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

[alsa-devel] [PATCH 1/4] ALSA: hda/realtek: ALC286 mic and headset-mode fixups for Acer Aspire U27-880
by Jian-Hong Pan 05 Dec '18

05 Dec '18

From: Chris Chiu <chiu(a)endlessm.com> Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. Unfortunately jack sensing for the headset mic is still not working. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0b289ba397f..f21d52eb2ed3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5516,6 +5516,7 @@ enum { ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, ALC295_FIXUP_HP_AUTO_MUTE, + ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6393,6 +6394,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_auto_mute_via_amp, }, + [ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7071,6 +7081,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, -- 2.11.0

3 6

[alsa-devel] [PATCH linux-next v3 5/7] clk: renesas: r8a77965: Add ADG clock
by Jiada Wang 05 Dec '18

05 Dec '18

From: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> This patch adds ADG clock to the R8A77965 SoC. Signed-off-by: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- drivers/clk/renesas/r8a77965-cpg-mssr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/clk/renesas/r8a77965-cpg-mssr.c b/drivers/clk/renesas/r8a77965-cpg-mssr.c index 1fcc411502da..3ce8b0b31647 100644 --- a/drivers/clk/renesas/r8a77965-cpg-mssr.c +++ b/drivers/clk/renesas/r8a77965-cpg-mssr.c @@ -211,6 +211,7 @@ static const struct mssr_mod_clk r8a77965_mod_clks[] __initconst = { DEF_MOD("can-if0", 916, R8A77965_CLK_S3D4), DEF_MOD("i2c6", 918, R8A77965_CLK_S0D6), DEF_MOD("i2c5", 919, R8A77965_CLK_S0D6), + DEF_MOD("adg", 922, R8A77965_CLK_S0D1), DEF_MOD("i2c-dvfs", 926, R8A77965_CLK_CP), DEF_MOD("i2c4", 927, R8A77965_CLK_S0D6), DEF_MOD("i2c3", 928, R8A77965_CLK_S0D6), -- 2.19.2

1 0

[alsa-devel] [PATCH linux-next v3 0/7] clk: renesas: adg: add AVB Clock
by Jiada Wang 05 Dec '18

05 Dec '18

on R-Car SoCs there are AVB Counter Clocks, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch-set adds 'adg' clock to R-Car Soc, and changes adg driver to register avb clocks when clock-cells of rcar_sound node is 2. --- v3: - Removed clock id header file, added device tree bindings documentation instead to describe clock id - Instead of hardcode parent clock name in adg driver, refer to parent clock via clock specifier in device tree with 'adg' name - Added patch to add ADG in r8a77965 - Some other fixes v2: - expends adg register size and register avb clocks instead of add new clk-avb driver - Add adg clock v1: initial version Jiada Wang (2): ASoC: rsnd: add avb clocks clk: renesas: Add binding document for ADG Takeshi Kihara (5): clk: renesas: r8a7795: Add ADG clock clk: renesas: r8a7796: Add ADG clock clk: renesas: r8a77990: Add ADG clock clk: renesas: r8a77995: Add ADG clock clk: renesas: r8a77965: Add ADG clock .../clock/renesas,rcar-adg-clocks.txt | 24 ++ drivers/clk/renesas/r8a7795-cpg-mssr.c | 1 + drivers/clk/renesas/r8a7796-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77965-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77990-cpg-mssr.c | 1 + drivers/clk/renesas/r8a77995-cpg-mssr.c | 1 + sound/soc/sh/rcar/adg.c | 316 +++++++++++++++++- sound/soc/sh/rcar/gen.c | 9 + sound/soc/sh/rcar/rsnd.h | 9 + 9 files changed, 354 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/renesas,rcar-adg-clocks.txt -- 2.19.2

1 4

[alsa-devel] axfer: check whether a terminal is referred for stdio
by GitHub pull_request - opened 05 Dec '18

05 Dec '18

alsa-project/alsa-utils pull request #5 was opened from takaswie: A reference to a terminal for standard input/output brings some troubles. For capture transmission, it can bring some terminal control codes and it's possible to make the terminal in disorder. For playback transmission, it can bring endless loop to read data for detection of type of container. This commit checks whether a terminal is referred for the standard input/output by a call isatty(3). When detecting a terminal, axfer run time prints message and goes to finish. Suggested-by: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-utils/pull/5 Patch URL : https://github.com/alsa-project/alsa-utils/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: pxa: remove raumfeld machine driver
by Daniel Mack 04 Dec '18

04 Dec '18

These boards are now fully ported to devicetree and make use of the simple-card driver, so the platform specific machine driver can be removed. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/Kconfig | 10 -- sound/soc/pxa/Makefile | 1 - sound/soc/pxa/raumfeld.c | 318 --------------------------------------- 3 files changed, 329 deletions(-) delete mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 943b44de1464..a6968bfe1d2d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -169,16 +169,6 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. -config SND_SOC_RAUMFELD - tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - depends on I2C && SPI_MASTER - select SND_PXA_SOC_SSP - select SND_SOC_CS4270 - select SND_SOC_AK4104 - help - Say Y if you want to add support for SoC audio on Raumfeld devices - config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" depends on SND_PXA2XX_SOC && MACH_H4700 && I2C diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..0ab2a9dcb720 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -49,6 +49,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o -obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c deleted file mode 100644 index 111a907c4eb9..000000000000 --- a/sound/soc/pxa/raumfeld.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * raumfeld_audio.c -- SoC audio for Raumfeld audio devices - * - * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> - * - * based on code from: - * - * Wolfson Microelectronics PLC. - * Openedhand Ltd. - * Liam Girdwood <lrg(a)slimlogic.co.uk> - * Richard Purdie <richard(a)openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/i2c.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> - -#include "pxa-ssp.h" - -#define GPIO_SPDIF_RESET (38) -#define GPIO_MCLK_RESET (111) -#define GPIO_CODEC_RESET (120) - -static struct i2c_client *max9486_client; -static struct i2c_board_info max9486_hwmon_info = { - I2C_BOARD_INFO("max9485", 0x63), -}; - -#define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 -#define MAX9485_MCLK_FREQ_225792 0x32 -#define MAX9485_MCLK_FREQ_245760 0x33 - -static void set_max9485_clk(char clk) -{ - i2c_master_send(max9486_client, &clk, 1); -} - -static void raumfeld_enable_audio(bool en) -{ - if (en) { - gpio_set_value(GPIO_MCLK_RESET, 1); - - /* wait some time to let the clocks become stable */ - msleep(100); - - gpio_set_value(GPIO_SPDIF_RESET, 1); - gpio_set_value(GPIO_CODEC_RESET, 1); - } else { - gpio_set_value(GPIO_MCLK_RESET, 0); - gpio_set_value(GPIO_SPDIF_RESET, 0); - gpio_set_value(GPIO_CODEC_RESET, 0); - } -} - -/* CS4270 */ -static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* set freq to 0 to enable all possible codec sample rates */ - snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); -} - -static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); - if (ret < 0) - return ret; - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static const struct snd_soc_ops raumfeld_cs4270_ops = { - .startup = raumfeld_cs4270_startup, - .shutdown = raumfeld_cs4270_shutdown, - .hw_params = raumfeld_cs4270_hw_params, -}; - -static int raumfeld_analog_suspend(struct snd_soc_card *card) -{ - raumfeld_enable_audio(false); - return 0; -} - -static int raumfeld_analog_resume(struct snd_soc_card *card) -{ - raumfeld_enable_audio(true); - return 0; -} - -/* AK4104 */ - -static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, clk = 0; - - switch (params_rate(params)) { - case 44100: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; - break; - case 48000: - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - clk = 12288000; - break; - case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_225792); - clk = 22579200; - break; - case 96000: - set_max9485_clk(MAX9485_MCLK_FREQ_245760); - clk = 24576000; - break; - default: - return -EINVAL; - } - - /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops raumfeld_ak4104_ops = { - .hw_params = raumfeld_ak4104_hw_params, -}; - -#define DAI_LINK_CS4270 \ -{ \ - .name = "CS4270", \ - .stream_name = "CS4270", \ - .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ - .codec_dai_name = "cs4270-hifi", \ - .codec_name = "cs4270.0-0048", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_cs4270_ops, \ -} - -#define DAI_LINK_AK4104 \ -{ \ - .name = "ak4104", \ - .stream_name = "Playback", \ - .cpu_dai_name = "pxa-ssp-dai.1", \ - .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ - .dai_fmt = SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBS_CFS, \ - .ops = &raumfeld_ak4104_ops, \ - .codec_name = "spi0.0", \ -} - -static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - DAI_LINK_CS4270, - DAI_LINK_AK4104, -}; - -static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - DAI_LINK_CS4270, -}; - -static struct snd_soc_card snd_soc_raumfeld_connector = { - .name = "Raumfeld Connector", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_connector_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct snd_soc_card snd_soc_raumfeld_speaker = { - .name = "Raumfeld Speaker", - .owner = THIS_MODULE, - .dai_link = snd_soc_raumfeld_speaker_dai, - .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), - .suspend_post = raumfeld_analog_suspend, - .resume_pre = raumfeld_analog_resume, -}; - -static struct platform_device *raumfeld_audio_device; - -static int __init raumfeld_audio_init(void) -{ - int ret; - - if (!machine_is_raumfeld_speaker() && - !machine_is_raumfeld_connector()) - return 0; - - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); - - if (!max9486_client) - return -ENOMEM; - - set_max9485_clk(MAX9485_MCLK_FREQ_122880); - - /* Register analog device */ - raumfeld_audio_device = platform_device_alloc("soc-audio", 0); - if (!raumfeld_audio_device) - return -ENOMEM; - - if (machine_is_raumfeld_speaker()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); - - ret = platform_device_add(raumfeld_audio_device); - if (ret < 0) { - platform_device_put(raumfeld_audio_device); - return ret; - } - - raumfeld_enable_audio(true); - return 0; -} - -static void __exit raumfeld_audio_exit(void) -{ - raumfeld_enable_audio(false); - - platform_device_unregister(raumfeld_audio_device); - - i2c_unregister_device(max9486_client); - - gpio_free(GPIO_MCLK_RESET); - gpio_free(GPIO_CODEC_RESET); - gpio_free(GPIO_SPDIF_RESET); -} - -module_init(raumfeld_audio_init); -module_exit(raumfeld_audio_exit); - -/* Module information */ -MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("Raumfeld audio SoC"); -MODULE_LICENSE("GPL"); -- 2.19.2

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Clapper
by Hans de Goede 04 Dec '18

04 Dec '18

The Clapper model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3 for the mclk, just like the Swanky model. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Clapper that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 9d9f6e41d81c..ad0c98383853 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -389,6 +389,13 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Clapper model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Clapper"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: indicates Channel and Mode for debug
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> For TDM debug purpose, indicating Channel and Mode is very useful. This patch indicate it if it has #define DEBUG Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6439a74..45ef295 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -329,7 +329,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, ssi->rate = rate; ssi->chan = chan; - dev_dbg(dev, "%s outputs %u Hz\n", rsnd_mod_name(mod), rate); + dev_dbg(dev, "%s outputs %d chan %u Hz\n", + rsnd_mod_name(mod), chan, rate); return 0; } @@ -360,6 +361,8 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); u32 cr_own = ssi->cr_own; @@ -371,6 +374,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); + if (is_tdm) + dev_dbg(dev, "TDM mode\n"); + if (is_tdm_split) + dev_dbg(dev, "TDM Split mode\n"); + cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai); if (rdai->bit_clk_inv) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup mod ID for CTU regmap read/write
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c16015f36cc12824 ("ASoC: rsnd: add .get_id/.get_id_sub") add new .get_id/.get_id_sub to indicate module ID/subID. It is used for SSIU and CTU. In SSIU case, subID indicates BUSIF, but register settings is based on SSIU ID. OTOH, in CTU case, subID indicates CTU channel, and register settings is based on it. This means regmap read/write function needs to care it. This patch fixup this issue. It can't play MIXed sound without this patch. Fixes: c16015f36cc12824 ("ASoC: rsnd: add .get_id/.get_id_sub") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 1 + sound/soc/sh/rcar/gen.c | 14 +++++++++++--- sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 4dda7d9..ff41a5e 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -317,6 +317,7 @@ static struct rsnd_mod_ops rsnd_ctu_ops = { .get_status = rsnd_mod_get_status, .id = rsnd_ctu_id, .id_sub = rsnd_ctu_id_sub, + .id_cmd = rsnd_mod_id_raw, }; struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca63940..3283d67 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -71,6 +71,14 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, return 1; } +static int rsnd_mod_id_cmd(struct rsnd_mod *mod) +{ + if (mod->ops->id_cmd) + return mod->ops->id_cmd(mod); + + return rsnd_mod_id(mod); +} + u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { @@ -81,7 +89,7 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + regmap_fields_read(gen->regs[reg], rsnd_mod_id_cmd(mod), &val); dev_dbg(dev, "r %s - %-18s (%4d) : %08x\n", rsnd_mod_name(mod), @@ -100,7 +108,7 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); + regmap_fields_force_write(gen->regs[reg], rsnd_mod_id_cmd(mod), data); dev_dbg(dev, "w %s - %-18s (%4d) : %08x\n", rsnd_mod_name(mod), @@ -117,7 +125,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, return; regmap_fields_force_update_bits(gen->regs[reg], - rsnd_mod_id(mod), mask, data); + rsnd_mod_id_cmd(mod), mask, data); dev_dbg(dev, "b %s - %-18s (%4d) : %08x/%08x\n", rsnd_mod_name(mod), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129..94d51fe 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -306,6 +306,7 @@ struct rsnd_mod_ops { enum rsnd_mod_type type); int (*id)(struct rsnd_mod *mod); int (*id_sub)(struct rsnd_mod *mod); + int (*id_cmd)(struct rsnd_mod *mod); }; struct rsnd_dai_stream; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: add missing TDM Split mode support for simple-card
by Kuninori Morimoto 04 Dec '18

04 Dec '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f69f452243e4e1 ("ASoC: rsnd: add TDM Split mode support") added TDM Split mode support for rsnd driver. But, it cares audio-graph-card style only. We can't use TDM Split mode on simple-card style now. This patch fixup this issue. Fixes: f69f452243e4e1 ("ASoC: rsnd: add TDM Split mode support") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 51 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 031cd5b..59e250c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1031,6 +1031,49 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { .prepare = rsnd_soc_dai_prepare, }; +static void rsnd_parse_connect_simple(struct rsnd_priv *priv, + struct device_node *dai_np, + int dai_i, int is_play) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai *rdai = rsnd_rdai_get(priv, dai_i); + struct rsnd_dai_stream *io = is_play ? + &rdai->playback : + &rdai->capture; + struct device_node *ssiu_np = rsnd_ssiu_of_node(priv); + struct device_node *np; + int i, j; + + if (!ssiu_np) + return; + + if (!rsnd_io_to_mod_ssi(io)) + return; + + /* + * This driver assumes that it is TDM Split mode + * if it includes ssiu node + */ + for (i = 0;; i++) { + struct device_node *node = is_play ? + of_parse_phandle(dai_np, "playback", i) : + of_parse_phandle(dai_np, "capture", i); + + if (!node) + break; + + j = 0; + for_each_child_of_node(ssiu_np, np) { + if (np == node) { + rsnd_flags_set(io, RSND_STREAM_TDM_SPLIT); + dev_dbg(dev, "%s is part of TDM Split\n", io->name); + } + j++; + } + + } +} + static void rsnd_parse_connect_graph(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct device_node *endpoint) @@ -1246,8 +1289,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i++; } } else { - for_each_child_of_node(dai_node, dai_np) - __rsnd_dai_probe(priv, dai_np, dai_i++); + for_each_child_of_node(dai_node, dai_np) { + __rsnd_dai_probe(priv, dai_np, dai_i); + if (rsnd_is_gen3(priv)) { + rsnd_parse_connect_simple(priv, dai_np, dai_i, 1); + rsnd_parse_connect_simple(priv, dai_np, dai_i, 0); + } + dai_i++; + } } return 0; -- 2.7.4

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[alsa-devel] [PATCH v2 0/5] ASoC: merge audio-graph-scu into audio-graph - step3
by Kuninori Morimoto 04 Dec '18

04 Dec '18

Hi Mark audio-graph card can handle normal sound card, and audio-graph-scu card can handle DPCM sound card. But, we can't use these feature in same time now. This is v2 of step3 patches. I mentioned that step4 patch-set will be based on OF-graph posted patch, but, I could solve it without posted patch. There is no blocker anymore. Thus, I included blocked patch on step3. Kuninori Morimoto (5): ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting ASoC: audio-graph-scu-card: care link / dai count ASoC: audio-graph-scu-card: use cpu/codec pointer on graph_dai_props ASoC: audio-graph-scu-card: care multi DPCM codec_conf ASoC: audio-graph-card: use cpu/codec pointer on graph_dai_props sound/soc/generic/audio-graph-card.c | 57 ++++++---- sound/soc/generic/audio-graph-scu-card.c | 180 +++++++++++++++++++++++-------- sound/soc/generic/simple-card-utils.c | 22 +--- 3 files changed, 173 insertions(+), 86 deletions(-) -- 2.7.4

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