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

[alsa-devel] [PATCH - alsa-lib 1/1] Patch for another bug in snd_pcm_area_silence().
by furrywolf 05 Feb '18

05 Feb '18

Only silence areas 64 bits at a time if it's possible to do so, which is when either the silence values are all zero, or when the format width divides evenly into 64 bits. For formats that are neither of these, let the width-specific code handle the entire silencing. Silencing formats that are not evenly divisible into 64 bits in 64-bit chunks, when the data isn't just zeroes, results in values being written to shifting positions in the sample, giving garbage. Makes Takashi Sakamoto's tester happy for all tests. Yay! (And Thanks!) Signed-off-by: furrywolf <alsa2(a)bushytails.net> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1753cda..5f9bd9f 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -2947,7 +2947,7 @@ int snd_pcm_area_silence(const snd_pcm_channel_area_t *dst_area, snd_pcm_uframes dst = snd_pcm_channel_area_addr(dst_area, dst_offset); width = snd_pcm_format_physical_width(format); silence = snd_pcm_format_silence_64(format); - if (dst_area->step == (unsigned int) width) { + if (dst_area->step == (unsigned int) width && (!silence || !(64 % width))) { unsigned int dwords = samples * width / 64; uint64_t *dstp = (uint64_t *)dst; samples -= dwords * 64 / width; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: dapm: fix debugfs read using path->connected
by kaichieh.chuang＠mediatek.com 05 Feb '18

05 Feb '18

From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> This fix a bug in dapm_widget_power_read_file(), where it may sent opposite order of source/sink widget into the p->connected(). for example, static int connected_check(source, sink); {"w_sink", NULL, "w_source", connected_check} the dapm_widget_power_read_file() will query p->connected() in following case p->conneted("w_source", "w_sink") p->conneted("w_sink", "w_source") we should avoid the last case, since it's the wrong order (source/sink) as declared in snd_soc_dapm_route. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a10b21c..ee6d9d9a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2026,7 +2026,7 @@ static ssize_t dapm_widget_power_read_file(struct file *fi le, snd_soc_dapm_for_each_direction(dir) { rdir = SND_SOC_DAPM_DIR_REVERSE(dir); snd_soc_dapm_widget_for_each_path(w, dir, p) { - if (p->connected && !p->connected(w, p->node[rdir])) + if (p->connected && !p->connected(p->source, p->sink)) continue; if (!p->connect) -- 1.7.9.5 ************* Email Confidentiality Notice ******************** The information contained in this e-mail message (including any attachments) may be confidential, proprietary, privileged, or otherwise exempt from disclosure under applicable laws. It is intended to be conveyed only to the designated recipient(s). Any use, dissemination, distribution, printing, retaining or copying of this e-mail (including its attachments) by unintended recipient(s) is strictly prohibited and may be unlawful. If you are not an intended recipient of this e-mail, or believe that you have received this e-mail in error, please notify the sender immediately (by replying to this e-mail), delete any and all copies of this e-mail (including any attachments) from your system, and do not disclose the content of this e-mail to any other person. Thank you!

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Re: [alsa-devel] [PATCH] alsaloop: fix a typo in the comparison
by Jaroslav Kysela 05 Feb '18

05 Feb '18

Dne 5.2.2018 v 07:47 Kirill Marinushkin napsal(a): > Hello maintainers, > > I would like to suggest you a patch which fixes a typo in the alsa-utils > alsaloop. Applied. Thanks. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] [PATCH v2 0/2] Add support for AK5558 ADC
by Daniel Baluta 05 Feb '18

05 Feb '18

We support normal mode, TDM mode and pm. Changes since v1: [addressed comments from Andy and Fabio] * fix GPIO polarity from active high to active low for correct documentation * fix license header by using SPDX identifier * remove debug prints at the beginning of functions. * only support auto clock switching (manual switching was dead code anyway) (in the future we could add a DT property to choose between manual and auto) * Use gpiod API * use GENMASK * introduce power_off/power_on One open question is the resume sequence which appears to need power_off/power_on. Just power_on alone isn't enough. With just power_on after resume aplay plays a song for 1 seconds and then the sound stops. Datasheet says, page 55" (1) The PDN pin should be held to “L” for more than 150 ns after AVDD and TVDD are powered up. Daniel Baluta (2): ASoC: codecs: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC Documentation/devicetree/bindings/sound/ak5558.txt | 23 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 626 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 52 ++ 5 files changed, 709 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: codecs: Add support for AK4458 DAC driver
by Rob Herring 05 Feb '18

05 Feb '18

On Wed, Jan 31, 2018 at 03:20:09PM +0200, Cosmin-Gabriel Samoila wrote: > The AK4458 is a 32-bit 8ch Premium DAC that corresponds > to a 768kHz PCM input and an 11.2MHz DSD input at maximum. > It supports I2S, DSD and TDM modes with 24 or 32 bit MSB > or 16, 24, 32 LSB formats. Its datasheet is available here: > https://www.akm.com/akm/en/file/datasheet/AK4458VN.pdf > > Signed-off-by: Junichi Wakasugi <wakasugi.jb(a)om.asahi-kasei.co.jp> > Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> > --- > Documentation/devicetree/bindings/sound/ak4458.txt | 23 + Separate patch please. > sound/soc/codecs/Kconfig | 18 + > sound/soc/codecs/Makefile | 6 + > sound/soc/codecs/ak4458-i2c.c | 79 ++ > sound/soc/codecs/ak4458-spi.c | 61 ++ > sound/soc/codecs/ak4458.c | 1132 ++++++++++++++++++++ > sound/soc/codecs/ak4458.h | 127 +++ > 7 files changed, 1446 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/ak4458.txt > create mode 100644 sound/soc/codecs/ak4458-i2c.c > create mode 100644 sound/soc/codecs/ak4458-spi.c > create mode 100644 sound/soc/codecs/ak4458.c > create mode 100644 sound/soc/codecs/ak4458.h > > diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt > new file mode 100644 > index 0000000..b9e6eb3 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/ak4458.txt > @@ -0,0 +1,23 @@ > +AK4458 audio DAC > + > +This device supports both I2C and SPI modes. > + > +Required properties: > + > +- compatible : "asahi-kasei,ak4458" > +- reg : The I2C address of the device for I2C, the chip select number for SPI. > +- asahi-kasei,pdn-gpios: A GPIO specifier for the GPIO controlling > + the power down & reset pin. reset-gpios like the Asahi Kasei ADC under review. > +- asahi-kasei,mute-gpios: A GPIO specifier for the GPIO controlling > + the soft mute pin. State the active state. > + > +Example: > + > +&i2c { > + ak4458: ak4458@10 { dac@10 > + compatible = "asahi-kasei,ak4458"; > + reg = <0x10>; > + asahi-kasei,pdn-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH> > + asahi-kasei,mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH> > + }; > +};

1 0

[alsa-devel] [PATCH v2 1/3] cplay: add option to specify codec ID
by Katsuhiro Suzuki 02 Feb '18

02 Feb '18

This patch adds a -I command line option to set the codec ID, same as crecord. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- src/utils/cplay.c | 86 +++++++++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 78 insertions(+), 8 deletions(-) diff --git a/src/utils/cplay.c b/src/utils/cplay.c index bc95f0d..1eb21ce 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -74,25 +74,75 @@ #include "tinycompress/tinymp3.h" static int verbose; +static const unsigned int DEFAULT_CODEC_ID = SND_AUDIOCODEC_PCM; + +static const struct { + const char *name; + unsigned int id; +} codec_ids[] = { + { "PCM", SND_AUDIOCODEC_PCM }, + { "MP3", SND_AUDIOCODEC_MP3 }, + { "AMR", SND_AUDIOCODEC_AMR }, + { "AMRWB", SND_AUDIOCODEC_AMRWB }, + { "AMRWBPLUS", SND_AUDIOCODEC_AMRWBPLUS }, + { "AAC", SND_AUDIOCODEC_AAC }, + { "WMA", SND_AUDIOCODEC_WMA }, + { "REAL", SND_AUDIOCODEC_REAL }, + { "VORBIS", SND_AUDIOCODEC_VORBIS }, + { "FLAC", SND_AUDIOCODEC_FLAC }, + { "IEC61937", SND_AUDIOCODEC_IEC61937 }, + { "G723_1", SND_AUDIOCODEC_G723_1 }, + { "G729", SND_AUDIOCODEC_G729 }, +/* BESPOKE isn't defined on older kernels */ +#ifdef SND_AUDIOCODEC_BESPOKE + { "BESPOKE", SND_AUDIOCODEC_BESPOKE }, +#endif +}; +#define CPLAY_NUM_CODEC_IDS (sizeof(codec_ids) / sizeof(codec_ids[0])) + +static const char *codec_name_from_id(unsigned int id) +{ + static char hexname[12]; + int i; + + for (i = 0; i < CPLAY_NUM_CODEC_IDS; ++i) { + if (codec_ids[i].id == id) + return codec_ids[i].name; + } + + snprintf(hexname, sizeof(hexname), "0x%x", id); + return hexname; /* a static is safe because we're single-threaded */ +} static void usage(void) { + int i; + fprintf(stderr, "usage: cplay [OPTIONS] filename\n" "-c\tcard number\n" "-d\tdevice node\n" + "-I\tspecify codec ID (default is mp3)\n" "-b\tbuffer size\n" "-f\tfragments\n\n" "-v\tverbose mode\n" "-h\tPrints this help list\n\n" "Example:\n" "\tcplay -c 1 -d 2 test.mp3\n" - "\tcplay -f 5 test.mp3\n"); + "\tcplay -f 5 test.mp3\n\n" + "Valid codec IDs:\n"); + + for (i = 0; i < CPLAY_NUM_CODEC_IDS; ++i) + fprintf(stderr, "%s%c", codec_ids[i].name, + ((i + 1) % 8) ? ' ' : '\n'); + + fprintf(stderr, "\nor the value in decimal or hex\n"); exit(EXIT_FAILURE); } void play_samples(char *name, unsigned int card, unsigned int device, - unsigned long buffer_size, unsigned int frag); + unsigned long buffer_size, unsigned int frag, + unsigned long codec_id); struct mp3_header { uint16_t sync; @@ -156,15 +206,15 @@ int main(int argc, char **argv) { char *file; unsigned long buffer_size = 0; - int c; + int c, i; unsigned int card = 0, device = 0, frag = 0; - + unsigned int codec_id = SND_AUDIOCODEC_MP3; if (argc < 2) usage(); verbose = 0; - while ((c = getopt(argc, argv, "hvb:f:c:d:")) != -1) { + while ((c = getopt(argc, argv, "hvb:f:c:d:I:")) != -1) { switch (c) { case 'h': usage(); @@ -181,6 +231,25 @@ int main(int argc, char **argv) case 'd': device = strtol(optarg, NULL, 10); break; + case 'I': + if (optarg[0] == '0') { + codec_id = strtol(optarg, NULL, 0); + } else { + for (i = 0; i < CPLAY_NUM_CODEC_IDS; ++i) { + if (strcmp(optarg, + codec_ids[i].name) == 0) { + codec_id = codec_ids[i].id; + break; + } + } + + if (i == CPLAY_NUM_CODEC_IDS) { + fprintf(stderr, "Unrecognised ID: %s\n", + optarg); + usage(); + } + } + break; case 'v': verbose = 1; break; @@ -193,14 +262,15 @@ int main(int argc, char **argv) file = argv[optind]; - play_samples(file, card, device, buffer_size, frag); + play_samples(file, card, device, buffer_size, frag, codec_id); fprintf(stderr, "Finish Playing.... Close Normally\n"); exit(EXIT_SUCCESS); } void play_samples(char *name, unsigned int card, unsigned int device, - unsigned long buffer_size, unsigned int frag) + unsigned long buffer_size, unsigned int frag, + unsigned long codec_id) { struct compr_config config; struct snd_codec codec; @@ -292,7 +362,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, printf("Playing file %s On Card %u device %u, with buffer of %lu bytes\n", name, card, device, buffer_size); printf("Format %u Channels %u, %u Hz, Bit Rate %d\n", - SND_AUDIOCODEC_MP3, channels, rate, bits); + codec.id, codec.ch_in, codec.sample_rate, codec.bit_rate); compress_start(compress); if (verbose) -- 2.15.0

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[alsa-devel] [PATCH] ASoC: codecs: Add support for AK5558 ADC driver
by Daniel Baluta 02 Feb '18

02 Feb '18

AK5558 is a 32-bit, 768 kHZ sampling, differential input ADC for digital audio systems. Datasheet is available at: https://www.akm.com/akm/en/file/datasheet/AK5558VN.pdf Initial patch includes support for normal and TDM modes. Signed-off-by: Junichi Wakasugi <wakasugi.jb(a)om.asahi-kasei.co.jp> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak5558.txt | 20 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 754 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 60 ++ 5 files changed, 842 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h diff --git a/Documentation/devicetree/bindings/sound/ak5558.txt b/Documentation/devicetree/bindings/sound/ak5558.txt new file mode 100644 index 0000000..c6c71af --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5558.txt @@ -0,0 +1,20 @@ +AK5558 8 channel differential 32-bit delta-sigma ADC + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak5558" +- reg : The I2C address of the device for I2C. +- asahi-kasei,pdn-gpios: A GPIO specifier for the GPIO controlling + the power down & reset pin. + +Example: + +&i2c { + ak5558: ak5558@10 { + compatible = "asahi-kasei,ak5558"; + reg = <0x10>; + asahi-kasei,pdn-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c367d11..84a15d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_AK5386 + select SND_SOC_AK5558 if I2C select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_BT_SCO @@ -394,6 +395,11 @@ config SND_SOC_AK4671 config SND_SOC_AK5386 tristate "AKM AK5638 CODEC" +config SND_SOC_AK5558 + tristate "AKM AK5558 CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_ALC5623 tristate "Realtek ALC5623 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 77c1818..3a3d8b1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -33,6 +33,7 @@ snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o +snd-soc-ak5558-objs := ak5558.o snd-soc-arizona-objs := arizona.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o @@ -271,6 +272,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o +obj-$(CONFIG_SND_SOC_AK5558) += snd-soc-ak5558.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c new file mode 100644 index 0000000..eeaeea9 --- /dev/null +++ b/sound/soc/codecs/ak5558.c @@ -0,0 +1,754 @@ +/* + * ak5558.c -- audio driver for AK5558 ADC + * + * Copyright (C) 2015 Asahi Kasei Microdevices Corporation + * Copyright 2018 NXP + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/pm_runtime.h> + +#include "ak5558.h" + +#define AK5558_SLAVE_CKS_AUTO + +/* AK5558 Codec Private Data */ +struct ak5558_priv { + struct snd_soc_codec codec; + struct regmap *regmap; + struct i2c_client *i2c; + int fs; /* Sampling Frequency */ + int rclk; /* Master Clock */ + int pdn_gpio; /* Power on / Reset GPIO */ + int slots; + int slot_width; +}; + +/* ak5558 register cache & default register settings */ +static const struct reg_default ak5558_reg[] = { + { 0x0, 0xFF }, /* 0x00 AK5558_00_POWER_MANAGEMENT1 */ + { 0x1, 0x01 }, /* 0x01 AK5558_01_POWER_MANAGEMENT2 */ + { 0x2, 0x01 }, /* 0x02 AK5558_02_CONTROL1 */ + { 0x3, 0x00 }, /* 0x03 AK5558_03_CONTROL2 */ + { 0x4, 0x00 }, /* 0x04 AK5558_04_CONTROL3 */ + { 0x5, 0x00 } /* 0x05 AK5558_05_DSD */ +}; + +static const char * const mono_texts[] = { + "8 Slot", "2 Slot", "4 Slot", "1 Slot", +}; + +static const struct soc_enum ak5558_mono_enum[] = { + SOC_ENUM_SINGLE(AK5558_01_POWER_MANAGEMENT2, 1, + ARRAY_SIZE(mono_texts), mono_texts) +}; + +static const char * const tdm_texts[] = { + "Off", "TDM128", "TDM256", "TDM512", +}; + +static const char * const digfil_texts[] = { + "Sharp Roll-Off", "Show Roll-Off", + "Short Delay Sharp Roll-Off", "Short Delay Show Roll-Off", +}; + +static const struct soc_enum ak5558_adcset_enum[] = { + SOC_ENUM_SINGLE(AK5558_03_CONTROL2, 5, + ARRAY_SIZE(tdm_texts), tdm_texts), + SOC_ENUM_SINGLE(AK5558_04_CONTROL3, 0, + ARRAY_SIZE(digfil_texts), digfil_texts), +}; + +static const char * const dsdon_texts[] = { + "PCM", "DSD", +}; + +static const char * const dsdsel_texts[] = { + "64fs", "128fs", "256fs" +}; + +static const char * const dckb_texts[] = { + "Falling", "Rising", +}; + +static const char * const dcks_texts[] = { + "512fs", "768fs", +}; + +static const struct soc_enum ak5558_dsdset_enum[] = { + SOC_ENUM_SINGLE(AK5558_04_CONTROL3, 7, + ARRAY_SIZE(dsdon_texts), dsdon_texts), + SOC_ENUM_SINGLE(AK5558_05_DSD, 0, + ARRAY_SIZE(dsdsel_texts), dsdsel_texts), + SOC_ENUM_SINGLE(AK5558_05_DSD, 2, ARRAY_SIZE(dckb_texts), dckb_texts), + SOC_ENUM_SINGLE(AK5558_05_DSD, 5, ARRAY_SIZE(dcks_texts), dcks_texts), +}; + +static const struct snd_kcontrol_new ak5558_snd_controls[] = { + SOC_ENUM("AK5558 Monaural Mode", ak5558_mono_enum[0]), + SOC_ENUM("AK5558 TDM mode", ak5558_adcset_enum[0]), + SOC_ENUM("AK5558 Digital Filter", ak5558_adcset_enum[1]), + + SOC_ENUM("AK5558 DSD Mode", ak5558_dsdset_enum[0]), + SOC_ENUM("AK5558 Frequency of DCLK", ak5558_dsdset_enum[1]), + SOC_ENUM("AK5558 Polarity of DCLK", ak5558_dsdset_enum[2]), + SOC_ENUM("AK5558 Master Clock Frequency at DSD Mode", + ak5558_dsdset_enum[3]), + + SOC_SINGLE("AK5558 DSD Phase Modulation", AK5558_05_DSD, 3, 1, 0), +}; + +static const char * const ak5558_channel_select_texts[] = {"Off", "On"}; + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel1_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel1_mux_control = + SOC_DAPM_ENUM("Ch1 Switch", ak5558_channel1_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel2_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel2_mux_control = + SOC_DAPM_ENUM("Ch2 Switch", ak5558_channel2_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel3_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel3_mux_control = + SOC_DAPM_ENUM("Ch3 Switch", ak5558_channel3_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel4_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel4_mux_control = + SOC_DAPM_ENUM("Ch4 Switch", ak5558_channel4_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel5_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel5_mux_control = + SOC_DAPM_ENUM("Ch5 Switch", ak5558_channel5_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel6_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel6_mux_control = + SOC_DAPM_ENUM("Ch6 Switch", ak5558_channel6_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel7_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel7_mux_control = + SOC_DAPM_ENUM("Ch7 Switch", ak5558_channel7_mux_enum); + +static SOC_ENUM_SINGLE_VIRT_DECL(ak5558_channel8_mux_enum, + ak5558_channel_select_texts); + +static const struct snd_kcontrol_new ak5558_channel8_mux_control = + SOC_DAPM_ENUM("Ch8 Switch", ak5558_channel8_mux_enum); + +static const struct snd_soc_dapm_widget ak5558_dapm_widgets[] = { + /* Analog Input */ + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("AIN3"), + SND_SOC_DAPM_INPUT("AIN4"), + SND_SOC_DAPM_INPUT("AIN5"), + SND_SOC_DAPM_INPUT("AIN6"), + SND_SOC_DAPM_INPUT("AIN7"), + SND_SOC_DAPM_INPUT("AIN8"), + + SND_SOC_DAPM_MUX("AK5558 Ch1 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel1_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch2 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel2_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch3 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel3_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch4 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel4_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch5 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel5_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch6 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel6_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch7 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel7_mux_control), + SND_SOC_DAPM_MUX("AK5558 Ch8 Enable", SND_SOC_NOPM, 0, 0, + &ak5558_channel8_mux_control), + + SND_SOC_DAPM_ADC("ADC Ch1", NULL, AK5558_00_POWER_MANAGEMENT1, 0, 0), + SND_SOC_DAPM_ADC("ADC Ch2", NULL, AK5558_00_POWER_MANAGEMENT1, 1, 0), + SND_SOC_DAPM_ADC("ADC Ch3", NULL, AK5558_00_POWER_MANAGEMENT1, 2, 0), + SND_SOC_DAPM_ADC("ADC Ch4", NULL, AK5558_00_POWER_MANAGEMENT1, 3, 0), + SND_SOC_DAPM_ADC("ADC Ch5", NULL, AK5558_00_POWER_MANAGEMENT1, 4, 0), + SND_SOC_DAPM_ADC("ADC Ch6", NULL, AK5558_00_POWER_MANAGEMENT1, 5, 0), + SND_SOC_DAPM_ADC("ADC Ch7", NULL, AK5558_00_POWER_MANAGEMENT1, 6, 0), + SND_SOC_DAPM_ADC("ADC Ch8", NULL, AK5558_00_POWER_MANAGEMENT1, 7, 0), + + SND_SOC_DAPM_AIF_OUT("SDTO", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route ak5558_intercon[] = { + {"AK5558 Ch1 Enable", "On", "AIN1"}, + {"ADC Ch1", NULL, "AK5558 Ch1 Enable"}, + {"SDTO", NULL, "ADC Ch1"}, + + {"AK5558 Ch2 Enable", "On", "AIN2"}, + {"ADC Ch2", NULL, "AK5558 Ch2 Enable"}, + {"SDTO", NULL, "ADC Ch2"}, + + {"AK5558 Ch3 Enable", "On", "AIN3"}, + {"ADC Ch3", NULL, "AK5558 Ch3 Enable"}, + {"SDTO", NULL, "ADC Ch3"}, + + {"AK5558 Ch4 Enable", "On", "AIN4"}, + {"ADC Ch4", NULL, "AK5558 Ch4 Enable"}, + {"SDTO", NULL, "ADC Ch4"}, + + {"AK5558 Ch5 Enable", "On", "AIN5"}, + {"ADC Ch5", NULL, "AK5558 Ch5 Enable"}, + {"SDTO", NULL, "ADC Ch5"}, + + {"AK5558 Ch6 Enable", "On", "AIN6"}, + {"ADC Ch6", NULL, "AK5558 Ch6 Enable"}, + {"SDTO", NULL, "ADC Ch6"}, + + {"AK5558 Ch7 Enable", "On", "AIN7"}, + {"ADC Ch7", NULL, "AK5558 Ch7 Enable"}, + {"SDTO", NULL, "ADC Ch7"}, + + {"AK5558 Ch8 Enable", "On", "AIN8"}, + {"ADC Ch8", NULL, "AK5558 Ch8 Enable"}, + {"SDTO", NULL, "ADC Ch8"}, +}; + +static int ak5558_set_mcki(struct snd_soc_codec *codec, int fs, int rclk) +{ + u8 mode; +#ifndef AK5558_SLAVE_CKS_AUTO + int mcki_rate; +#endif + + dev_dbg(codec->dev, "%s fs=%d rclk=%d\n", __func__, fs, rclk); + + mode = snd_soc_read(codec, AK5558_02_CONTROL1); + mode &= ~AK5558_CKS; + +#ifdef AK5558_SLAVE_CKS_AUTO + mode |= AK5558_CKS_AUTO; +#else + if (fs != 0 && rclk != 0) { + if (rclk % fs) + return -EINVAL; + mcki_rate = rclk / fs; + + if (fs > 400000) { + switch (mcki_rate) { + case 32: + mode |= AK5558_CKS_32FS_768KHZ; + break; + case 48: + mode |= AK5558_CKS_48FS_768KHZ; + break; + case 64: + mode |= AK5558_CKS_64FS_768KHZ; + break; + default: + return -EINVAL; + } + } else if (fs > 200000) { + switch (mcki_rate) { + case 64: + mode |= AK5558_CKS_64FS_384KHZ; + break; + case 96: + mode |= AK5558_CKS_96FS_384KHZ; + break; + default: + return -EINVAL; + } + } else if (fs > 108000) { + switch (mcki_rate) { + case 128: + mode |= AK5558_CKS_128FS_192KHZ; + break; + case 192: + mode |= AK5558_CKS_192FS_192KHZ; + break; + default: + return -EINVAL; + } + } else if (fs > 54000) { + switch (mcki_rate) { + case 256: + mode |= AK5558_CKS_256FS_96KHZ; + break; + case 384: + mode |= AK5558_CKS_384FS_96KHZ; + break; + default: + return -EINVAL; + } + } else { + switch (mcki_rate) { + case 256: + mode |= AK5558_CKS_256FS_48KHZ; + break; + case 384: + mode |= AK5558_CKS_384FS_48KHZ; + break; + case 512: + mode |= AK5558_CKS_512FS_48KHZ; + break; + case 768: + mode |= AK5558_CKS_768FS_48KHZ; + break; + case 1024: + if (fs > 32000) + return -EINVAL; + mode |= AK5558_CKS_1024FS_16KHZ; + break; + default: + return -EINVAL; + } + } + } +#endif + snd_soc_update_bits(codec, AK5558_02_CONTROL1, AK5558_CKS, mode); + + return 0; +} + +static int ak5558_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + u8 bits; + int pcm_width = max(params_physical_width(params), ak5558->slot_width); + + dev_dbg(dai->dev, "%s(%d)\n", __func__, __LINE__); + + /* set master/slave audio interface */ + bits = snd_soc_read(codec, AK5558_02_CONTROL1); + bits &= ~AK5558_BITS; + + switch (pcm_width) { + case 16: + bits |= AK5558_DIF_24BIT_MODE; + break; + case 32: + bits |= AK5558_DIF_32BIT_MODE; + break; + default: + return -EINVAL; + } + + ak5558->fs = params_rate(params); + snd_soc_update_bits(codec, AK5558_02_CONTROL1, AK5558_BITS, bits); + + ak5558_set_mcki(codec, ak5558->fs, ak5558->rclk); + + return 0; +} + +static int ak5558_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(dai->dev, "%s(%d)\n", __func__, __LINE__); + + ak5558->rclk = freq; + ak5558_set_mcki(codec, ak5558->fs, ak5558->rclk); + + return 0; +} + +static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 format; + + dev_dbg(dai->dev, "%s(%d)\n", __func__, __LINE__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: +#ifdef AK5558_SLAVE_CKS_AUTO + break; +#else + return -EINVAL; +#endif + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "Clock mode unsupported"); + return -EINVAL; + } + + /* set master/slave audio interface */ + format = snd_soc_read(codec, AK5558_02_CONTROL1); + format &= ~AK5558_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK5558_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK5558_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_B: + format |= AK5558_DIF_MSB_MODE; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, AK5558_02_CONTROL1, AK5558_DIF, format); + + return 0; +} + +static int ak5558_set_dai_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + int ndt; + + if (mute) { + ndt = 0; + if (ak5558->fs != 0) + ndt = 583000 / ak5558->fs; + if (ndt < 5) + ndt = 5; + msleep(ndt); + } + + return 0; +} + +static int ak5558_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "%s bias level=%d\n", __func__, (int)level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK5558_00_POWER_MANAGEMENT1, 0x00); + break; + } + return 0; +} + +static int ak5558_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + int tdm_mode = 0; + int reg; + + ak5558->slots = slots; + ak5558->slot_width = slot_width; + + switch (slots * slot_width) { + case 128: + tdm_mode = AK5558_MODE_TDM128; + break; + case 256: + tdm_mode = AK5558_MODE_TDM256; + break; + case 512: + tdm_mode = AK5558_MODE_TDM512; + break; + default: + tdm_mode = AK5558_MODE_NORMAL; + break; + } + + reg = snd_soc_read(codec, AK5558_03_CONTROL2); + reg &= ~AK5558_MODE_BITS; + reg |= tdm_mode; + snd_soc_write(codec, AK5558_03_CONTROL2, reg); + + return 0; +} + +#define AK5558_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static const unsigned int ak5558_rates[] = { + 8000, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 352800, + 384000, 705600, 768000, 1411200, + 2822400, +}; + +static const struct snd_pcm_hw_constraint_list ak5558_rate_constraints = { + .count = ARRAY_SIZE(ak5558_rates), + .list = ak5558_rates, +}; + +static int ak5558_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret; + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &ak5558_rate_constraints); + return ret; +} + +static struct snd_soc_dai_ops ak5558_dai_ops = { + .startup = ak5558_startup, + .hw_params = ak5558_hw_params, + .set_sysclk = ak5558_set_dai_sysclk, + .set_fmt = ak5558_set_dai_fmt, + .digital_mute = ak5558_set_dai_mute, + .set_tdm_slot = ak5558_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ak5558_dai = { + .name = "ak5558-aif", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = AK5558_FORMATS, + }, + .ops = &ak5558_dai_ops, +}; + +static int ak5558_init_reg(struct snd_soc_codec *codec) +{ + int ret; + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s(%d)\n", __func__, __LINE__); + + usleep_range(10000, 11000); + if (gpio_is_valid(ak5558->pdn_gpio)) { + gpio_set_value_cansleep(ak5558->pdn_gpio, 0); + usleep_range(1000, 2000); + gpio_set_value_cansleep(ak5558->pdn_gpio, 1); + usleep_range(1000, 2000); + } + + ret = snd_soc_write(codec, AK5558_00_POWER_MANAGEMENT1, 0x0); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, AK5558_02_CONTROL1, AK5558_CKS, + AK5558_CKS_AUTO); + if (ret < 0) + return ret; + + return 0; +} + +static int ak5558_probe(struct snd_soc_codec *codec) +{ + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + dev_dbg(codec->dev, "%s(%d)\n", __func__, __LINE__); + + ret = ak5558_init_reg(codec); + + ak5558->fs = 48000; + ak5558->rclk = 0; + + return ret; +} + +static int ak5558_remove(struct snd_soc_codec *codec) +{ + struct ak5558_priv *ak5558 = snd_soc_codec_get_drvdata(codec); + + ak5558_set_bias_level(codec, SND_SOC_BIAS_OFF); + + if (gpio_is_valid(ak5558->pdn_gpio)) { + gpio_set_value_cansleep(ak5558->pdn_gpio, 0); + usleep_range(1000, 2000); + } + + return 0; +} + +#ifdef CONFIG_PM +static int ak5558_runtime_suspend(struct device *dev) +{ + struct ak5558_priv *ak5558 = dev_get_drvdata(dev); + + regcache_cache_only(ak5558->regmap, true); + + if (gpio_is_valid(ak5558->pdn_gpio)) { + gpio_set_value_cansleep(ak5558->pdn_gpio, 0); + usleep_range(1000, 2000); + } + + return 0; +} + +static int ak5558_runtime_resume(struct device *dev) +{ + struct ak5558_priv *ak5558 = dev_get_drvdata(dev); + + if (gpio_is_valid(ak5558->pdn_gpio)) { + gpio_set_value_cansleep(ak5558->pdn_gpio, 0); + usleep_range(1000, 2000); + gpio_set_value_cansleep(ak5558->pdn_gpio, 1); + usleep_range(1000, 2000); + } + + regcache_cache_only(ak5558->regmap, false); + regcache_mark_dirty(ak5558->regmap); + + return regcache_sync(ak5558->regmap); +} +#endif + +const struct dev_pm_ops ak5558_pm = { + SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +struct snd_soc_codec_driver soc_codec_dev_ak5558 = { + .probe = ak5558_probe, + .remove = ak5558_remove, + .idle_bias_off = true, + .set_bias_level = ak5558_set_bias_level, + + .component_driver = { + .controls = ak5558_snd_controls, + .num_controls = ARRAY_SIZE(ak5558_snd_controls), + .dapm_widgets = ak5558_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5558_dapm_widgets), + .dapm_routes = ak5558_intercon, + .num_dapm_routes = ARRAY_SIZE(ak5558_intercon), + }, +}; + +static const struct regmap_config ak5558_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK5558_05_DSD, + .reg_defaults = ak5558_reg, + .num_reg_defaults = ARRAY_SIZE(ak5558_reg), + .cache_type = REGCACHE_RBTREE, +}; + +static int ak5558_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device_node *np = i2c->dev.of_node; + struct ak5558_priv *ak5558; + int ret = 0; + + dev_err(&i2c->dev, "%s(%d)\n", __func__, __LINE__); + + ak5558 = devm_kzalloc(&i2c->dev, sizeof(struct ak5558_priv), + GFP_KERNEL); + if (!ak5558) + return -ENOMEM; + + ak5558->regmap = devm_regmap_init_i2c(i2c, &ak5558_regmap); + if (IS_ERR(ak5558->regmap)) + return PTR_ERR(ak5558->regmap); + + i2c_set_clientdata(i2c, ak5558); + ak5558->i2c = i2c; + + ak5558->pdn_gpio = of_get_named_gpio(np, "ak5558,pdn-gpio", 0); + if (gpio_is_valid(ak5558->pdn_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ak5558->pdn_gpio, + GPIOF_OUT_INIT_LOW, "ak5558,pdn"); + if (ret) { + dev_err(&i2c->dev, "unable to get pdn gpio\n"); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak5558, + &ak5558_dai, 1); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + + return 0; +} + +static int ak5558_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + pm_runtime_disable(&client->dev); + + return 0; +} + +static const struct of_device_id ak5558_i2c_dt_ids[] = { + { .compatible = "asahi-kasei,ak5558"}, + { } +}; + +static const struct i2c_device_id ak5558_i2c_id[] = { + { "ak5558", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak5558_i2c_id); + +static struct i2c_driver ak5558_i2c_driver = { + .driver = { + .name = "ak5558", + .of_match_table = of_match_ptr(ak5558_i2c_dt_ids), + .pm = &ak5558_pm, + }, + .probe = ak5558_i2c_probe, + .remove = ak5558_i2c_remove, + .id_table = ak5558_i2c_id, +}; + +module_i2c_driver(ak5558_i2c_driver); + +MODULE_AUTHOR("Junichi Wakasugi <wakasugi.jb(a)om.asahi-kasei.co.jp>"); +MODULE_AUTHOR("Mihai Serban <mihai.serban(a)nxp.com>"); +MODULE_DESCRIPTION("ASoC AK5558 ADC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak5558.h b/sound/soc/codecs/ak5558.h new file mode 100644 index 0000000..0e40c4a --- /dev/null +++ b/sound/soc/codecs/ak5558.h @@ -0,0 +1,60 @@ +/* + * ak5558.h -- audio driver for AK5558 + * + * Copyright (C) 2016 Asahi Kasei Microdevices Corporation + * Copyright 2018 NXP + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * 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. + */ + +#ifndef _AK5558_H +#define _AK5558_H + +#define AK5558_00_POWER_MANAGEMENT1 0x00 +#define AK5558_01_POWER_MANAGEMENT2 0x01 +#define AK5558_02_CONTROL1 0x02 +#define AK5558_03_CONTROL2 0x03 +#define AK5558_04_CONTROL3 0x04 +#define AK5558_05_DSD 0x05 + +/* AK5558_02_CONTROL1 fields */ +#define AK5558_DIF 0x02 +#define AK5558_DIF_MSB_MODE (0 << 1) +#define AK5558_DIF_I2S_MODE (1 << 1) + +#define AK5558_BITS 0x04 +#define AK5558_DIF_24BIT_MODE (0 << 2) +#define AK5558_DIF_32BIT_MODE (1 << 2) + +#define AK5558_CKS 0x78 +#define AK5558_CKS_128FS_192KHZ (0 << 3) +#define AK5558_CKS_192FS_192KHZ (1 << 3) +#define AK5558_CKS_256FS_48KHZ (2 << 3) +#define AK5558_CKS_256FS_96KHZ (3 << 3) +#define AK5558_CKS_384FS_96KHZ (4 << 3) +#define AK5558_CKS_384FS_48KHZ (5 << 3) +#define AK5558_CKS_512FS_48KHZ (6 << 3) +#define AK5558_CKS_768FS_48KHZ (7 << 3) +#define AK5558_CKS_64FS_384KHZ (8 << 3) +#define AK5558_CKS_32FS_768KHZ (9 << 3) +#define AK5558_CKS_96FS_384KHZ (10 << 3) +#define AK5558_CKS_48FS_768KHZ (11 << 3) +#define AK5558_CKS_64FS_768KHZ (12 << 3) +#define AK5558_CKS_1024FS_16KHZ (13 << 3) +#define AK5558_CKS_AUTO (15 << 3) + +/* AK5558_03_CONTROL2 fields */ +#define AK5558_MODE_BITS 0x60 +#define AK5558_MODE_NORMAL (0 << 5) +#define AK5558_MODE_TDM128 (1 << 5) +#define AK5558_MODE_TDM256 (2 << 5) +#define AK5558_MODE_TDM512 (3 << 5) + +#endif -- 2.7.4

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[alsa-devel] [GIT PULL]: kbl audio firmware update
by Vinod Koul 01 Feb '18

01 Feb '18

Hi, Please PULL to receive the update for KBL audio DSP firmware. The following changes since commit 2a713be25a44bd6cec90d8affc54b246a2ca9c7b: amdgpu: update uvd firmware for polaris asics (2018-01-19 07:16:32 -0500) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git kbl for you to fetch changes up to aab8b6fff9417a9938f6393dc09cf6ce90e55565: linux-firmware: intel: Update Kabylake audio firmware (2018-01-30 15:42:45 +0530) ---------------------------------------------------------------- Guneshwor Singh (1): linux-firmware: intel: Update Kabylake audio firmware WHENCE | 4 +++- intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v3266.bin | Bin 0 -> 243016 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_kbl_v3266.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH 1/1] ASoC: stm32: add of dependency for stm32 drivers
by Olivier Moysan 01 Feb '18

01 Feb '18

Add of dependency for STM32 ASoC drivers. DFSDM of dependency is already inherited from STM32_DFSDM_ADC dependency. Signed-off-by: olivier moysan <olivier.moysan(a)st.com> --- sound/soc/stm/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 3ad881f..b5375f9 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -3,6 +3,7 @@ menu "STMicroelectronics STM32 SOC audio support" config SND_SOC_STM32_SAI tristate "STM32 SAI interface (Serial Audio Interface) support" depends on ARCH_STM32 || COMPILE_TEST + depends on OF depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO @@ -12,6 +13,7 @@ config SND_SOC_STM32_SAI config SND_SOC_STM32_I2S tristate "STM32 I2S interface (SPI/I2S block) support" depends on ARCH_STM32 || COMPILE_TEST + depends on OF depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO @@ -21,6 +23,7 @@ config SND_SOC_STM32_I2S config SND_SOC_STM32_SPDIFRX tristate "STM32 S/PDIF receiver (SPDIFRX) support" depends on ARCH_STM32 || COMPILE_TEST + depends on OF depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO -- 1.9.1

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[alsa-devel] [PATCH 1/2] cplay: fix size and nmemb of mp3 header of fread
by Katsuhiro Suzuki 01 Feb '18

01 Feb '18

This patch swaps 'size' and 'nmemb' arguments of fread. The fread returns always 1 (because 'nmemb' is 1) so checking the return value is always failure. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- src/utils/cplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/utils/cplay.c b/src/utils/cplay.c index e0603fd..bc95f0d 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -220,7 +220,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, exit(EXIT_FAILURE); } - read = fread(&header, sizeof(header), 1, file); + read = fread(&header, 1, sizeof(header), file); if (read != sizeof(header)) { fprintf(stderr, "Unable to read header \n"); fclose(file); -- 2.15.0

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