support both I2C and SPI as the configuration interface, +the selection is made by the MODE strap-in pin. +SSM2603 and SSM2604 only support I2C as the configuration interface. + +Required properties: + + - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + Example: + + ssm2602: ssm2602@1a { + compatible = "adi,ssm2602"; + reg = <0x1a>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..3649e73 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -520,12 +520,16 @@ config SND_SOC_SSM2602 tristate config SND_SOC_SSM2602_SPI + tristate "Analog Devices SSM2602 CODEC - SPI" + depends on SPI_MASTER select SND_SOC_SSM2602 - tristate + select REGMAP_SPI config SND_SOC_SSM2602_I2C + tristate "Analog Devices SSM2602 CODEC - I2C" + depends on I2C select SND_SOC_SSM2602 - tristate + select REGMAP_I2C config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c index abd63d5..0d9779d 100644 --- a/sound/soc/codecs/ssm2602-i2c.c +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -41,10 +41,19 @@ static const struct i2c_device_id ssm2602_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { .compatible = "adi,ssm2603", }, + { .compatible = "adi,ssm2604", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct i2c_driver ssm2602_i2c_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_i2c_probe, .remove = ssm2602_i2c_remove, diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c index 2bf55e2..b5df14f 100644 --- a/sound/soc/codecs/ssm2602-spi.c +++ b/sound/soc/codecs/ssm2602-spi.c @@ -26,10 +26,17 @@ static int ssm2602_spi_remove(struct spi_device *spi) return 0; } +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct spi_driver ssm2602_spi_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_spi_probe, .remove = ssm2602_spi_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH v2 3/3] ASoC: ssm2602: add support for 11.025kHz and 22.5kHz sample rates
by Stefan Kristiansson 29 Sep '14

29 Sep '14

This adds the necessary values to the constraint list and register values to the coefficient table in order to configure the device for 11.025kHz and 22.5kHz sample rates. Signed-off-by: Stefan Kristiansson <stefan.kristiansson(a)saunalahti.fi> --- Changes in v2: - Keep SSM2602_RATES sorted. --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 4021cd4..7c41848 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -192,7 +192,7 @@ static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { }; static const unsigned int ssm2602_rates_11289600[] = { - 8000, 44100, 88200, + 8000, 11025, 22050, 44100, 88200, }; static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { @@ -237,6 +237,16 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = { {18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)}, {12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)}, + /* 11.025k */ + {11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)}, + {16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)}, + {12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)}, + + /* 22.05k */ + {11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)}, + {16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)}, + {12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)}, + /* 44.1k */ {11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)}, {16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)}, @@ -467,7 +477,8 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Add dts properties for input/output differential configuration
by Anatol Pomozov 29 Sep '14

29 Sep '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 9 +++++++++ include/sound/rt5677.h | 5 ++++- sound/soc/codecs/rt5677.c | 23 ++++++++++++++++++++++ 3 files changed, 36 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index bd28df6..0701b83 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -19,6 +19,14 @@ Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. +- realtek,in1-differential +- realtek,in2-differential +- realtek,lout1-differential +- realtek,lout2-differential +- realtek,lout3-differential + Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential, + rather than single-ended. + Pins on the device (for linking into audio routes): * IN1P @@ -47,4 +55,5 @@ rt5677 { realtek,pow-ldo2-gpio = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; + realtek,in1-differential = "true"; }; diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h index a676717..082670e 100644 --- a/include/sound/rt5677.h +++ b/include/sound/rt5677.h @@ -19,9 +19,12 @@ enum rt5677_dmic2_clk { struct rt5677_platform_data { - /* IN1 IN2 can optionally be differential */ + /* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */ bool in1_diff; bool in2_diff; + bool lout1_diff; + bool lout2_diff; + bool lout3_diff; /* DMIC2 clock source selection */ enum rt5677_dmic2_clk dmic2_clk_pin; }; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index cecbdb1..16aa4d9 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3566,6 +3566,17 @@ MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) { + rt5677->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5677->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + rt5677->pdata.lout1_diff = of_property_read_bool(np, + "realtek,lout1-differential"); + rt5677->pdata.lout2_diff = of_property_read_bool(np, + "realtek,lout2-differential"); + rt5677->pdata.lout3_diff = of_property_read_bool(np, + "realtek,lout3-differential"); + rt5677->pow_ldo2 = of_get_named_gpio(np, "realtek,pow-ldo2-gpio", 0); @@ -3657,6 +3668,18 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); + if (rt5677->pdata.lout1_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT1_L_DF, RT5677_LOUT1_L_DF); + + if (rt5677->pdata.lout2_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT2_L_DF, RT5677_LOUT2_L_DF); + + if (rt5677->pdata.lout3_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT3_L_DF, RT5677_LOUT3_L_DF); + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2, RT5677_GPIO5_FUNC_MASK, -- 2.1.0.rc2.206.gedb03e5

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[alsa-devel] [PATCH 1/2] ASoC: Intel: byt-rt5640: Remove IN2N pin from DAPM route table
by Jarkko Nikula 29 Sep '14

29 Sep '14

I tested couple byt-rt5640 based platforms and they have single-ended headset microphone connection to IN2P only. I guess IN2N was either defined by accident or some early platform had floating ground for headset. It's better to remove IN2N and add a custom route for such a platform if needed. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-rt5640.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 234a58de3c53..d6d8b19c22dc 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -36,7 +36,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, - {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, -- 2.1.0

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[alsa-devel] [PATCH 1/3] ASoC: wm8{350, 753, 971}: Use snd_soc_dapm_to_codec() instead of dapm->codec
by Lars-Peter Clausen 29 Sep '14

29 Sep '14

The CODEC struct in the snd_soc_dapm_context struct is deprecated and scheduled for removal. Use the snd_soc_dapm_to_codec() function instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3dfdcc4..628ec77 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e54e097..21ca3a9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8753_set_bias_level(codec, dapm->bias_level); } diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4..39ddb9b 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8971_set_bias_level(codec, codec->dapm.bias_level); } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is
by Arnd Bergmann 29 Sep '14

29 Sep '14

It is possible to configure a kernel with SND_DAVINCI_SOC=m and SND_DM365_VOICE_CODEC=y, which results in a link error: sound/built-in.o: In function `davinci_vcif_probe': sound/soc/davinci/davinci-vcif.c:223: undefined reference to `davinci_soc_platform_register' The best way to avoid this is to make SND_DM365_VOICE_CODEC a tristate option that depends on SND_DAVINCI_SOC, so it can only be a module or disabled when the base driver is a loadable module Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index d69510c53239..8e948c63f3d9 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -63,7 +63,8 @@ config SND_DM365_AIC3X_CODEC Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC - bool "Voice Codec - CQ93VC" + tristate "Voice Codec - CQ93VC" + depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC

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[alsa-devel] [BUG] Bad decibel data on C-Media USB sound card
by Allen M. 29 Sep '14

29 Sep '14

Clemens Ladisch wrote: >What is the value of the "Headphone Playback Volume" mixer control >(as shown by "amixer contents") above and below this threshold? using "amixer --device hw:Set contents" 36% yields "values=8,8" 34% yields "values=0,0" interestingly, using "alsamixer" shows that while (default) reports 36%, the C-media device reports 3%

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[alsa-devel] [PATCH 1/2] ASoC: ssm2602: add device tree bindings
by Stefan Kristiansson 29 Sep '14

29 Sep '14

Allow the ssm2602/ssm2603/ssm2604 codec driver to be instantiated from the device tree. Also, add Kconfig prompts to allow manual selection of both the I2C and SPI configuration versions of the driver. Signed-off-by: Stefan Kristiansson <stefan.kristiansson(a)saunalahti.fi> --- .../devicetree/bindings/sound/adi,ssm2602.txt | 19 +++++++++++++++++++ sound/soc/codecs/Kconfig | 8 ++++++-- sound/soc/codecs/ssm2602-i2c.c | 9 +++++++++ sound/soc/codecs/ssm2602-spi.c | 9 +++++++++ 4 files changed, 43 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2602.txt diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2602.txt b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt new file mode 100644 index 0000000..3b3302f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2602.txt @@ -0,0 +1,19 @@ +Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices + +SSM2602 support both I2C and SPI as the configuration interface, +the selection is made by the MODE strap-in pin. +SSM2603 and SSM2604 only support I2C as the configuration interface. + +Required properties: + + - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + Example: + + ssm2602: ssm2602@1a { + compatible = "adi,ssm2602"; + reg = <0x1a>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..3649e73 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -520,12 +520,16 @@ config SND_SOC_SSM2602 tristate config SND_SOC_SSM2602_SPI + tristate "Analog Devices SSM2602 CODEC - SPI" + depends on SPI_MASTER select SND_SOC_SSM2602 - tristate + select REGMAP_SPI config SND_SOC_SSM2602_I2C + tristate "Analog Devices SSM2602 CODEC - I2C" + depends on I2C select SND_SOC_SSM2602 - tristate + select REGMAP_I2C config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c index abd63d5..0d9779d 100644 --- a/sound/soc/codecs/ssm2602-i2c.c +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -41,10 +41,19 @@ static const struct i2c_device_id ssm2602_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { .compatible = "adi,ssm2603", }, + { .compatible = "adi,ssm2604", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct i2c_driver ssm2602_i2c_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_i2c_probe, .remove = ssm2602_i2c_remove, diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c index 2bf55e2..fb4cc46 100644 --- a/sound/soc/codecs/ssm2602-spi.c +++ b/sound/soc/codecs/ssm2602-spi.c @@ -26,10 +26,19 @@ static int ssm2602_spi_remove(struct spi_device *spi) return 0; } +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { .compatible = "adi,ssm2603", }, + { .compatible = "adi,ssm2604", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct spi_driver ssm2602_spi_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_spi_probe, .remove = ssm2602_spi_remove, -- 1.9.1

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[alsa-devel] [BUG] AMD vga-switcheroo deactivates GPU+soundcard - ALSA can't handle this.
by Paul Ludwig Konecny 29 Sep '14

29 Sep '14

Hi all, I'm an average user and not a programmer. David Henningsson advised me to contact this mailing-list to report my issue. As this is my first post here I hope I'm doing everything right. I'm having a problem with my HP EliteBook 850 G1. This machine features Intel(i965, HD4400) / AMD(CIK, mesa-radeonsi, HD8750M) hybrid-graphics which is handled by vgaswitcheroo/PRIME and runtime power management. When the discrete GPU and it's soundchip are deactivated by runtime power management it seems that it is still reported to ALSA as available. This "hangs" my entire sound system. I don't get any sound out of this machine anymore. As you can see in my alsa-info run it took around 400 seconds to access HDMI2 which is the radeon soundchip. I believe that alsa-info only finished because I did a "xrandr --listproviders" which breifly powered up the AMD GPU. As this corresponds with the timespan I stared alsa-info and issuing the xrandr command. [ 36.768463] HDMI ATI/AMD: no speaker allocation for ELD [ 36.804442] hda-intel 0000:03:00.1: Disabling via VGA-switcheroo [ 36.836422] hda-intel 0000:03:00.1: Cannot lock devices! [ 37.219319] [drm] Disabling audio 0 support -- [ 464.855900] status: r [ 465.620947] hda-intel 0000:03:00.1: Enabling via VGA-switcheroo [ 465.621063] snd_hda_intel 0000:03:00.1: irq 63 for MSI/MSI-X [ 465.623672] HDMI ATI/AMD: no speaker allocation for ELD At first I believed it to be a pulseaudio problem and after several bug reports I think I have traced the bug to the ALSA drivers. This bug report is the main/original one: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1310260 David Henningsson pointed out here: https://bugs.freedesktop.org/show_bug.cgi?id=83446 that he believes the bug to be in the alsa driver used by my machine. This is why he changed the affected package in launchpad from "pulseaudio" to "alsa-driver". He is also suggested posting this issue here as the alsa bug tracker was taken offline. A workaround that involves disabling the card via etc/modprobe.d (see ubuntu bugreport) circumvents this issue for the time being but the problem is that this kills my audio output via the laptop's DisplayPort (It has no native HDMI Port). The output of my alsa-info.sh run can be found here: http://www.alsa-project.org/db/?f=fd4de60706f487ec48e41f7ac451030bbe79a9d7 I hope you can help me! Thanks in advance! Paul P.S.: There is already a message to this list which is awaiting moderation. David Henningsson told me to subscribe and send it again this time directly to the list. The message in moderation can therefore be deleted.

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