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[PATCH v2] ASoc: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec
by Brent Lu 29 Oct '21

29 Oct '21

Detect the codec variant in probe function and update DAI link accordingly. Also add an new entry in enumeration table for machine driver enumeration. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 52 +++++++++++++++---- .../intel/common/soc-acpi-intel-glk-match.c | 8 +++ 3 files changed, 52 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f693383eb6e3..2dd5ff7e35ce 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -427,6 +427,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_MAX98357A select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 9d75beec09d1..bad3829e52ca 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -18,14 +18,18 @@ #include <sound/soc.h> #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" +#include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" #include "hda_dsp_common.h" /* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ #define GLK_PLAT_CLK_FREQ 19200000 #define RT5682_PLL_FREQ (48000 * 512) -#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define RT5682_DAI_NAME "rt5682-aif1" +#define RT5682S_DAI_NAME "rt5682s-aif1" #define GLK_MAXIM_CODEC_DAI "HiFi" +#define RT5682_DEV0_NAME "i2c-10EC5682:00" +#define RT5682S_DEV0_NAME "i2c-RTL5682:00" #define MAXIM_DEV0_NAME "MX98357A:00" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 @@ -43,6 +47,7 @@ struct glk_card_private { struct snd_soc_jack geminilake_headset; struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int is_rt5682s; }; enum { @@ -139,9 +144,19 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; - int ret; + int pll_id, pll_source, clk_id, ret; + + if (ctx->is_rt5682s) { + pll_id = RT5682S_PLL2; + pll_source = RT5682S_PLL_S_MCLK; + clk_id = RT5682S_SCLK_S_PLL2; + } else { + pll_id = RT5682_PLL1; + pll_source = RT5682_PLL1_S_MCLK; + clk_id = RT5682_SCLK_S_PLL1; + } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); if (ret < 0) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); @@ -149,7 +164,7 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); @@ -344,9 +359,12 @@ SND_SOC_DAILINK_DEF(ssp1_codec, SND_SOC_DAILINK_DEF(ssp2_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(ssp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", - GLK_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682_DEV0_NAME, + RT5682_DAI_NAME))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682s, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682S_DEV0_NAME, + RT5682S_DAI_NAME))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -492,7 +510,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .ops = &geminilake_rt5682_ops, .dpcm_playback = 1, .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec, platform), + SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec_5682, platform), }, { .name = "dmic01", @@ -592,12 +610,28 @@ static int geminilake_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; const char *platform_name; struct snd_soc_card *card; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + /* Detect the headset codec variant */ + if (acpi_dev_present("RTL5682", NULL, -1)) { + /* ALC5682I-VS is detected */ + ctx->is_rt5682s = 1; + + for (i = 0; i < glk_audio_card_rt5682_m98357a.num_links; i++) { + if (strcmp(geminilake_dais[i].name, "SSP2-Codec")) + continue; + + /* update the dai link to use rt5682s codec */ + geminilake_dais[i].codecs = ssp2_codec_5682s; + geminilake_dais[i].num_codecs = ARRAY_SIZE(ssp2_codec_5682s); + break; + } + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = &glk_audio_card_rt5682_m98357a; diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 32fff9389eb3..3900e3dbae30 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -40,6 +40,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_fw_filename = "sof-glk.ri", .sof_tplg_filename = "sof-glk-rt5682.tplg", }, + { + .id = "RTL5682", + .drv_name = "glk_rt5682_max98357a", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "sof-glk.ri", + .sof_tplg_filename = "sof-glk-rt5682.tplg", + }, { .id = "10134242", .drv_name = "glk_cs4242_mx98357a", -- 2.25.1

3 2

[PATCH] ALSA: oxfw: fix functional regression for Mackie Onyx 1640i in v5.14 or later
by Takashi Sakamoto 29 Oct '21

29 Oct '21

A user reports functional regression for Mackie Onyx 1640i that the device generates slow sound with ALSA oxfw driver which supports media clock recovery. Although the device is based on OXFW971 ASIC, it does not transfer isochronous packet with own event frequency as expected. The device seems to adjust event frequency according to events in received isochronous packets in the beginning of packet streaming. This is unknown quirk. This commit fixes the regression to turn the recovery off in driver side. As a result, nominal frequency is used in duplex packet streaming between device and driver. For stability of sampling rate in events of transferred isochronous packet, 4,000 isochronous packets are skipped in the beginning of packet streaming. Reference: https://github.com/takaswie/snd-firewire-improve/issues/38 Fixes: 029ffc429440 ("ALSA: oxfw: perform sequence replay for media clock recovery") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-stream.c | 7 ++++++- sound/firewire/oxfw/oxfw.c | 8 ++++++++ sound/firewire/oxfw/oxfw.h | 5 +++++ 3 files changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index fff18b5d4e05..f4a702def397 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -9,7 +9,7 @@ #include <linux/delay.h> #define AVC_GENERIC_FRAME_MAXIMUM_BYTES 512 -#define READY_TIMEOUT_MS 200 +#define READY_TIMEOUT_MS 600 /* * According to datasheet of Oxford Semiconductor: @@ -367,6 +367,11 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw) // Just after changing sampling transfer frequency, many cycles are // skipped for packet transmission. tx_init_skip_cycles = 400; + } else if (oxfw->quirks & SND_OXFW_QUIRK_VOLUNTARY_RECOVERY) { + // It takes a bit time for target device to adjust event frequency + // according to nominal event frequency in isochronous packets from + // ALSA oxfw driver. + tx_init_skip_cycles = 4000; } else { replay_seq = true; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index daf731364695..b496f87841ae 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -25,6 +25,7 @@ #define MODEL_SATELLITE 0x00200f #define MODEL_SCS1M 0x001000 #define MODEL_DUET_FW 0x01dddd +#define MODEL_ONYX_1640I 0x001640 #define SPECIFIER_1394TA 0x00a02d #define VERSION_AVC 0x010001 @@ -192,6 +193,13 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id // OXFW971-based models may transfer events by blocking method. if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + + if (model == MODEL_ONYX_1640I) { + //Unless receiving packets without NOINFO packet, the device transfers + //mostly half of events in packets than expected. + oxfw->quirks |= SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET | + SND_OXFW_QUIRK_VOLUNTARY_RECOVERY; + } } return 0; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index c13034f6c2ca..d728e451a25c 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -47,6 +47,11 @@ enum snd_oxfw_quirk { // the device to process audio data even if the value is invalid in a point of // IEC 61883-1/6. SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET = 0x10, + // Loud Technologies Mackie Onyx 1640i seems to configure OXFW971 ASIC so that it decides + // event frequency according to events in received isochronous packets. The device looks to + // performs media clock recovery voluntarily. In the recovery, the packets with NO_INFO + // are ignored, thus driver should transfer packets with timestamp. + SND_OXFW_QUIRK_VOLUNTARY_RECOVERY = 0x20, }; /* This is an arbitrary number for convinience. */ -- 2.30.2

2 4

[PATCH] ALSA: firewire-motu: remove TODO for interaction with userspace about control message
by Takashi Sakamoto 29 Oct '21

29 Oct '21

Now UAPI of ALSA firewire stack got enough functions to interact with userspace for protocol of MOTU FireWire series. Let's remove the relevant TODO. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 3ea91e281147..2fb52f481d12 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -424,8 +424,6 @@ static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, if (p->midi_ports) write_midi_messages(s, buf, data_blocks); - // TODO: how to interact control messages between userspace? - write_sph(p->cache, buf, data_blocks, s->data_block_quadlets); } -- 2.30.2

2 1

[PATCH 0/3] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 29 Oct '21

29 Oct '21

This patch series is to make lpass variant independent pin control functions common and to add lpass sc7280 pincontrol support. It also includes dt-bindings for lpass sc7280 lpi compatible. Srinivasa Rao Mandadapu (3): pinctrl: qcom: Update lpass variant independent functions as generic dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl compatible pinctrl: qcom: Add SC7280 lpass pin configuration .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 4 +- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 57 +++++++++++++++++++--- 2 files changed, 52 insertions(+), 9 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 11

[PATCH RESEND] ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_DMIC
by Julian Braha 28 Oct '21

28 Oct '21

When SND_SOC_AMD_RENOIR_MACH or SND_SOC_AMD_RV_RT5682_MACH are selected, and GPIOLIB is not selected, Kbuild gives the following warnings, respectively: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y] and WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RV_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_ACP3x [=y] && I2C [=y] && CROS_EC [=y] This is because SND_SOC_DMIC and SND_SOC_MAX98357A are selected by SND_SOC_AMD_RV_RT5682_MACH and SND_SOC_AMD_RENOIR_MACH, respectively. However, neither of the selectors depend on or select GPIOLIB, despite their selectees depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/amd/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 56e7c079deac..2c6af3f8f296 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -38,7 +38,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select I2C_CROS_EC_TUNNEL select SND_SOC_RT1015 select SND_SOC_RT1015P - depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC + depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC && GPIOLIB help This option enables machine driver for RT5682 and MAX9835. @@ -51,7 +51,7 @@ config SND_SOC_AMD_RENOIR config SND_SOC_AMD_RENOIR_MACH tristate "AMD Renoir support for DMIC" select SND_SOC_DMIC - depends on SND_SOC_AMD_RENOIR + depends on SND_SOC_AMD_RENOIR && GPIOLIB help This option enables machine driver for DMIC -- 2.30.2

3 5

[PATCH] ASoC: Intel: add sof-nau8825 machine driver
by Mac Chiang 28 Oct '21

28 Oct '21

From: David Lin <CTLIN0(a)nuvoton.com> The machine driver is a generic machine driver for SOF with nau8825 codec w or w/o speaker additionally. Depending on the SOC HDMI, DMIC, Bluetooth offload support are added dynamically. Only add information related to SOF since the machine driver was only tested with SOF. There are currently 4 i2s machine variants of ADL. This supports the headphone NUA8825(SSP0) alone or with smart or dumb speakers. Board 2,3,4 use SSP2 for Bluetooth offload support except board 1. Board 1 : NAU8825 + RT1019P(SSP2) Board 2 : NAU8825 + MAX98373(SSP1) Board 3 : NAU8825 + MAX98360A(SSP1) Board 4 : NAU8825 Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Co-developed-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 19 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_nau8825.c | 683 ++++++++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 35 + 4 files changed, 739 insertions(+) create mode 100644 sound/soc/intel/boards/sof_nau8825.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5a225a4ce604..0944934119f5 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -526,6 +526,25 @@ config SND_SOC_INTEL_SOF_ES8336_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_SOF_NAU8825_MACH + tristate "SOF with nau8825 codec in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ + (MFD_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_NAU8825 + select SND_SOC_RT1019 + select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_HDA_DSP_COMMON + select SND_SOC_INTEL_SOF_MAXIM_COMMON + help + This adds support for ASoC machine driver for SOF platforms + with nau8825 codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9ee8ed864f5d..3ea273d27168 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,6 +22,7 @@ snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-sof_rt5682-objs := sof_rt5682.o sof_realtek_common.o snd-soc-sof_cs42l42-objs := sof_cs42l42.o snd-soc-sof_es8336-objs := sof_es8336.o +snd-soc-sof_nau8825-objs := sof_nau8825.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o @@ -44,6 +45,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c new file mode 100644 index 000000000000..04f676922018 --- /dev/null +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -0,0 +1,683 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2021 Intel Corporation. +// Copyright(c) 2021 Nuvoton Corporation. + +/* + * Intel SOF Machine Driver with Nuvoton headphone codec NAU8825 + * and speaker codec RT1019P MAX98360a or MAX98373 + */ +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/dmi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> +#include "../../codecs/nau8825.h" +#include "../../codecs/hdac_hdmi.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" +#include "sof_realtek_common.h" +#include "sof_maxim_common.h" + +#define NAME_SIZE 32 + +#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) +#define SOF_SPEAKER_AMP_PRESENT BIT(3) +#define SOF_NAU8825_SSP_AMP_SHIFT 4 +#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) +#define SOF_NAU8825_SSP_AMP(quirk) \ + (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) +#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 +#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) +#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ + (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) + +/* BT audio offload: reserve 3 bits for future */ +#define SOF_BT_OFFLOAD_SSP_SHIFT 10 +#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) +#define SOF_BT_OFFLOAD_SSP(quirk) \ + (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) +#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) +#define SOF_RT1019P_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) + +static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +static struct snd_soc_jack sof_hdmi[3]; + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct clk *mclk; + struct snd_soc_jack sof_headset; + struct list_head hdmi_pcm_list; + bool common_hdmi_codec_drv; +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + struct snd_soc_jack *jack; + int ret; + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->sof_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static void sof_nau8825_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + +static int sof_nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int clk_freq, ret; + + clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ + + if (clk_freq <= 0) { + dev_err(rtd->dev, "get bclk freq failed: %d\n", clk_freq); + return -EINVAL; + } + + /* Configure clock for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_BLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK clock %d\n", ret); + return ret; + } + + /* Configure pll for codec */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_freq, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK: %d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops sof_nau8825_ops = { + .hw_params = sof_nau8825_hw_params, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + struct snd_soc_dapm_context *dapm = &card->dapm; + char jack_name[NAME_SIZE]; + struct sof_hdmi_pcm *pcm; + int err; + int i = 0; + + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, + head); + component = pcm->codec_dai->component; + return hda_dsp_hdmi_build_controls(card, component); + } + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &sof_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &sof_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { + /* Disable Left and Right Spk pin after boot */ + snd_soc_dapm_disable_pin(dapm, "Left Spk"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_kcontrol_new speaker_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_widget speaker_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, +}; + +static const struct snd_soc_dapm_route speaker_map[] = { + /* speaker */ + { "Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, speaker_widgets, + ARRAY_SIZE(speaker_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, speaker_controls, + ARRAY_SIZE(speaker_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map, + ARRAY_SIZE(speaker_map)); + + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for nau8825 codec */ +static struct snd_soc_card sof_audio_card_nau8825 = { + .name = "nau8825", /* the sof- prefix is added by the core */ + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static struct snd_soc_dai_link_component nau8825_component[] = { + { + .name = "i2c-10508825:00", + .dai_name = "nau8825-hifi", + } +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component rt1019p_component[] = { + { + .name = "RTL1019:00", + .dai_name = "HiFi", + } +}; + +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = nau8825_component; + links[id].num_codecs = ARRAY_SIZE(nau8825_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_nau8825_codec_init; + links[id].exit = sof_nau8825_codec_exit; + links[id].ops = &sof_nau8825_ops; + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + /* speaker amp */ + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_amp); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + if (sof_nau8825_quirk & SOF_RT1019P_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1019p_component; + links[id].num_codecs = ARRAY_SIZE(rt1019p_component); + links[id].init = speaker_codec_init; + } else if (sof_nau8825_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98373_ops; + /* feedback stream */ + links[id].dpcm_capture = 1; + } else if (sof_nau8825_quirk & + SOF_MAX98360A_SPEAKER_AMP_PRESENT) { + max_98360a_dai_link(&links[id]); + } else { + max_98360a_dai_link(&links[id]); + } + + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_amp); + if (!links[id].cpus->dai_name) + goto devm_err; + id++; + } + + /* BT audio offload */ + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", port); + if (!links[id].cpus->dai_name) + goto devm_err; + links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); + if (!links[id].name) + goto devm_err; + links[id].codecs = dummy_component; + links[id].num_codecs = ARRAY_SIZE(dummy_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].num_cpus = 1; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_amp, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; + + mach = pdev->dev.platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_nau8825_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + + dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); + + /* default number of DMIC DAI's */ + dmic_be_num = 2; + hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> + SOF_NAU8825_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; + + ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> + SOF_NAU8825_SSP_AMP_SHIFT; + + ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_nau8825.num_links = 1 + dmic_be_num + hdmi_num; + + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) + sof_audio_card_nau8825.num_links++; + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + max_98373_set_codec_conf(&sof_audio_card_nau8825); + + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + sof_audio_card_nau8825.num_links++; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, + dmic_be_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_nau8825.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_nau8825.dev = &pdev->dev; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_nau8825, + mach->mach_params.platform); + if (ret) + return ret; + + ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; + + snd_soc_card_set_drvdata(&sof_audio_card_nau8825, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_nau8825); +} + +static const struct platform_device_id board_ids[] = { + { + .name = "sof_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { + .name = "adl_rt1019p_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019P_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(2) | + SOF_NAU8825_NUM_HDMIDEV(4)), + }, + { + .name = "adl_max98373_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + /* The limitation of length of char array, shorten the name */ + .name = "adl_mx98360a_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { } +}; +MODULE_DEVICE_TABLE(platform, board_ids); + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .driver = { + .name = "sof_nau8825", + .pm = &snd_soc_pm_ops, + }, + .id_table = board_ids, +}; +module_platform_driver(sof_audio) + +/* Module information */ +MODULE_DESCRIPTION("SOF Audio Machine driver for NAU8825"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index f5b21a95d222..98dc41b682cf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -285,6 +285,11 @@ static const struct snd_soc_acpi_codecs adl_max98360a_amp = { .codecs = {"MX98360A"} }; +static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -310,6 +315,36 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", }, + { + .id = "10508825", + .drv_name = "adl_rt1019p_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_rt1019p_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_max98373_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98373_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_mx98360a_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "sof_nau8825", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-nau8825.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.20.1

2 1

[PATCH] ASoc: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec
by Brent Lu 28 Oct '21

28 Oct '21

Detect the codec variant in probe function and update DAI link accordingly. Also add an new entry in enumeration table for machine driver enumeration. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 53 +++++++++++++++---- .../intel/common/soc-acpi-intel-glk-match.c | 9 ++++ 3 files changed, 54 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f693383eb6e3..2dd5ff7e35ce 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -427,6 +427,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_MAX98357A select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 9d75beec09d1..058aa7cb899a 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -18,14 +18,18 @@ #include <sound/soc.h> #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" +#include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" #include "hda_dsp_common.h" /* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ #define GLK_PLAT_CLK_FREQ 19200000 #define RT5682_PLL_FREQ (48000 * 512) -#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define RT5682_DAI_NAME "rt5682-aif1" +#define RT5682S_DAI_NAME "rt5682s-aif1" #define GLK_MAXIM_CODEC_DAI "HiFi" +#define RT5682_DEV0_NAME "i2c-10EC5682:00" +#define RT5682S_DEV0_NAME "i2c-RTL5682:00" #define MAXIM_DEV0_NAME "MX98357A:00" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 @@ -43,6 +47,7 @@ struct glk_card_private { struct snd_soc_jack geminilake_headset; struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int is_rt5682s; }; enum { @@ -139,9 +144,19 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; - int ret; + int pll_id, pll_source, clk_id, ret; + + if (ctx->is_rt5682s) { + pll_id = RT5682S_PLL2; + pll_source = RT5682S_PLL_S_MCLK; + clk_id = RT5682S_SCLK_S_PLL2; + } else { + pll_id = RT5682_PLL1; + pll_source = RT5682_PLL1_S_MCLK; + clk_id = RT5682_SCLK_S_PLL1; + } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); if (ret < 0) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); @@ -149,7 +164,7 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); @@ -344,9 +359,12 @@ SND_SOC_DAILINK_DEF(ssp1_codec, SND_SOC_DAILINK_DEF(ssp2_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(ssp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", - GLK_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682_DEV0_NAME, + RT5682_DAI_NAME))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682s, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682S_DEV0_NAME, + RT5682S_DAI_NAME))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -492,7 +510,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .ops = &geminilake_rt5682_ops, .dpcm_playback = 1, .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec, platform), + SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec_5682, platform), }, { .name = "dmic01", @@ -592,12 +610,29 @@ static int geminilake_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; const char *platform_name; struct snd_soc_card *card; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + /* Detect the headset codec variant */ + if (acpi_dev_present("RTL5682", NULL, -1)) { + /* ALC5682I-VS is detected */ + ctx->is_rt5682s = 1; + + for (i = 0; i < glk_audio_card_rt5682_m98357a.num_links; i++) { + if (strcmp(geminilake_dais[i].name, "SSP2-Codec")) + continue; + + /* update the dai link to use rt5682s codec */ + geminilake_dais[i].codecs = ssp2_codec_5682s; + geminilake_dais[i].num_codecs = ARRAY_SIZE(ssp2_codec_5682s); + break; + } + } else + ctx->is_rt5682s = 0; + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = &glk_audio_card_rt5682_m98357a; diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 32fff9389eb3..4de4add74443 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -40,6 +40,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_fw_filename = "sof-glk.ri", .sof_tplg_filename = "sof-glk-rt5682.tplg", }, + { + .id = "RTL5682", + .drv_name = "glk_rt5682_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "sof-glk.ri", + .sof_tplg_filename = "sof-glk-rt5682.tplg", + }, { .id = "10134242", .drv_name = "glk_cs4242_mx98357a", -- 2.25.1

3 2

[PATCH RFC] ALSA: control: Use xarray for faster lookups
by Takashi Iwai 28 Oct '21

28 Oct '21

The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray in addition to the linked list for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load(). When nothing hits, we look through the full list entries as a fallback. So, the inquiry for a non-existing element would take (even slightly longer) time with this patch. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is not really meant to be merged now, but just shown as a reference implementation, as the issue doesn't surface unless you have a large number of elements and query the late element frequently. In my test, 1,000,000 queries of the 1000th element took over 12 seconds, while the patched kernel took 0.2 second. But for the most cases (at most 20 or 30 elements), it doesn't matter much. include/sound/core.h | 3 + sound/core/control.c | 142 +++++++++++++++++++++++++++++++++---------- sound/core/init.c | 2 + 3 files changed, 115 insertions(+), 32 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index b7e9b58d3c78..e8bb6d943f3f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/xarray.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -103,6 +104,8 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ + struct xarray ctl_numids; /* hash table for numids */ + struct xarray ctl_hash; /* hash table for ctl id matching */ struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..d0253cc4b22d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,78 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i, err; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + err = xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL); + /* ignore errors - it's a best effort */ + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +480,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +553,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +586,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,9 +727,11 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } @@ -665,15 +752,9 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; + return xa_load(&card->ctl_numids, numid); } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +780,15 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2270,11 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index ac335f5906c6..a1979fe27d53 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -282,6 +282,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- 2.31.1

1 0

[PATCH] ASoC: qdsp6: audioreach: Fix clang -Wimplicit-fallthrough
by Nathan Chancellor 28 Oct '21

28 Oct '21

Clang warns: sound/soc/qcom/qdsp6/topology.c:465:3: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough] default: ^ sound/soc/qcom/qdsp6/topology.c:465:3: note: insert 'break;' to avoid fall-through default: ^ break; 1 warning generated. Clang is a little more pedantic than GCC, which permits implicit fallthroughs to cases that contain just break or return. Clang's version is more in line with the kernel's own stance in deprecated.rst, which states that all switch/case blocks must end in either break, fallthrough, continue, goto, or return. Add the missing break to fix the warning. Link: https://github.com/ClangBuiltLinux/linux/issues/1495 Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/qcom/qdsp6/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c index f31895379925..bd649c232a06 100644 --- a/sound/soc/qcom/qdsp6/topology.c +++ b/sound/soc/qcom/qdsp6/topology.c @@ -461,7 +461,7 @@ static struct audioreach_module *audioreach_parse_common_tokens(struct q6apm *ap break; case AR_TKN_U32_MODULE_DST_IN_PORT_ID: dst_mod_ip_port_id = le32_to_cpu(mod_elem->value); - + break; default: break; base-commit: 1dcc81d95b910dc404c40ff6101bfa2520a3528e -- 2.33.1.637.gf443b226ca

2 1

[PATCH] dt-bindings: sound: wlf,wm8903: Convert txt bindings to yaml
by David Heidelberg 28 Oct '21

28 Oct '21

Convert the Wolfson WM8903 Ultra-Low Power Stereo CODEC Device Tree binding documentation to json-schema. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- .../devicetree/bindings/sound/wlf,wm8903.yaml | 116 ++++++++++++++++++ .../devicetree/bindings/sound/wm8903.txt | 82 ------------- 2 files changed, 116 insertions(+), 82 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8903.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000000..afdcf9002b60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/wlf,wm8903.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: WM8903 audio codec + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + +maintainers: + - Mark Brown <broonie(a)opensource.wolfsonmicro.com> + +properties: + compatible: + const: wlf,wm8903 + + reg: + maxItems: 1 + + gpio-controller: true + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + micdet-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + description: Default register value for R6 (Mic Bias). + + micdet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 100 + description: The debounce delay for microphone detection in mS. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt deleted file mode 100644 index 6371c2434afe..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ /dev/null @@ -1,82 +0,0 @@ -WM8903 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8903" - - - reg : the I2C address of the device. - - - gpio-controller : Indicates this device is a GPIO controller. - - - #gpio-cells : Should be two. The first cell is the pin number and the - second cell is used to specify optional parameters (currently unused). - -Optional properties: - - - interrupts : The interrupt line the codec is connected to. - - - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the - default is 0. - - - micdet-delay : The debounce delay for microphone detection in mS. If - absent, the default is 100. - - - gpio-cfg : A list of GPIO configuration register values. The list must - be 5 entries long. If absent, no configuration of these registers is - performed. If any entry has the value 0xffffffff, that GPIO's - configuration will not be modified. - - - AVDD-supply : Analog power supply regulator on the AVDD pin. - - - CPVDD-supply : Charge pump supply regulator on the CPVDD pin. - - - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin. - - - DCVDD-supply : Digital core supply regulator for the DCVDD pin. - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - -Example: - -wm8903: codec@1a { - compatible = "wlf,wm8903"; - reg = <0x1a>; - interrupts = < 347 >; - - AVDD-supply = <&fooreg_a>; - CPVDD-supply = <&fooreg_b>; - DBVDD-supply = <&fooreg_c>; - DCVDC-supply = <&fooreg_d>; - - gpio-controller; - #gpio-cells = <2>; - - micdet-cfg = <0>; - micdet-delay = <100>; - gpio-cfg = < - 0x0600 /* DMIC_LR, output */ - 0x0680 /* DMIC_DAT, input */ - 0x0000 /* GPIO, output, low */ - 0x0200 /* Interrupt, output */ - 0x01a0 /* BCLK, input, active high */ - >; -}; -- 2.33.0

1 0

[PATCH 1/2] alsaloop: Renamed field capt_pitch to ctl_pitch
by Pavel Hofman 28 Oct '21

28 Oct '21

Renaming capt_pitch to ctl_pitch to respect the naming style of snd_ctl_elem_value_t fields in loopback_handle. Also making the variable name suitable for planned support of playback pitch. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.h | 2 +- alsaloop/pcmjob.c | 16 ++++++++-------- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 7a98ef3..a5539ef 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -122,7 +122,7 @@ struct loopback_handle { unsigned int ctl_pollfd_count; snd_ctl_elem_value_t *ctl_notify; snd_ctl_elem_value_t *ctl_rate_shift; - snd_ctl_elem_value_t *capt_pitch; + snd_ctl_elem_value_t *ctl_pitch; snd_ctl_elem_value_t *ctl_active; snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 13fa595..7b47c40 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1063,9 +1063,9 @@ static int set_rate_shift(struct loopback_handle *lhandle, double pitch) if (lhandle->ctl_rate_shift) { snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); - } else if (lhandle->capt_pitch) { - snd_ctl_elem_value_set_integer(lhandle->capt_pitch, 0, (1 / pitch) * 1000000); - err = snd_ctl_elem_write(lhandle->ctl, lhandle->capt_pitch); + } else if (lhandle->ctl_pitch) { + snd_ctl_elem_value_set_integer(lhandle->ctl_pitch, 0, (1 / pitch) * 1000000); + err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_pitch); } else { return 0; } @@ -1252,7 +1252,7 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", &lhandle->ctl_rate_shift); openctl_elem(lhandle, device, subdevice, "Capture Pitch 1000000", - &lhandle->capt_pitch); + &lhandle->ctl_pitch); set_rate_shift(lhandle, 1); openctl_elem(lhandle, device, subdevice, "PCM Slave Active", &lhandle->ctl_active); @@ -1338,9 +1338,9 @@ static int closeit(struct loopback_handle *lhandle) if (lhandle->ctl_rate_shift) snd_ctl_elem_value_free(lhandle->ctl_rate_shift); lhandle->ctl_rate_shift = NULL; - if (lhandle->capt_pitch) - snd_ctl_elem_value_free(lhandle->capt_pitch); - lhandle->capt_pitch = NULL; + if (lhandle->ctl_pitch) + snd_ctl_elem_value_free(lhandle->ctl_pitch); + lhandle->ctl_pitch = NULL; if (lhandle->ctl) err = snd_ctl_close(lhandle->ctl); lhandle->ctl = NULL; @@ -1386,7 +1386,7 @@ int pcmjob_init(struct loopback *loop) snprintf(id, sizeof(id), "%s/%s", loop->play->id, loop->capt->id); id[sizeof(id)-1] = '\0'; loop->id = strdup(id); - if (loop->sync == SYNC_TYPE_AUTO && (loop->capt->ctl_rate_shift || loop->capt->capt_pitch)) + if (loop->sync == SYNC_TYPE_AUTO && (loop->capt->ctl_rate_shift || loop->capt->ctl_pitch)) loop->sync = SYNC_TYPE_CAPTRATESHIFT; if (loop->sync == SYNC_TYPE_AUTO && loop->play->ctl_rate_shift) loop->sync = SYNC_TYPE_PLAYRATESHIFT; -- 2.25.1

2 2

[PATCH alsa-utils] alsactl: Add --quiet option to suppress alsa-lib error messages
by Takashi Iwai 28 Oct '21

28 Oct '21

alsactl prints error messages from alsa-lib as is, and it's rather annoying to see the error messages like alsactl[xxx]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HDA Intel PCH...) alsactl[xxx]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 that are recorded in the syslog at each boot. This patch adds --quiet (or -q) option to suppress those error messages from alsa-lib, and applies to the systemd services as default. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/alsa-restore.service.in | 4 ++-- alsactl/alsa-state.service.in | 4 ++-- alsactl/alsactl.1 | 4 ++++ alsactl/alsactl.c | 5 +++++ alsactl/alsactl.h | 2 ++ alsactl/utils.c | 2 ++ 6 files changed, 17 insertions(+), 4 deletions(-) diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index 80fd5fd48203..06bc21c355de 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -11,5 +11,5 @@ ConditionPathExistsGlob=/dev/snd/control* [Service] Type=oneshot RemainAfterExit=true -ExecStart=-@sbindir@/alsactl restore -ExecStop=-@sbindir@/alsactl store +ExecStart=-@sbindir@/alsactl -q restore +ExecStop=-@sbindir@/alsactl -q store diff --git a/alsactl/alsa-state.service.in b/alsactl/alsa-state.service.in index 5a8fe5eeed7e..6b6e704ad8e5 100644 --- a/alsactl/alsa-state.service.in +++ b/alsactl/alsa-state.service.in @@ -9,5 +9,5 @@ ConditionPathExists=@daemonswitch@ [Service] Type=simple -ExecStart=-@sbindir@/alsactl -s -n 19 -c rdaemon -ExecStop=-@sbindir@/alsactl -s kill save_and_quit +ExecStart=-@sbindir@/alsactl -q -s -n 19 -c rdaemon +ExecStop=-@sbindir@/alsactl -q -s kill save_and_quit diff --git a/alsactl/alsactl.1 b/alsactl/alsactl.1 index 8296663a7c2d..0a0b4597a137 100644 --- a/alsactl/alsactl.1 +++ b/alsactl/alsactl.1 @@ -94,6 +94,10 @@ Use debug mode: a bit more verbose. \fI\-v, \-\-version\fP Print alsactl version number. +.TP +\fI\-q, \-\-quiet\fP +Suppress error messages from alsa-lib. + .TP \fI\-f, \-\-file\fP Select the configuration file to use. The default is /var/lib/alsa/asound.state. diff --git a/alsactl/alsactl.c b/alsactl/alsactl.c index 05738fb6b35f..e3bef2c69558 100644 --- a/alsactl/alsactl.c +++ b/alsactl/alsactl.c @@ -52,6 +52,7 @@ int use_syslog = 0; char *command; char *statefile = NULL; char *lockfile = SYS_LOCKFILE; +int verbose_error = 1; #define TITLE 0x0100 #define HEADER 0x0200 @@ -74,6 +75,7 @@ static struct arg args[] = { { 'h', "help", "this help" }, { 'd', "debug", "debug mode" }, { 'v', "version", "print version of this program" }, +{ 'q', "quiet", "suppress errors from alsa-lib" }, { HEADER, NULL, "Available state options:" }, { FILEARG | 'f', "file", "configuration file (default " SYS_ASOUNDRC ")" }, { FILEARG | 'a', "config-dir", "boot / hotplug configuration directory (default " SYS_ASOUND_DIR ")" }, @@ -370,6 +372,9 @@ int main(int argc, char *argv[]) printf("alsactl version " SND_UTIL_VERSION_STR "\n"); res = EXIT_SUCCESS; goto out; + case 'q': + verbose_error = 0; + break; case '?': // error msg already printed help(); res = EXIT_FAILURE; diff --git a/alsactl/alsactl.h b/alsactl/alsactl.h index bbdf6c88baeb..f9c25931c924 100644 --- a/alsactl/alsactl.h +++ b/alsactl/alsactl.h @@ -83,3 +83,5 @@ static inline int hextodigit(int c) } #define ARRAY_SIZE(a) (sizeof (a) / sizeof (a)[0]) + +extern int verbose_error; diff --git a/alsactl/utils.c b/alsactl/utils.c index a50797259f21..dfb06e1a4a15 100644 --- a/alsactl/utils.c +++ b/alsactl/utils.c @@ -182,6 +182,8 @@ void error_handler(const char *file, int line, const char *function, int err, co char buf[2048]; va_list arg; + if (!verbose_error) + return; va_start(arg, fmt); vsnprintf(buf, sizeof(buf), fmt, arg); va_end(arg); -- 2.31.1

2 3

[PATCH 0/3] Make genaral and simple for new sof machine driver
by David Lin 28 Oct '21

28 Oct '21

The series of features will make general and simple for new sof machine driver. David Lin (3): ASoC: nau8825: add set_jack coponment support ASoC: nau8825: add disable jack detection support ASoC: nau8825: add clock management for power saving sound/soc/codecs/nau8825.c | 49 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-)

4 8

[PATCH] ALSA: hda/realtek: Add a quirk for HP OMEN 15 mute LED
by Takashi Iwai 28 Oct '21

28 Oct '21

HP OMEN 15 laptop requires the quirk to fiddle with COEF 0x0b bit 2 for toggling the mute LED. It's already implemented for other HP laptops, and we just need to add a proper fixup entry. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214735 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e82d756ab850..6322fac9e694 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8634,6 +8634,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.31.1

1 0

[PATCH v4 0/3] Update SoundWire RX and TX cgcr register control
by Srinivasa Rao Mandadapu 28 Oct '21

28 Oct '21

This patch series is to add v1.6.0 compatable name for qcom soundwire driver and to update soundwire RX and TX cgcr register control. Changes Since V3: -- Add v1.6.0 compatable name and soundwire data structure. -- Change macro define name properly. -- Update bindings for new property. -- Change commit message description. -- Change signedoff by sequence. Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. Srinivasa Rao Mandadapu (3): ASoC: qcom: soundwire: Disable soundwire rxtx cgcr hardware control dt-bindings: soundwire: qcom: Add bindings for RX and TX cgcr register control soundwire: qcom: Add compatible name for v1.6.0 .../devicetree/bindings/soundwire/qcom,sdw.txt | 9 +++++++++ drivers/soundwire/qcom.c | 20 ++++++++++++++++++++ 2 files changed, 29 insertions(+) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 7

[PATCH] ASoC: amd: acp: fix Kconfig dependencies
by Arnd Bergmann 27 Oct '21

27 Oct '21

From: Arnd Bergmann <arnd(a)arndb.de> The CONFIG_SND_SOC_AMD_MACH_COMMON has some dependencies that are not checked by the symbols that select it: WARNING: unmet direct dependencies detected for SND_SOC_AMD_MACH_COMMON Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && X86 && PCI [=y] && I2C [=n] Selected by [m]: - SND_SOC_AMD_LEGACY_MACH [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] - SND_SOC_AMD_SOF_MACH [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] WARNING: unmet direct dependencies detected for SND_SOC_AMD_MACH_COMMON Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && X86 && PCI [=n] && I2C [=m] Selected by [m]: - SND_SOC_AMD_LEGACY_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] Make this more consistent by adding the same checks everywhere. Fixes: 9d8a7be88b33 ("ASoC: amd: acp: Add legacy sound card support for Chrome audio") Fixes: 9f84940f5004 ("ASoC: amd: acp: Add SOF audio support on Chrome board") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/acp/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 98ec18791d35..737135357ee2 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -28,24 +28,26 @@ config SND_AMD_ASOC_RENOIR config SND_SOC_AMD_MACH_COMMON tristate + depends on X86 && PCI && I2C select CLK_FIXED_FCH select SND_SOC_RT5682_I2C select SND_SOC_DMIC select SND_SOC_RT1019 select SND_SOC_MAX98357A select SND_SOC_RT5682S - depends on X86 && PCI && I2C help This option enables common Machine driver module for ACP. config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" + depends on X86 && PCI && I2C select SND_SOC_AMD_MACH_COMMON help This option enables legacy sound card support for ACP audio. config SND_SOC_AMD_SOF_MACH tristate "AMD SOF Machine Driver Support" + depends on X86 && PCI && I2C select SND_SOC_AMD_MACH_COMMON help This option enables SOF sound card support for ACP audio. -- 2.29.2

2 1

[PATCH v2 1/1] ASoC: meson: t9015: Add missing AVDD-supply property
by Alexander Stein 27 Oct '21

27 Oct '21

Fixes the schema check warning "audio-controller@32000: 'AVDD-supply' do not match any of the regexes: 'pinctrl-[0-9]+'" Fixes: 5c36abcd2621 ("ASoC: meson: add t9015 internal codec binding documentation") Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Alexander Stein <alexander.stein(a)mailbox.org> --- Thanks all for their review. Here is v2 with the following changes: v2: * Separated from other meson DT patches (2/4 - 4/4) * Adjusted recipients * Added missing property in example * Adjusted commit message subject * Added Reviewed-by: Jerome Brunet tag Documentation/devicetree/bindings/sound/amlogic,t9015.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml index c7613ea728d4..db7b04da0b39 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -34,6 +34,10 @@ properties: resets: maxItems: 1 + AVDD-supply: + description: + Analogue power supply. + required: - "#sound-dai-cells" - compatible @@ -41,6 +45,7 @@ required: - clocks - clock-names - resets + - AVDD-supply additionalProperties: false @@ -56,4 +61,5 @@ examples: clocks = <&clkc CLKID_AUDIO_CODEC>; clock-names = "pclk"; resets = <&reset RESET_AUDIO_CODEC>; + AVDD-supply = <&vddao_1v8>; }; -- 2.33.1

2 1

[PATCH] ASoC: Intel: soc-acpi: add entry for ESSX8336 on JSL
by Bard Liao 27 Oct '21

27 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We have configurations for this codec on APL, GLK and TGL, somehow the information that some designs rely on JasperLake was not shared. BugLink: https://github.com/thesofproject/linux/issues/3210 Fixes: 790049fb6623 ('ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 20fd9dcc74af..e7d22e397299 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -105,6 +105,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .id = "ESSX8336", + .drv_name = "sof-essx8336", + .sof_fw_filename = "sof-jsl.ri", + .sof_tplg_filename = "sof-jsl-es8336.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.17.1

2 1

[RESEND,v6 1/2] ASoC: dt-bindings: max98520: add initial bindings
by George Song 27 Oct '21

27 Oct '21

add initial bindings for max98520 audio amplifier Signed-off-by: George Song <george.song(a)maximintegrated.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/maxim,max98520.yaml | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..b6509cb2c8e0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; + -- 2.25.1

2 2

[PATCH -next] ASoC: amd: acp: Fix return value check in acp_machine_select()
by Yang Yingliang 27 Oct '21

27 Oct '21

In case of error, platform_device_register_data() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e646b51f5dd5 ("ASoC: amd: acp: Add callback for machine driver on ACP") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/amd/acp/acp-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 75003d0a41e3..65a809e2c29f 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -81,7 +81,7 @@ int acp_machine_select(struct acp_dev_data *adata) adata->mach_dev = platform_device_register_data(adata->dev, mach->drv_name, PLATFORM_DEVID_NONE, mach, size); - if (!adata->mach_dev) + if (IS_ERR(adata->mach_dev)) dev_warn(adata->dev, "Unable to register Machine device\n"); return 0; -- 2.25.1

2 1

[PATCH 0/8] ASoC: minor cleanup of warnings
by Pierre-Louis Bossart 27 Oct '21

27 Oct '21

Sparse, make W=1 and cppcheck all report minor warnings. The only functional change is in patch7 where the error code is now returned to the caller. Pierre-Louis Bossart (8): ASoC: topology: handle endianness warning ASoC: rt5682s: use 'static' qualifier ASoC: nau8821: fix kernel-doc ASoC: nau8821: clarify out-of-bounds check ASoC: mediatek: remove unnecessary initialization ASoC: mediatek: mt8195: rename shadowed array ASoC: mediatek: mt8195: fix return value ASoC: rockchip: i2s_tdm: improve return value handling sound/soc/codecs/nau8821.c | 6 ++++-- sound/soc/codecs/rt5682s.c | 6 +++--- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/soc-topology.c | 2 +- 7 files changed, 13 insertions(+), 11 deletions(-) -- 2.25.1

2 9

[PATCH] ASoC: rt5682-i2c: Use devm_clk_get_optional for optional clock
by AngeloGioacchino Del Regno 27 Oct '21

27 Oct '21

The mclk clock is optional, but it's currently using devm_clk_get: simplify the handling by using devm_clk_get_optional instead. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/codecs/rt5682-i2c.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index a30e42932cf7..983347b65127 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -280,14 +280,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, #ifdef CONFIG_COMMON_CLK /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(&i2c->dev, "mclk"); - if (IS_ERR(rt5682->mclk)) { - if (PTR_ERR(rt5682->mclk) != -ENOENT) { - ret = PTR_ERR(rt5682->mclk); - return ret; - } - rt5682->mclk = NULL; - } + rt5682->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) + return PTR_ERR(rt5682->mclk); /* Register CCF DAI clock control */ ret = rt5682_register_dai_clks(rt5682); -- 2.33.0

2 1

[PATCH 0/3] Make genaral and simple for new sof machine driver
by David Lin 27 Oct '21

27 Oct '21

The series of features will make general and simple for new sof machine driver. David Lin (3): ASoC: nau8825: add set_jack coponment support ASoC: nau8825: add disable jack detection support ASoC: nau8825: add clock management for power saving sound/soc/codecs/nau8825.c | 49 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-)

4 9

[PATCH 0/2] Make genaral and simple for new sof machine driver
by David Lin 27 Oct '21

27 Oct '21

The series of features will make general and simple for new sof machine driver. David Lin (2): ASoC: nau8825: add set_jack coponment support ASoC: nau8825: add clock management for power saving sound/soc/codecs/nau8825.c | 48 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 46 insertions(+), 2 deletions(-)

3 7

dt-bindings: treewide: Update @st.com email address to @foss.st.com
by patrice.chotard＠foss.st.com 27 Oct '21

27 Oct '21

From: Patrice Chotard <patrice.chotard(a)foss.st.com> Not all @st.com email address are concerned, only people who have a specific @foss.st.com email will see their entry updated. For some people, who left the company, remove their email. Signed-off-by: Patrice Chotard <patrice.chotard(a)foss.st.com> --- Documentation/devicetree/bindings/arm/sti.yaml | 2 +- Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml | 4 ++-- .../devicetree/bindings/arm/stm32/st,stm32-syscon.yaml | 4 ++-- Documentation/devicetree/bindings/arm/stm32/stm32.yaml | 2 +- Documentation/devicetree/bindings/clock/st,stm32mp1-rcc.yaml | 2 +- Documentation/devicetree/bindings/crypto/st,stm32-crc.yaml | 2 +- Documentation/devicetree/bindings/crypto/st,stm32-cryp.yaml | 2 +- Documentation/devicetree/bindings/crypto/st,stm32-hash.yaml | 2 +- .../devicetree/bindings/display/bridge/snps,dw-mipi-dsi.yaml | 2 +- .../devicetree/bindings/display/panel/orisetech,otm8009a.yaml | 2 +- .../devicetree/bindings/display/panel/raydium,rm68200.yaml | 2 +- Documentation/devicetree/bindings/display/st,stm32-dsi.yaml | 4 ++-- Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml | 4 ++-- Documentation/devicetree/bindings/dma/st,stm32-dma.yaml | 2 +- Documentation/devicetree/bindings/dma/st,stm32-dmamux.yaml | 2 +- Documentation/devicetree/bindings/dma/st,stm32-mdma.yaml | 2 +- .../devicetree/bindings/hwlock/st,stm32-hwspinlock.yaml | 3 +-- Documentation/devicetree/bindings/i2c/st,stm32-i2c.yaml | 2 +- .../devicetree/bindings/iio/adc/sigma-delta-modulator.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml | 2 +- .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 ++-- Documentation/devicetree/bindings/iio/dac/st,stm32-dac.yaml | 2 +- .../bindings/interrupt-controller/st,stm32-exti.yaml | 4 ++-- Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml | 4 ++-- Documentation/devicetree/bindings/media/st,stm32-cec.yaml | 3 +-- Documentation/devicetree/bindings/media/st,stm32-dcmi.yaml | 2 +- .../bindings/memory-controllers/st,stm32-fmc2-ebi.yaml | 2 +- Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml | 2 +- Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml | 3 +-- Documentation/devicetree/bindings/mfd/st,stmfx.yaml | 2 +- Documentation/devicetree/bindings/mfd/st,stpmic1.yaml | 2 +- Documentation/devicetree/bindings/mtd/st,stm32-fmc2-nand.yaml | 2 +- Documentation/devicetree/bindings/net/snps,dwmac.yaml | 2 +- Documentation/devicetree/bindings/net/stm32-dwmac.yaml | 4 ++-- Documentation/devicetree/bindings/nvmem/st,stm32-romem.yaml | 2 +- Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml | 2 +- .../devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml | 2 +- .../devicetree/bindings/regulator/st,stm32-booster.yaml | 2 +- .../devicetree/bindings/regulator/st,stm32-vrefbuf.yaml | 2 +- .../devicetree/bindings/regulator/st,stm32mp1-pwr-reg.yaml | 2 +- .../devicetree/bindings/remoteproc/st,stm32-rproc.yaml | 4 ++-- Documentation/devicetree/bindings/rng/st,stm32-rng.yaml | 2 +- Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml | 2 +- Documentation/devicetree/bindings/serial/st,stm32-uart.yaml | 2 +- Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml | 2 +- Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 2 +- Documentation/devicetree/bindings/sound/st,stm32-sai.yaml | 2 +- Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml | 2 +- Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml | 4 ++-- Documentation/devicetree/bindings/spi/st,stm32-spi.yaml | 4 ++-- .../devicetree/bindings/thermal/st,stm32-thermal.yaml | 2 +- Documentation/devicetree/bindings/timer/st,stm32-timer.yaml | 3 ++- Documentation/devicetree/bindings/usb/st,stusb160x.yaml | 2 +- Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml | 4 ++-- 54 files changed, 67 insertions(+), 69 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/sti.yaml b/Documentation/devicetree/bindings/arm/sti.yaml index b1f28d16d3fb..a41cd8764885 100644 --- a/Documentation/devicetree/bindings/arm/sti.yaml +++ b/Documentation/devicetree/bindings/arm/sti.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: ST STi Platforms Device Tree Bindings maintainers: - - Patrice Chotard <patrice.chotard(a)st.com> + - Patrice Chotard <patrice.chotard(a)foss.st.com> properties: $nodename: diff --git a/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml b/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml index 8e711bd202fd..ecb28e90fd11 100644 --- a/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml @@ -7,8 +7,8 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: STMicroelectronics STM32 ML-AHB interconnect bindings maintainers: - - Fabien Dessenne <fabien.dessenne(a)st.com> - - Arnaud Pouliquen <arnaud.pouliquen(a)st.com> + - Fabien Dessenne <fabien.dessenne(a)foss.st.com> + - Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> description: | These bindings describe the STM32 SoCs ML-AHB interconnect bus which connects diff --git a/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml b/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml index 149afb5df5af..6f846d69c5e1 100644 --- a/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml @@ -7,8 +7,8 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: STMicroelectronics STM32 Platforms System Controller bindings maintainers: - - Alexandre Torgue <alexandre.torgue(a)st.com> - - Christophe Roullier <christophe.roullier(a)st.com> + - Alexandre Torgue <alexandre.torgue(a)foss.st.com> + - Christophe Roullier <christophe.roullier(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/arm/stm32/stm32.yaml b/Documentation/devicetree/bindings/arm/stm32/stm32.yaml index 9a77ab74be99..e33fbb62b430 100644 --- a/Documentation/devicetree/bindings/arm/stm32/stm32.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/stm32.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Platforms Device Tree Bindings maintainers: - - Alexandre Torgue <alexandre.torgue(a)st.com> + - Alexandre Torgue <alexandre.torgue(a)foss.st.com> properties: $nodename: diff --git a/Documentation/devicetree/bindings/clock/st,stm32mp1-rcc.yaml b/Documentation/devicetree/bindings/clock/st,stm32mp1-rcc.yaml index 8b1ecb2ecdd5..a0ae4867ed27 100644 --- a/Documentation/devicetree/bindings/clock/st,stm32mp1-rcc.yaml +++ b/Documentation/devicetree/bindings/clock/st,stm32mp1-rcc.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Reset Clock Controller Binding maintainers: - - Gabriel Fernandez <gabriel.fernandez(a)st.com> + - Gabriel Fernandez <gabriel.fernandez(a)foss.st.com> description: | The RCC IP is both a reset and a clock controller. diff --git a/Documentation/devicetree/bindings/crypto/st,stm32-crc.yaml b/Documentation/devicetree/bindings/crypto/st,stm32-crc.yaml index cee624c14f07..b72e4858f9aa 100644 --- a/Documentation/devicetree/bindings/crypto/st,stm32-crc.yaml +++ b/Documentation/devicetree/bindings/crypto/st,stm32-crc.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 CRC bindings maintainers: - - Lionel Debieve <lionel.debieve(a)st.com> + - Lionel Debieve <lionel.debieve(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/crypto/st,stm32-cryp.yaml b/Documentation/devicetree/bindings/crypto/st,stm32-cryp.yaml index a4574552502a..ed23bf94a8e0 100644 --- a/Documentation/devicetree/bindings/crypto/st,stm32-cryp.yaml +++ b/Documentation/devicetree/bindings/crypto/st,stm32-cryp.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 CRYP bindings maintainers: - - Lionel Debieve <lionel.debieve(a)st.com> + - Lionel Debieve <lionel.debieve(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/crypto/st,stm32-hash.yaml b/Documentation/devicetree/bindings/crypto/st,stm32-hash.yaml index 6dd658f0912c..10ba94792d95 100644 --- a/Documentation/devicetree/bindings/crypto/st,stm32-hash.yaml +++ b/Documentation/devicetree/bindings/crypto/st,stm32-hash.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 HASH bindings maintainers: - - Lionel Debieve <lionel.debieve(a)st.com> + - Lionel Debieve <lionel.debieve(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/display/bridge/snps,dw-mipi-dsi.yaml b/Documentation/devicetree/bindings/display/bridge/snps,dw-mipi-dsi.yaml index 3c3e51af154b..11fd68a70dca 100644 --- a/Documentation/devicetree/bindings/display/bridge/snps,dw-mipi-dsi.yaml +++ b/Documentation/devicetree/bindings/display/bridge/snps,dw-mipi-dsi.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Synopsys DesignWare MIPI DSI host controller maintainers: - - Philippe CORNU <philippe.cornu(a)st.com> + - Philippe CORNU <philippe.cornu(a)foss.st.com> description: | This document defines device tree properties for the Synopsys DesignWare MIPI diff --git a/Documentation/devicetree/bindings/display/panel/orisetech,otm8009a.yaml b/Documentation/devicetree/bindings/display/panel/orisetech,otm8009a.yaml index 4b6dda6dbc0f..17cbd0ad32bf 100644 --- a/Documentation/devicetree/bindings/display/panel/orisetech,otm8009a.yaml +++ b/Documentation/devicetree/bindings/display/panel/orisetech,otm8009a.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Orise Tech OTM8009A 3.97" 480x800 TFT LCD panel (MIPI-DSI video mode) maintainers: - - Philippe CORNU <philippe.cornu(a)st.com> + - Philippe CORNU <philippe.cornu(a)foss.st.com> description: | The Orise Tech OTM8009A is a 3.97" 480x800 TFT LCD panel connected using diff --git a/Documentation/devicetree/bindings/display/panel/raydium,rm68200.yaml b/Documentation/devicetree/bindings/display/panel/raydium,rm68200.yaml index 39477793d289..e8ce2315631a 100644 --- a/Documentation/devicetree/bindings/display/panel/raydium,rm68200.yaml +++ b/Documentation/devicetree/bindings/display/panel/raydium,rm68200.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Raydium Semiconductor Corporation RM68200 5.5" 720p MIPI-DSI TFT LCD panel maintainers: - - Philippe CORNU <philippe.cornu(a)st.com> + - Philippe CORNU <philippe.cornu(a)foss.st.com> description: | The Raydium Semiconductor Corporation RM68200 is a 5.5" 720x1280 TFT LCD diff --git a/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml b/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml index ed310bbe3afe..ce1ef93cce93 100644 --- a/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml +++ b/Documentation/devicetree/bindings/display/st,stm32-dsi.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 DSI host controller maintainers: - - Philippe Cornu <philippe.cornu(a)st.com> - - Yannick Fertre <yannick.fertre(a)st.com> + - Philippe Cornu <philippe.cornu(a)foss.st.com> + - Yannick Fertre <yannick.fertre(a)foss.st.com> description: The STMicroelectronics STM32 DSI controller uses the Synopsys DesignWare MIPI-DSI host controller. diff --git a/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml b/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml index 4ae3d75492d3..01e2da23790b 100644 --- a/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml +++ b/Documentation/devicetree/bindings/display/st,stm32-ltdc.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 lcd-tft display controller maintainers: - - Philippe Cornu <philippe.cornu(a)st.com> - - Yannick Fertre <yannick.fertre(a)st.com> + - Philippe Cornu <philippe.cornu(a)foss.st.com> + - Yannick Fertre <yannick.fertre(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/dma/st,stm32-dma.yaml b/Documentation/devicetree/bindings/dma/st,stm32-dma.yaml index 4bf676fd25dc..55faab6a468e 100644 --- a/Documentation/devicetree/bindings/dma/st,stm32-dma.yaml +++ b/Documentation/devicetree/bindings/dma/st,stm32-dma.yaml @@ -50,7 +50,7 @@ description: | maintainers: - - Amelie Delaunay <amelie.delaunay(a)st.com> + - Amelie Delaunay <amelie.delaunay(a)foss.st.com> allOf: - $ref: "dma-controller.yaml#" diff --git a/Documentation/devicetree/bindings/dma/st,stm32-dmamux.yaml b/Documentation/devicetree/bindings/dma/st,stm32-dmamux.yaml index c8d2b51d8410..f751796531c9 100644 --- a/Documentation/devicetree/bindings/dma/st,stm32-dmamux.yaml +++ b/Documentation/devicetree/bindings/dma/st,stm32-dmamux.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 DMA MUX (DMA request router) bindings maintainers: - - Amelie Delaunay <amelie.delaunay(a)st.com> + - Amelie Delaunay <amelie.delaunay(a)foss.st.com> allOf: - $ref: "dma-router.yaml#" diff --git a/Documentation/devicetree/bindings/dma/st,stm32-mdma.yaml b/Documentation/devicetree/bindings/dma/st,stm32-mdma.yaml index c30be840be1c..87b4afd2cf62 100644 --- a/Documentation/devicetree/bindings/dma/st,stm32-mdma.yaml +++ b/Documentation/devicetree/bindings/dma/st,stm32-mdma.yaml @@ -50,7 +50,7 @@ description: | if no HW ack signal is used by the MDMA client maintainers: - - Amelie Delaunay <amelie.delaunay(a)st.com> + - Amelie Delaunay <amelie.delaunay(a)foss.st.com> allOf: - $ref: "dma-controller.yaml#" diff --git a/Documentation/devicetree/bindings/hwlock/st,stm32-hwspinlock.yaml b/Documentation/devicetree/bindings/hwlock/st,stm32-hwspinlock.yaml index 47cf9c8d97e9..b18c616035a8 100644 --- a/Documentation/devicetree/bindings/hwlock/st,stm32-hwspinlock.yaml +++ b/Documentation/devicetree/bindings/hwlock/st,stm32-hwspinlock.yaml @@ -7,8 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Hardware Spinlock bindings maintainers: - - Benjamin Gaignard <benjamin.gaignard(a)st.com> - - Fabien Dessenne <fabien.dessenne(a)st.com> + - Fabien Dessenne <fabien.dessenne(a)foss.st.com> properties: "#hwlock-cells": diff --git a/Documentation/devicetree/bindings/i2c/st,stm32-i2c.yaml b/Documentation/devicetree/bindings/i2c/st,stm32-i2c.yaml index d747f4990ad8..c07289a643d8 100644 --- a/Documentation/devicetree/bindings/i2c/st,stm32-i2c.yaml +++ b/Documentation/devicetree/bindings/i2c/st,stm32-i2c.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: I2C controller embedded in STMicroelectronics STM32 I2C platform maintainers: - - Pierre-Yves MORDRET <pierre-yves.mordret(a)st.com> + - Pierre-Yves MORDRET <pierre-yves.mordret(a)foss.st.com> allOf: - $ref: /schemas/i2c/i2c-controller.yaml# diff --git a/Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.yaml b/Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.yaml index a390343d0c2a..2287697f1f61 100644 --- a/Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.yaml +++ b/Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Device-Tree bindings for sigma delta modulator maintainers: - - Arnaud Pouliquen <arnaud.pouliquen(a)st.com> + - Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml index a58334c3bb76..a055c78cc047 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml @@ -19,7 +19,7 @@ description: | Each STM32 ADC block can have up to 3 ADC instances. maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 733351dee252..7c260f209687 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 DFSDM ADC device driver maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> - - Olivier Moysan <olivier.moysan(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> + - Olivier Moysan <olivier.moysan(a)foss.st.com> description: | STM32 DFSDM ADC is a sigma delta analog-to-digital converter dedicated to diff --git a/Documentation/devicetree/bindings/iio/dac/st,stm32-dac.yaml b/Documentation/devicetree/bindings/iio/dac/st,stm32-dac.yaml index 393f7005941a..6adeda4087fc 100644 --- a/Documentation/devicetree/bindings/iio/dac/st,stm32-dac.yaml +++ b/Documentation/devicetree/bindings/iio/dac/st,stm32-dac.yaml @@ -15,7 +15,7 @@ description: | current. maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml index 6d3e68eb2e8b..d19c881b4abc 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STM32 External Interrupt Controller Device Tree Bindings maintainers: - - Alexandre Torgue <alexandre.torgue(a)st.com> - - Ludovic Barre <ludovic.barre(a)st.com> + - Alexandre Torgue <alexandre.torgue(a)foss.st.com> + - Ludovic Barre <ludovic.barre(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml index b15da9ba90b2..8eb4bf52ea27 100644 --- a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml +++ b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml @@ -13,8 +13,8 @@ description: channels (N) can be read from a dedicated register. maintainers: - - Fabien Dessenne <fabien.dessenne(a)st.com> - - Arnaud Pouliquen <arnaud.pouliquen(a)st.com> + - Fabien Dessenne <fabien.dessenne(a)foss.st.com> + - Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/media/st,stm32-cec.yaml b/Documentation/devicetree/bindings/media/st,stm32-cec.yaml index d75019c093a4..77144cc6f7db 100644 --- a/Documentation/devicetree/bindings/media/st,stm32-cec.yaml +++ b/Documentation/devicetree/bindings/media/st,stm32-cec.yaml @@ -7,8 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 CEC bindings maintainers: - - Benjamin Gaignard <benjamin.gaignard(a)st.com> - - Yannick Fertre <yannick.fertre(a)st.com> + - Yannick Fertre <yannick.fertre(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/media/st,stm32-dcmi.yaml b/Documentation/devicetree/bindings/media/st,stm32-dcmi.yaml index 41e1d0cd80e5..9c1262a276b5 100644 --- a/Documentation/devicetree/bindings/media/st,stm32-dcmi.yaml +++ b/Documentation/devicetree/bindings/media/st,stm32-dcmi.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Digital Camera Memory Interface (DCMI) binding maintainers: - - Hugues Fruchet <hugues.fruchet(a)st.com> + - Hugues Fruchet <hugues.fruchet(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/memory-controllers/st,stm32-fmc2-ebi.yaml b/Documentation/devicetree/bindings/memory-controllers/st,stm32-fmc2-ebi.yaml index cba74205846a..6b516d3895af 100644 --- a/Documentation/devicetree/bindings/memory-controllers/st,stm32-fmc2-ebi.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/st,stm32-fmc2-ebi.yaml @@ -19,7 +19,7 @@ description: | Select. The FMC2 performs only one access at a time to an external device. maintainers: - - Christophe Kerello <christophe.kerello(a)st.com> + - Christophe Kerello <christophe.kerello(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml b/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml index 8bcea8dd7d90..ec7f0190f46e 100644 --- a/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml @@ -17,7 +17,7 @@ description: | - simple counter from IN1 input signal. maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml b/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml index dace35362a7a..10b330d42901 100644 --- a/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml @@ -17,8 +17,7 @@ description: | programmable prescaler. maintainers: - - Benjamin Gaignard <benjamin.gaignard(a)st.com> - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/mfd/st,stmfx.yaml b/Documentation/devicetree/bindings/mfd/st,stmfx.yaml index 19e9afb385ac..b2a4e4aa7ff6 100644 --- a/Documentation/devicetree/bindings/mfd/st,stmfx.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stmfx.yaml @@ -12,7 +12,7 @@ description: ST Multi-Function eXpander (STMFX) is a slave controller using I2C through VDD) and resistive touchscreen controller. maintainers: - - Amelie Delaunay <amelie.delaunay(a)st.com> + - Amelie Delaunay <amelie.delaunay(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/mfd/st,stpmic1.yaml b/Documentation/devicetree/bindings/mfd/st,stpmic1.yaml index 305123e74a58..426658ad81d4 100644 --- a/Documentation/devicetree/bindings/mfd/st,stpmic1.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stpmic1.yaml @@ -9,7 +9,7 @@ title: STMicroelectonics STPMIC1 Power Management IC bindings description: STMicroelectronics STPMIC1 Power Management IC maintainers: - - pascal Paillet <p.paillet(a)st.com> + - pascal Paillet <p.paillet(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/mtd/st,stm32-fmc2-nand.yaml b/Documentation/devicetree/bindings/mtd/st,stm32-fmc2-nand.yaml index 29c5ef24ac6a..eab8ea3da1fa 100644 --- a/Documentation/devicetree/bindings/mtd/st,stm32-fmc2-nand.yaml +++ b/Documentation/devicetree/bindings/mtd/st,stm32-fmc2-nand.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics Flexible Memory Controller 2 (FMC2) Bindings maintainers: - - Christophe Kerello <christophe.kerello(a)st.com> + - Christophe Kerello <christophe.kerello(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/net/snps,dwmac.yaml b/Documentation/devicetree/bindings/net/snps,dwmac.yaml index 42689b7d03a2..a5c55127cc44 100644 --- a/Documentation/devicetree/bindings/net/snps,dwmac.yaml +++ b/Documentation/devicetree/bindings/net/snps,dwmac.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Synopsys DesignWare MAC Device Tree Bindings maintainers: - - Alexandre Torgue <alexandre.torgue(a)st.com> + - Alexandre Torgue <alexandre.torgue(a)foss.st.com> - Giuseppe Cavallaro <peppe.cavallaro(a)st.com> - Jose Abreu <joabreu(a)synopsys.com> diff --git a/Documentation/devicetree/bindings/net/stm32-dwmac.yaml b/Documentation/devicetree/bindings/net/stm32-dwmac.yaml index d3f05d5934d5..577f4e284425 100644 --- a/Documentation/devicetree/bindings/net/stm32-dwmac.yaml +++ b/Documentation/devicetree/bindings/net/stm32-dwmac.yaml @@ -8,8 +8,8 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: STMicroelectronics STM32 / MCU DWMAC glue layer controller maintainers: - - Alexandre Torgue <alexandre.torgue(a)st.com> - - Christophe Roullier <christophe.roullier(a)st.com> + - Alexandre Torgue <alexandre.torgue(a)foss.st.com> + - Christophe Roullier <christophe.roullier(a)foss.st.com> description: This file documents platform glue layer for stmmac. diff --git a/Documentation/devicetree/bindings/nvmem/st,stm32-romem.yaml b/Documentation/devicetree/bindings/nvmem/st,stm32-romem.yaml index 0b80ce22a2f8..a48c8fa56bce 100644 --- a/Documentation/devicetree/bindings/nvmem/st,stm32-romem.yaml +++ b/Documentation/devicetree/bindings/nvmem/st,stm32-romem.yaml @@ -13,7 +13,7 @@ description: | internal vref (VREFIN_CAL), unique device ID... maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> allOf: - $ref: "nvmem.yaml#" diff --git a/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml b/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml index 3329f1d33a4f..7bae8dff3ef6 100644 --- a/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml +++ b/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml @@ -24,7 +24,7 @@ description: |_ UTMI switch_______| OTG controller maintainers: - - Amelie Delaunay <amelie.delaunay(a)st.com> + - Amelie Delaunay <amelie.delaunay(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml index dfee6d38a701..ac88e01ec430 100644 --- a/Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml @@ -8,7 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STM32 GPIO and Pin Mux/Config controller maintainers: - - Alexandre TORGUE <alexandre.torgue(a)st.com> + - Alexandre TORGUE <alexandre.torgue(a)foss.st.com> description: | STMicroelectronics's STM32 MCUs intregrate a GPIO and Pin mux/config hardware diff --git a/Documentation/devicetree/bindings/regulator/st,stm32-booster.yaml b/Documentation/devicetree/bindings/regulator/st,stm32-booster.yaml index 9f1c70381b82..df0191b1ceba 100644 --- a/Documentation/devicetree/bindings/regulator/st,stm32-booster.yaml +++ b/Documentation/devicetree/bindings/regulator/st,stm32-booster.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 booster for ADC analog input switches bindings maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> description: | Some STM32 devices embed a 3.3V booster supplied by Vdda, that can be used diff --git a/Documentation/devicetree/bindings/regulator/st,stm32-vrefbuf.yaml b/Documentation/devicetree/bindings/regulator/st,stm32-vrefbuf.yaml index 3cd4a254e4cb..836d4156d54c 100644 --- a/Documentation/devicetree/bindings/regulator/st,stm32-vrefbuf.yaml +++ b/Documentation/devicetree/bindings/regulator/st,stm32-vrefbuf.yaml @@ -12,7 +12,7 @@ description: | components through the dedicated VREF+ pin. maintainers: - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> allOf: - $ref: "regulator.yaml#" diff --git a/Documentation/devicetree/bindings/regulator/st,stm32mp1-pwr-reg.yaml b/Documentation/devicetree/bindings/regulator/st,stm32mp1-pwr-reg.yaml index e6322bc3e447..bd07b9c81570 100644 --- a/Documentation/devicetree/bindings/regulator/st,stm32mp1-pwr-reg.yaml +++ b/Documentation/devicetree/bindings/regulator/st,stm32mp1-pwr-reg.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STM32MP1 PWR voltage regulators maintainers: - - Pascal Paillet <p.paillet(a)st.com> + - Pascal Paillet <p.paillet(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml index 1e6225677e00..b587c97c282b 100644 --- a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml @@ -11,8 +11,8 @@ description: boots firmwares on the ST32MP family chipset. maintainers: - - Fabien Dessenne <fabien.dessenne(a)st.com> - - Arnaud Pouliquen <arnaud.pouliquen(a)st.com> + - Fabien Dessenne <fabien.dessenne(a)foss.st.com> + - Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml b/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml index 82bb2e97e889..9a6e4eaf4d3c 100644 --- a/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml +++ b/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml @@ -11,7 +11,7 @@ description: | IP and is fully separated from other crypto functions. maintainers: - - Lionel Debieve <lionel.debieve(a)st.com> + - Lionel Debieve <lionel.debieve(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml b/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml index 5456604b1c14..2359f541b770 100644 --- a/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml +++ b/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Real Time Clock Bindings maintainers: - - Gabriel Fernandez <gabriel.fernandez(a)st.com> + - Gabriel Fernandez <gabriel.fernandez(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/serial/st,stm32-uart.yaml b/Documentation/devicetree/bindings/serial/st,stm32-uart.yaml index f50f4ca893a0..333dc42722d2 100644 --- a/Documentation/devicetree/bindings/serial/st,stm32-uart.yaml +++ b/Documentation/devicetree/bindings/serial/st,stm32-uart.yaml @@ -5,7 +5,7 @@ $id: http://devicetree.org/schemas/serial/st,stm32-uart.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# maintainers: - - Erwan Le Ray <erwan.leray(a)st.com> + - Erwan Le Ray <erwan.leray(a)foss.st.com> title: STMicroelectronics STM32 USART bindings diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml index 0d87e2c86a42..963a871e74da 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: CS42L51 audio codec DT bindings maintainers: - - Olivier Moysan <olivier.moysan(a)st.com> + - Olivier Moysan <olivier.moysan(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml index 6feb5a09c184..d3966ae04ad0 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 SPI/I2S Controller maintainers: - - Olivier Moysan <olivier.moysan(a)st.com> + - Olivier Moysan <olivier.moysan(a)foss.st.com> description: The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index f97132400bb6..1538d11ce9a8 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Serial Audio Interface (SAI) maintainers: - - Olivier Moysan <olivier.moysan(a)st.com> + - Olivier Moysan <olivier.moysan(a)foss.st.com> description: The SAI interface (Serial Audio Interface) offers a wide set of audio diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml index b7f7dc452231..837e830c47ac 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) maintainers: - - Olivier Moysan <olivier.moysan(a)st.com> + - Olivier Moysan <olivier.moysan(a)foss.st.com> description: | The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with diff --git a/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml b/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml index 983c4e54c0be..6ec6f556182f 100644 --- a/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml +++ b/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Quad Serial Peripheral Interface (QSPI) bindings maintainers: - - Christophe Kerello <christophe.kerello(a)st.com> - - Patrice Chotard <patrice.chotard(a)st.com> + - Christophe Kerello <christophe.kerello(a)foss.st.com> + - Patrice Chotard <patrice.chotard(a)foss.st.com> allOf: - $ref: "spi-controller.yaml#" diff --git a/Documentation/devicetree/bindings/spi/st,stm32-spi.yaml b/Documentation/devicetree/bindings/spi/st,stm32-spi.yaml index 2d9af4c506bb..3d64bed266ac 100644 --- a/Documentation/devicetree/bindings/spi/st,stm32-spi.yaml +++ b/Documentation/devicetree/bindings/spi/st,stm32-spi.yaml @@ -13,8 +13,8 @@ description: | from 4 to 32-bit data size. maintainers: - - Erwan Leray <erwan.leray(a)st.com> - - Fabrice Gasnier <fabrice.gasnier(a)st.com> + - Erwan Leray <erwan.leray(a)foss.st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> allOf: - $ref: "spi-controller.yaml#" diff --git a/Documentation/devicetree/bindings/thermal/st,stm32-thermal.yaml b/Documentation/devicetree/bindings/thermal/st,stm32-thermal.yaml index c0f59c56003d..bee41cff5142 100644 --- a/Documentation/devicetree/bindings/thermal/st,stm32-thermal.yaml +++ b/Documentation/devicetree/bindings/thermal/st,stm32-thermal.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 digital thermal sensor (DTS) binding maintainers: - - David Hernandez Sanchez <david.hernandezsanchez(a)st.com> + - Pascal Paillet <p.paillet(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/timer/st,stm32-timer.yaml b/Documentation/devicetree/bindings/timer/st,stm32-timer.yaml index 176aa3c9baf8..937aa8a56366 100644 --- a/Documentation/devicetree/bindings/timer/st,stm32-timer.yaml +++ b/Documentation/devicetree/bindings/timer/st,stm32-timer.yaml @@ -7,7 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 general-purpose 16 and 32 bits timers bindings maintainers: - - Benjamin Gaignard <benjamin.gaignard(a)st.com> + - Fabrice Gasnier <fabrice.gasnier(a)foss.st.com> + - Patrice Chotard <patrice.chotard(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/usb/st,stusb160x.yaml b/Documentation/devicetree/bindings/usb/st,stusb160x.yaml index 9a51efa9d101..ead1571e0e43 100644 --- a/Documentation/devicetree/bindings/usb/st,stusb160x.yaml +++ b/Documentation/devicetree/bindings/usb/st,stusb160x.yaml @@ -7,7 +7,7 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: STMicroelectronics STUSB160x Type-C controller bindings maintainers: - - Amelie Delaunay <amelie.delaunay(a)st.com> + - Amelie Delaunay <amelie.delaunay(a)foss.st.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml b/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml index 481bf91f988a..39736449ba64 100644 --- a/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml +++ b/Documentation/devicetree/bindings/watchdog/st,stm32-iwdg.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: STMicroelectronics STM32 Independent WatchDoG (IWDG) bindings maintainers: - - Yannick Fertre <yannick.fertre(a)st.com> - - Christophe Roullier <christophe.roullier(a)st.com> + - Yannick Fertre <yannick.fertre(a)foss.st.com> + - Christophe Roullier <christophe.roullier(a)foss.st.com> allOf: - $ref: "watchdog.yaml#" -- 2.17.1

5 7

[PATCH 0/3] ALSA: firewire-motu: fix message parser and refine UAPI
by Takashi Sakamoto 27 Oct '21

27 Oct '21

Hi, This patchset is to fix message parser and refine UAPI added in below patchset. https://lore.kernel.org/alsa-devel/20211015080826.34847-1-o-takashi@sakamoc… Current implementation of message parser for register DSP models has a bug to cause invalid memory access in the other type of models. I attempted to fix it in my previous patch[1] but got a comment for simpler solution. The first patch in the patchset is revised version of the previous one. After working for libhinawa(g-i binding)[2], Rust bindings[3], and service program[4] as application of the new features in UAPI, I found that the number of input meters is greater than current structure in UAPI for some models. The second patch is to refine the structure and message parser. I note that the other part of UAPI is pretty good, and the service program now supports control notification in register DSP mdoels as well as command DSP models. After discussion about usage of float type in UAPI, it shall be allowed in some cases. The third patch is to interpret the UAPI for float type in userspace and u32 type in kernel space. [1] https://lore.kernel.org/alsa-devel/20211020042555.40866-1-o-takashi@sakamoc… [2] https://github.com/alsa-project/libhinawa/tree/topic/motu-mixer-ioctl [3] https://github.com/alsa-project/hinawa-rs/tree/topic/v0.5.0 [4] https://github.com/alsa-project/snd-firewire-ctl-services/tree/topic/motu-m… Takashi Sakamoto (3): ALSA: firewire-motu: fix null pointer dereference when polling hwdep character device ALSA: firewire-motu: refine parser for meter information in register DSP models ALSA: firewire-motu: export mete information to userspace as float value include/uapi/sound/firewire.h | 13 +++++++++---- .../motu/motu-command-dsp-message-parser.c | 7 +++++-- sound/firewire/motu/motu-hwdep.c | 16 +++++++++++----- .../motu/motu-register-dsp-message-parser.c | 14 +++++++++----- 4 files changed, 34 insertions(+), 16 deletions(-) -- 2.30.2

1 3

[PATCH] sound: ua101: fix division by zero at probe
by Johan Hovold 27 Oct '21

27 Oct '21

Add the missing endpoint max-packet sanity check to probe() to avoid division by zero in alloc_stream_buffers() in case a malicious device has broken descriptors (or when doing descriptor fuzz testing). Note that USB core will reject URBs submitted for endpoints with zero wMaxPacketSize but that drivers doing packet-size calculations still need to handle this (cf. commit 2548288b4fb0 ("USB: Fix: Don't skip endpoint descriptors with maxpacket=0")). Fixes: 63978ab3e3e9 ("sound: add Edirol UA-101 support") Cc: stable(a)vger.kernel.org # 2.6.34 Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/usb/misc/ua101.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 5834d1dc317e..4f6b20ed29dd 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1000,7 +1000,7 @@ static int detect_usb_format(struct ua101 *ua) fmt_playback->bSubframeSize * ua->playback.channels; epd = &ua->intf[INTF_CAPTURE]->altsetting[1].endpoint[0].desc; - if (!usb_endpoint_is_isoc_in(epd)) { + if (!usb_endpoint_is_isoc_in(epd) || usb_endpoint_maxp(epd) == 0) { dev_err(&ua->dev->dev, "invalid capture endpoint\n"); return -ENXIO; } @@ -1008,7 +1008,7 @@ static int detect_usb_format(struct ua101 *ua) ua->capture.max_packet_bytes = usb_endpoint_maxp(epd); epd = &ua->intf[INTF_PLAYBACK]->altsetting[1].endpoint[0].desc; - if (!usb_endpoint_is_isoc_out(epd)) { + if (!usb_endpoint_is_isoc_out(epd) || usb_endpoint_maxp(epd) == 0) { dev_err(&ua->dev->dev, "invalid playback endpoint\n"); return -ENXIO; } -- 2.32.0

2 1

[PATCH 1/2] ALSA SoC MAX98520 dt-bindings : Added device tree binding for max98520 Signed-off-by: George Song <george.song@maximintegrated.com>
by George Song 26 Oct '21

26 Oct '21

From: "george.song" <george.song(a)analog.com> --- .../bindings/sound/maxim,max98520.yaml | 35 +++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..02481428e2ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - Goerge Song <George.song(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; -- 2.25.1

2 1

[PATCH 0/3] ASoC: amd: acp: Dependency cleanup
by Geert Uytterhoeven 26 Oct '21

26 Oct '21

Hi all, This patch series cleans up the dependencies of the various configuration symbols related to AMD Audio-Coprocessor support. Feel free to squash into a single patch if you prefer that. Thanks! Geert Uytterhoeven (3): ASoC: amd: acp: Wrap AMD Audio ACP components in SND_SOC_AMD_ACP_COMMON ASoC: amd: acp: SND_SOC_AMD_{LEGACY_MACH,SOF_MACH} should depend on X86 && PCI && I2C ASoC: amd: acp: SND_SOC_AMD_ACP_COMMON should depend on X86 && PCI sound/soc/amd/acp/Kconfig | 7 +++++++ 1 file changed, 7 insertions(+) -- 2.25.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

2 4

[PATCH 0/2] ASoC: cs42l42: Fixes to power-down
by Richard Fitzgerald 26 Oct '21

26 Oct '21

Driver probe and remove were inconsistent in what they did to power-down and neither did all steps. In addition to that, neither function prevented the interrupt handler from running during and after power-down. Richard Fitzgerald (2): ASoC: cs42l42: Reset and power-down on remove() and failed probe() ASoC: cs42l42: free_irq() before powering-down on probe() fail sound/soc/codecs/cs42l42.c | 43 +++++++++++++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 12 deletions(-) -- 2.11.0

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[PATCH v11 00/17] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 26 Oct '21

26 Oct '21

Hi Mark, This version is a respin of v10 fixing a build error in 12/17 patch. QCOM SoC relevant non-audio patches in this series has been merged into the Qualcomm drivers-for-5.16 tree, as this series depends those patches an immutable tag is available at: https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git tags/20210927135559.738-6-srinivas.kandagatla(a)linaro.org This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v10: - fix build error during arm64 defconfig build reported by Mark in 12/17 patch for audioreach_tplg_init symbol Srinivas Kandagatla (17): ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1130 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 416 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 822 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 25 files changed, 6664 insertions(+), 1227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

2 18

[PATCH v4 0/5] Update Lpass digital codec macro drivers
by Srinivasa Rao Mandadapu 26 Oct '21

26 Oct '21

This patch set is to add support for lpass sc7280 based targets. Upadate compatible name and change of bulk clock voting to optional clock voting in digital codecs va, rx, tx macro drivers. Changes Since V3: -- Removed fixes tag. -- Change signedoff by sequence. Changes Since V2: -- Add Tx macro deafults for lpass sc7280 Changes Since V1: -- Removed individual clock voting and used bulk clock optional. -- Removed volatile changes and fixed default values. -- Typo errors. Srinivasa Rao Mandadapu (5): ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Enable tx top soundwire mic clock ASoC: codecs: tx-macro: Update tx default values ASoC: codecs: Change bulk clock voting to optional voting in digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +++- sound/soc/codecs/lpass-rx-macro.c | 3 ++- sound/soc/codecs/lpass-tx-macro.c | 25 +++++++++++++++++++--- sound/soc/codecs/lpass-va-macro.c | 3 ++- sound/soc/codecs/lpass-wsa-macro.c | 1 + 8 files changed, 39 insertions(+), 9 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 7

[RESEND PATCH v10 00/17] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 26 Oct '21

26 Oct '21

Hi Mark, Am resending this series as I ended up sending incorrect patches for v10. Sorry for the resend noise. This version is a respin of v9 fixing a build error in 12/17 patch. QCOM SoC relevant non-audio patches in this series has been merged into the Qualcomm drivers-for-5.16 tree, as this series depends those patches an immutable tag is available at: https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git tags/20210927135559.738-6-srinivas.kandagatla(a)linaro.org This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v9: - fix build error during arm64 defconfig build reported by Mark in 12/17 patch Srinivas Kandagatla (17): ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1130 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 416 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 822 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 25 files changed, 6664 insertions(+), 1227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

2 22

[PATCH v10 00/17] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 26 Oct '21

26 Oct '21

Hi Mark, This version is a respin of v9 fixing a build error in 12/17 patch. QCOM SoC relevant non-audio patches in this series has been merged into the Qualcomm drivers-for-5.16 tree, as this series depends those patches an immutable tag is available at: https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git tags/20210927135559.738-6-srinivas.kandagatla(a)linaro.org This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v9: - fix build error during arm64 defconfig build reported by Mark in 12/17 patch Srinivas Kandagatla (17): ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1130 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 416 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 822 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 25 files changed, 6664 insertions(+), 1227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

2 19

[PATCH v9 00/17] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 26 Oct '21

26 Oct '21

Hi Mark, This version addresses all the comments raised as part of v8 review. QCOM SoC relevant non-audio patches in this series has been merged into the Qualcomm drivers-for-5.16 tree, as this series depends those patches an immutable tag is available at: https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git tags/20210927135559.738-6-srinivas.kandagatla(a)linaro.org This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v8: - made use of struct_size where possible. - fixed an incorrect indent. - adding __packed for consistency on structures that are passed to dsp. - rebased to sound-next branch. Srinivas Kandagatla (17): ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1130 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 416 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 822 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 25 files changed, 6664 insertions(+), 1227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

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Re: 回复: [PATCH] sound/usb: fix null pointer dereference on pointer cs_desc
by Takashi Iwai 26 Oct '21

26 Oct '21

On Sun, 24 Oct 2021 13:20:48 +0200, YE Chengfeng wrote: > > Hi, > > I found another potential null-ptr-dereference problem in this file, and not > sure whether it is true. I send this patch to you just for reference, thinks > a lot if you could spare some time to look at it. The Fixes tag doesn't look correct (the code before the refactoring also didn't have NULL checks), so applied without it now. thanks, Takashi > > Thanks so much, > Chengfeng > ------------------------------------------------------------------------------ > 发件人: YE Chengfeng <cyeaa(a)connect.ust.hk> > 发送时间: 2021年10月24日 19:17 > 收件人: perex(a)perex.cz <perex(a)perex.cz>; tiwai(a)suse.com <tiwai(a)suse.com>; > chihhao.chen(a)mediatek.com <chihhao.chen(a)mediatek.com>; damien(a)zamaudio.com > <damien(a)zamaudio.com> > 抄送: alsa-devel(a)alsa-project.org <alsa-devel(a)alsa-project.org>; > linux-kernel(a)vger.kernel.org <linux-kernel(a)vger.kernel.org>; YE Chengfeng > <cyeaa(a)connect.ust.hk> > 主题: [PATCH] sound/usb: fix null pointer dereference on pointer cs_desc > > The pointer cs_desc return from snd_usb_find_clock_source could > be null, so there is a potential null pointer dereference issue. > Fix this by adding a null check before dereference. > > Fixes: 9ec73005 ("ALSA: usb-audio: Refactoring UAC2/3 clock setup code") > Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> > --- > sound/usb/clock.c | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/sound/usb/clock.c b/sound/usb/clock.c > index 81d5ce07d548..98345a695dcc 100644 > --- a/sound/usb/clock.c > +++ b/sound/usb/clock.c > @@ -496,6 +496,10 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio > *chip, > union uac23_clock_source_desc *cs_desc; > > cs_desc = snd_usb_find_clock_source(chip, clock, fmt->protocol); > + > + if (!cs_desc) > + return 0; > + > if (fmt->protocol == UAC_VERSION_3) > bmControls = le32_to_cpu(cs_desc->v3.bmControls); > else > -- > 2.17.1 > >

2 1

[PATCH v3] ASoC: qcom: soundwire: Enable soundwire bus clock for version 1.6
by Srinivasa Rao Mandadapu 26 Oct '21

26 Oct '21

Add support for soundwire 1.6 version to gate RX/TX bus clock. Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. drivers/soundwire/qcom.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 0ef79d6..bd6fabd 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -109,6 +109,7 @@ #define SWR_MAX_CMD_ID 14 #define MAX_FIFO_RD_RETRY 3 #define SWR_OVERFLOW_RETRY_COUNT 30 +#define SWRM_HCTL_REG_MASK ~BIT(1) struct qcom_swrm_port_config { u8 si; @@ -127,6 +128,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + char __iomem *swrm_hctl_reg; struct completion broadcast; struct completion enumeration; struct work_struct slave_work; @@ -610,6 +612,12 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + if (ctrl->swrm_hctl_reg) { + val = ioread32(ctrl->swrm_hctl_reg); + val &= SWRM_HCTL_REG_MASK; + iowrite32(val, ctrl->swrm_hctl_reg); + } + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1200,7 +1208,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl; const struct qcom_swrm_data *data; int ret; - u32 val; + int val, swrm_hctl_reg = 0; ctrl = devm_kzalloc(dev, sizeof(*ctrl), GFP_KERNEL); if (!ctrl) @@ -1251,6 +1259,11 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + if (!of_property_read_u32(dev->of_node, "qcom,swrm-hctl-reg", &swrm_hctl_reg)) { + ctrl->swrm_hctl_reg = devm_ioremap(&pdev->dev, swrm_hctl_reg, 0x4); + if (!ctrl->swrm_hctl_reg) + return -ENODEV; + } ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH v2] ASoC: qcom: soundwire: Enable soundwire bus clock for version 1.6
by Srinivasa Rao Mandadapu 26 Oct '21

26 Oct '21

Add support for soundwire 1.6 version to gate RX/TX bus clock. Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v1: -- Add const name to mask value. drivers/soundwire/qcom.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 0ef79d6..491407f 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -109,6 +109,7 @@ #define SWR_MAX_CMD_ID 14 #define MAX_FIFO_RD_RETRY 3 #define SWR_OVERFLOW_RETRY_COUNT 30 +#define SWRM_HCTL_REG_MASK 0xFFFFFFFD struct qcom_swrm_port_config { u8 si; @@ -127,6 +128,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + char __iomem *swrm_hctl_reg; struct completion broadcast; struct completion enumeration; struct work_struct slave_work; @@ -610,6 +612,12 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + if (ctrl->swrm_hctl_reg) { + val = ioread32(ctrl->swrm_hctl_reg); + val &= SWRM_HCTL_REG_MASK; + iowrite32(val, ctrl->swrm_hctl_reg); + } + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1200,7 +1208,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl; const struct qcom_swrm_data *data; int ret; - u32 val; + int val, swrm_hctl_reg = 0; ctrl = devm_kzalloc(dev, sizeof(*ctrl), GFP_KERNEL); if (!ctrl) @@ -1251,6 +1259,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + if (!of_property_read_u32(dev->of_node, "qcom,swrm-hctl-reg", &swrm_hctl_reg)) + ctrl->swrm_hctl_reg = devm_ioremap(&pdev->dev, swrm_hctl_reg, 0x4); + ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH 2/2] ASoC: codec: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 26 Oct '21

26 Oct '21

New codec driver for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1239 ++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 381 +++++++++ 4 files changed, 1629 insertions(+) create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index deda5ee02ebb..6385142826db 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -217,6 +217,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 imply SND_SOC_TFA989X + imply SND_SOC_TLV320ADC3XXX imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI @@ -1463,6 +1464,12 @@ config SND_SOC_TFA989X Note that the driver currently bypasses the built-in "CoolFlux DSP" and does not support (hardware) volume control. +config SND_SOC_TLV320ADC3XXX + tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" + help + Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 + ADCs. + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4cf939d0d3fb..227d597fd662 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -235,6 +235,7 @@ snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o +snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -566,6 +567,7 @@ obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TFA989X) += snd-soc-tfa989x.o +obj-$(CONFIG_SND_SOC_TLV320ADC3XXX) += snd-soc-tlv320adc3xxx.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c new file mode 100644 index 000000000000..894c07efaaf7 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -0,0 +1,1239 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * History: + * + * Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > + * Copyright: (C) 2010 Mistral Solutions Pvt Ltd. + * + * Author: Dongge wu <dgwu(a)ambarella.com> + * 2015/10/28 - [Dongge wu] Created file + * Copyright (C) 2014-2018, Ambarella, Inc. + * + * Author: Ricard Wanderlof <ricardw(a)axis.com> + * 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. + * 2021/09/03: Porting driver to Linux 5.4. + * Add simple card and DT settable PLL mode, GPIO pin control. + * Copyright (C) 2021 Axis Communications AB + * + */ + +/***************************** INCLUDES ************************************/ + +#include <dt-bindings/sound/tlv320adc3xxx.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320adc3xxx.h" + +/* + * **************************************************************************** + * Macros + * **************************************************************************** + * + */ + +#define PLL_MODE_TEXT(mode) (mode == ADC3XXX_PLL_ENABLE ? "PLL enable" : \ + (mode == ADC3XXX_PLL_BYPASS ? "PLL bypass" : \ + "PLL auto")) + +enum adc3xxx_type { + ADC3001 = 0, + ADC3101 +}; + +/* codec private data */ +struct adc3xxx_priv { + struct device *dev; + enum adc3xxx_type type; + struct clk *mclk; + struct regmap *regmap; + unsigned int pll_mode; + unsigned int rst_pin; + unsigned int rst_active; + unsigned int sysclk; + unsigned int gpio1_cfg; /* GPIO1_CTRL value + 1 (0 => not set) */ + unsigned int gpio2_cfg; /* GPIO2_CTRL value + 1 (0 => not set) */ + int master; + u8 page_no; + int use_pll; +}; + +static int adc3xxx_set_bias_level(struct snd_soc_component *, + enum snd_soc_bias_level); + +static const struct reg_default adc3xxx_defaults[] = { + /* Page 0 */ + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x00 }, + { 4, 0x00 }, { 5, 0x11 }, { 6, 0x04 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x00 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x00 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x01 }, { 19, 0x01 }, + { 20, 0x80 }, { 21, 0x80 }, { 22, 0x04 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x01 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x02 }, { 30, 0x01 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x10 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x02 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x00 }, + { 44, 0x00 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x00 }, + { 52, 0x00 }, { 53, 0x12 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x00 }, { 59, 0x44 }, + { 60, 0x00 }, { 61, 0x01 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x00 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x00 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x88 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x7F }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x7F }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x00 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, { 110, 0x00 }, { 111, 0x00 }, + { 112, 0x00 }, { 113, 0x00 }, { 114, 0x00 }, { 115, 0x00 }, + { 116, 0x00 }, { 117, 0x00 }, { 118, 0x00 }, { 119, 0x00 }, + { 120, 0x00 }, { 121, 0x00 }, { 122, 0x00 }, { 123, 0x00 }, + { 124, 0x00 }, { 125, 0x00 }, { 126, 0x00 }, { 127, 0x00 }, + + /* Page 1 */ + { 128, 0x00 }, { 129, 0x00 }, { 130, 0x00 }, { 131, 0x00 }, + { 132, 0x00 }, { 133, 0x00 }, { 134, 0x00 }, { 135, 0x00 }, + { 136, 0x00 }, { 137, 0x00 }, { 138, 0x00 }, { 139, 0x00 }, + { 140, 0x00 }, { 141, 0x00 }, { 142, 0x00 }, { 143, 0x00 }, + { 144, 0x00 }, { 145, 0x00 }, { 146, 0x00 }, { 147, 0x00 }, + { 148, 0x00 }, { 149, 0x00 }, { 150, 0x00 }, { 151, 0x00 }, + { 152, 0x00 }, { 153, 0x00 }, { 154, 0x00 }, { 155, 0x00 }, + { 156, 0x00 }, { 157, 0x00 }, { 158, 0x00 }, { 159, 0x00 }, + { 160, 0x00 }, { 161, 0x00 }, { 162, 0x00 }, { 163, 0x00 }, + { 164, 0x00 }, { 165, 0x00 }, { 166, 0x00 }, { 167, 0x00 }, + { 168, 0x00 }, { 169, 0x00 }, { 170, 0x00 }, { 171, 0x00 }, + { 172, 0x00 }, { 173, 0x00 }, { 174, 0x00 }, { 175, 0x00 }, + { 176, 0x00 }, { 177, 0x00 }, { 178, 0x00 }, { 179, 0x00 }, + { 180, 0xFF }, { 181, 0x00 }, { 182, 0x3F }, { 183, 0xFF }, + { 184, 0x00 }, { 185, 0x3F }, { 186, 0x00 }, { 187, 0x80 }, + { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, +}; + +static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PAGE_SELECT: /* required by regmap implementation */ + case RESET: + return true; + default: + return false; + } +} + +static const struct regmap_range_cfg adc3xxx_ranges[] = { + { + .range_min = 0, + .range_max = 2 * ADC3xxx_PAGE_SIZE, + .selector_reg = PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = ADC3xxx_PAGE_SIZE, + } +}; + +static const struct regmap_config adc3xxx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = adc3xxx_defaults, + .num_reg_defaults = ARRAY_SIZE(adc3xxx_defaults), + + .volatile_reg = adc3xxx_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + + .ranges = adc3xxx_ranges, + .num_ranges = ARRAY_SIZE(adc3xxx_ranges), + .max_register = 2 * ADC3xxx_PAGE_SIZE, +}; + +/* + * adc3xxx initialization data + * This structure initialization contains the initialization required for + * ADC3xxx. + * These registers values (reg_val) are written into the respective ADC3xxx + * register offset (reg_offset) to initialize ADC3xxx. + * These values are used in adc3xxx_init() function only. + */ +struct adc3xxx_configs { + u8 reg_offset; + u8 reg_val; +}; + +/* The global Register Initialization sequence Array. During the Audio Driver + * Initialization, this array will be utilized to perform the default + * initialization of the audio Driver. + */ +static const struct adc3xxx_configs adc3xxx_reg_init[] = { + /* The default (out-of-reset) values in the x_PGA_SEL_x registers + * disable the inputs by default, but also set the input attenuation + * to -6 dB by default, so we leave the inputs disabled but set + * the attenuation to a more natural 0 dB. + */ + { LEFT_PGA_SEL_1, 0xaa }, + { LEFT_PGA_SEL_2, 0x2a }, + { RIGHT_PGA_SEL_1, 0xaa }, + { RIGHT_PGA_SEL_2, 0x2a }, + /* mute Left PGA + default gain 0 dB (same as reset value) */ + { LEFT_APGA_CTRL, 0x80 }, + /* mute Right PGA + default gain 0 dB (same as reset value) */ + { RIGHT_APGA_CTRL, 0x80 }, + /* Disable AGC (same as reset value) */ + { LEFT_CHN_AGC_1, 0x00 }, + { RIGHT_CHN_AGC_1, 0x00 }, + /* Maximum AGC gain 40 dB */ + /* (Oddly enough, the data sheet says the register field default + * value for the maximum gain is 0b111111 = 127 (verified empirically) + * which is specified as 'Reserved. Do not use.' in the accompanying + * table). + */ + { LEFT_CHN_AGC_3, 0x50 }, + { RIGHT_CHN_AGC_3, 0x50 }, + /* Leave MICBIAS1 and MICBIAS2 powered down (same as reset value) */ + { MICBIAS_CTRL, 0x00 }, + /* Fine Gain 0dB, Left/Right ADC Unmute */ + { ADC_FGA, 0x00 }, +}; + +/* Additional initialization for TLV320ADC3101 + */ +static const struct adc3xxx_configs adc3101_reg_init[] = { + /* Use Primary BCLK and WCLK (same as reset value) */ + { INTERFACE_CTRL_4, 0x00 }, + /* DMCLK output = ADC_MOD_CLK */ + { GPIO2_CTRL, 0x28 }, + /* DMDIN is in input (used for Dig_Mic_In) mode */ + { GPIO1_CTRL, 0x04 }, +}; + +/* + * PLL and Clock settings. + * If p member is 0, PLL is not used. + * The order of the entries in this table have the PLL entries before + * the non-PLL entries, so that the PLL modes are preferred unless + * the PLL mode setting says otherwise. + */ +static const struct adc3xxx_rate_divs adc3xxx_divs[] = { + /* mclk, rate, p, r, j, d, nadc, madc, aosr */ + /* 8k rate */ + {12000000, 8000, 1, 1, 7, 1680, 42, 2, 128}, + {12288000, 8000, 1, 1, 7, 0000, 42, 2, 128}, + /* 11.025k rate */ + {12000000, 11025, 1, 1, 6, 8208, 29, 2, 128}, + /* 16k rate */ + {12000000, 16000, 1, 1, 7, 1680, 21, 2, 128}, + {12288000, 16000, 1, 1, 7, 0000, 21, 2, 128}, + /* 22.05k rate */ + {12000000, 22050, 1, 1, 7, 560, 15, 2, 128}, + /* 32k rate */ + {12000000, 32000, 1, 1, 8, 1920, 12, 2, 128}, + {12288000, 32000, 1, 1, 8, 0000, 12, 2, 128}, + /* 44.1k rate */ + {12000000, 44100, 1, 1, 7, 5264, 8, 2, 128}, + /* 48k rate */ + {12000000, 48000, 1, 1, 7, 1680, 7, 2, 128}, + {12288000, 48000, 1, 1, 7, 0000, 7, 2, 128}, + {24576000, 48000, 1, 1, 3, 5000, 7, 2, 128}, /* With PLL */ + {24576000, 48000, 0, 0, 0, 0000, 2, 2, 128}, /* Without PLL */ + /* 88.2k rate */ + {12000000, 88200, 1, 1, 7, 5264, 4, 4, 64}, + /* 96k rate */ + {12000000, 96000, 1, 1, 8, 1920, 4, 4, 64}, +}; + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_get_divs + * Purpose : This function is to get required divisor from the "adc3xxx_divs" + * table. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_get_divs(struct device *dev, int mclk, int rate, int pll_mode) +{ + int i; + + dev_info(dev, "mclk = %d, rate = %d, clock mode %u\n", + mclk, rate, pll_mode); + for (i = 0; i < ARRAY_SIZE(adc3xxx_divs); i++) { + const struct adc3xxx_rate_divs *mode = &adc3xxx_divs[i]; + + /* Skip this entry if it doesn't fulfill the intended clock + * mode requirement. We consider anything besides the two + * modes below to be the same as ADC3XXX_PLL_AUTO. + */ + if ((pll_mode == ADC3XXX_PLL_BYPASS && mode->pll_p) || + (pll_mode == ADC3XXX_PLL_ENABLE && !mode->pll_p)) + continue; + + if (mode->rate == rate && mode->mclk == mclk) + return i; + } + + dev_info(dev, "Master clock rate %d and sample rate %d is not supported\n", + mclk, rate); + return -EINVAL; +} + +static const char * const micbias_voltage[] = { "off", "2V", "2.5V", "AVDD" }; +static const char * const linein_attenuation[] = { "0db", "-6dB" }; +static const char * const adc_softstepping[] = { "1 step", "2 step", "off" }; +static const char * const multiplier[] = { "1", "2", "4", "8", "16", "32", "64", "128" }; +static const char * const dither_dc_offset[] = { + "0mV", "15mV", "30mV", "45mV", "60mV", "75mV", "90mV", "105mV", + "reserved", "-15mV", "-30mV", "-45mV", "-60mV", "-75mV", "-90mV", "-105mV" +}; + +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +/* agc_target_tlv: 8 values: -5.5, -8, -10, -12, -14, -17, -20, -24 dB */ +/* It would be nice to declare these in the order above, but empirically + * TLV_DB_SCALE_ITEM doesn't take lightly to the increment (second) parameter + * being negative, despite there being examples to the contrary in other + * drivers. So declare these in the order from lowest to highest, and + * set the invert flag in the SOC_DOUBLE_R_TLV declaration instead. + */ +static const DECLARE_TLV_DB_RANGE(agc_target_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-2400, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2000, 300, 0), + 4, 6, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 7, 7, TLV_DB_SCALE_ITEM(-550, 0, 0)); +/* Since the 'disabled' value (mute) is at the highest value in the dB + * range (i.e. just before -32 dB) rather than the lowest, we need to resort + * to using a TLV_DB_RANGE in order to get the mute value in the right place. + */ +static const DECLARE_TLV_DB_RANGE(agc_thresh_tlv, + 0, 30, TLV_DB_SCALE_ITEM(-9000, 200, 0), + 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +/* agc_hysteresis: 4 values: 1, 2, 4 dB, disabled (= mute) */ +static const DECLARE_TLV_DB_RANGE(agc_hysteresis_tlv, + 0, 1, TLV_DB_SCALE_ITEM(100, 100, 0), + 2, 2, TLV_DB_SCALE_ITEM(400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +static const DECLARE_TLV_DB_SCALE(agc_max_tlv, 0, 50, 0); + +#define MICBIAS1_ENUM 0 +#define MICBIAS2_ENUM 1 +#define ADCSOFTSTEP_ENUM 2 +#define ATT_LEFT_IN_1L_ENUM 3 +#define ATT_LEFT_IN_2L_ENUM 4 +#define ATT_LEFT_IN_3L_ENUM 5 +#define ATT_LEFT_IN_1R_ENUM 6 +#define ATT_LEFT_DIF_2L_3L_ENUM 7 +#define ATT_LEFT_DIF_1L_1R_ENUM 8 +#define ATT_LEFT_DIF_2R_3R_ENUM 9 +#define ATT_RIGHT_IN_1R_ENUM 10 +#define ATT_RIGHT_IN_2R_ENUM 11 +#define ATT_RIGHT_IN_3R_ENUM 12 +#define ATT_RIGHT_IN_1L_ENUM 13 +#define ATT_RIGHT_DIF_2R_3R_ENUM 14 +#define ATT_RIGHT_DIF_1L_1R_ENUM 15 +#define ATT_RIGHT_DIF_2L_3L_ENUM 16 +#define AGCATKMULTL_ENUM 17 +#define AGCATKMULTR_ENUM 18 +#define AGCDECMULTL_ENUM 19 +#define AGCDECMULTR_ENUM 20 +#define DITHER_DC_OFFSET_ENUM 21 + +/* Creates an array of the Single Ended Widgets*/ +static const struct soc_enum adc3xxx_enum[] = { + SOC_ENUM_SINGLE(MICBIAS_CTRL, 5, 4, micbias_voltage), + SOC_ENUM_SINGLE(MICBIAS_CTRL, 3, 4, micbias_voltage), + SOC_ENUM_SINGLE(ADC_DIGITAL, 0, 3, adc_softstepping), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_1, 0, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_1, 2, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_1, 4, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_2, 0, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_1, 6, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_2, 4, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_PGA_SEL_2, 2, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_1, 0, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_1, 2, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_1, 4, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_2, 0, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_1, 6, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_2, 4, 2, linein_attenuation), + SOC_ENUM_SINGLE(RIGHT_PGA_SEL_2, 2, 2, linein_attenuation), + SOC_ENUM_SINGLE(LEFT_CHN_AGC_4, 0, 8, multiplier), + SOC_ENUM_SINGLE(RIGHT_CHN_AGC_4, 0, 8, multiplier), + SOC_ENUM_SINGLE(LEFT_CHN_AGC_5, 0, 8, multiplier), + SOC_ENUM_SINGLE(RIGHT_CHN_AGC_5, 0, 8, multiplier), + SOC_ENUM_DOUBLE(DITHER_CTRL, 4, 0, 16, dither_dc_offset), +}; + +/* Various Controls For adc3xxx */ +static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { + SOC_DOUBLE_R_TLV("PGA Gain Volume", LEFT_APGA_CTRL, RIGHT_APGA_CTRL, + 0, 80, 0, pga_tlv), + SOC_DOUBLE_R("AGC Enable", LEFT_CHN_AGC_1, + RIGHT_CHN_AGC_1, 7, 1, 0), + SOC_DOUBLE_R_TLV("AGC Target Level Volume", LEFT_CHN_AGC_1, + RIGHT_CHN_AGC_2, 4, 0x07, 1, agc_target_tlv), + SOC_DOUBLE_R_TLV("AGC Noise Threshold Volume", LEFT_CHN_AGC_2, + RIGHT_CHN_AGC_2, 1, 0x1f, 1, agc_thresh_tlv), + SOC_DOUBLE_R_TLV("AGC Hysteresis Volume", LEFT_CHN_AGC_2, + RIGHT_CHN_AGC_2, 6, 3, 0, agc_hysteresis_tlv), + SOC_DOUBLE_R("AGC Clip Stepping Enable", LEFT_CHN_AGC_2, + RIGHT_CHN_AGC_2, 0, 1, 0), + SOC_DOUBLE_R_TLV("AGC Maximum Gain Volume", LEFT_CHN_AGC_3, + RIGHT_CHN_AGC_3, 0, 0x50, 0, agc_max_tlv), + SOC_DOUBLE_R("AGC Attack Time", LEFT_CHN_AGC_4, + RIGHT_CHN_AGC_4, 3, 0x1F, 0), + /* Would like to have the multipliers as LR pairs, but there is + * no SOC_ENUM_foo which accepts two values in separate registers. + */ + SOC_ENUM("AGC Left Attack Time Multiplier", + adc3xxx_enum[AGCATKMULTL_ENUM]), + SOC_ENUM("AGC Right Attack Time Multiplier", + adc3xxx_enum[AGCATKMULTR_ENUM]), + SOC_DOUBLE_R("AGC Decay Time", LEFT_CHN_AGC_5, + RIGHT_CHN_AGC_5, 3, 0x1F, 0), + SOC_ENUM("AGC Left Decay Time Multiplier", + adc3xxx_enum[AGCDECMULTL_ENUM]), + SOC_ENUM("AGC Right Decay Time Multiplier", + adc3xxx_enum[AGCDECMULTR_ENUM]), + SOC_DOUBLE_R("AGC Noise Debounce", LEFT_CHN_AGC_6, + RIGHT_CHN_AGC_6, 0, 0x1F, 0), + SOC_DOUBLE_R("AGC Signal Debounce", LEFT_CHN_AGC_7, + RIGHT_CHN_AGC_7, 0, 0x0F, 0), + /* Read only register */ + SOC_DOUBLE_R_S_TLV("AGC Applied Gain Volume", LEFT_AGC_GAIN, + RIGHT_AGC_GAIN, 0, -24, 40, 6, 0, adc_tlv), + /* Mic Bias voltage */ + SOC_ENUM("Mic Bias 1 Voltage", adc3xxx_enum[MICBIAS1_ENUM]), + SOC_ENUM("Mic Bias 2 Voltage", adc3xxx_enum[MICBIAS2_ENUM]), + /* ADC soft stepping */ + SOC_ENUM("ADC soft stepping", adc3xxx_enum[ADCSOFTSTEP_ENUM]), + /* Left/Right Input attenuation */ + SOC_ENUM("Left IN_1L input attenuation", + adc3xxx_enum[ATT_LEFT_IN_1L_ENUM]), + SOC_ENUM("Left IN_2L input attenuation", + adc3xxx_enum[ATT_LEFT_IN_1L_ENUM]), + SOC_ENUM("Left IN_3L input attenuation", + adc3xxx_enum[ATT_LEFT_IN_1L_ENUM]), + SOC_ENUM("Left IN_1R input attenuation", + adc3xxx_enum[ATT_LEFT_IN_1R_ENUM]), + SOC_ENUM("Left DIF_2L_3L input attenuation", + adc3xxx_enum[ATT_LEFT_DIF_2L_3L_ENUM]), + SOC_ENUM("Left DIF_1L_1R input attenuation", + adc3xxx_enum[ATT_LEFT_DIF_1L_1R_ENUM]), + SOC_ENUM("Left DIF_2R_3R input attenuation", + adc3xxx_enum[ATT_LEFT_DIF_2R_3R_ENUM]), + SOC_ENUM("Right IN_1R input attenuation", + adc3xxx_enum[ATT_RIGHT_IN_1R_ENUM]), + SOC_ENUM("Right IN_2R input attenuation", + adc3xxx_enum[ATT_RIGHT_IN_2R_ENUM]), + SOC_ENUM("Right IN_3R input attenuation", + adc3xxx_enum[ATT_RIGHT_IN_3R_ENUM]), + SOC_ENUM("Right IN_1L input attenuation", + adc3xxx_enum[ATT_RIGHT_IN_1L_ENUM]), + SOC_ENUM("Right DIF_2R_3R input attenuation", + adc3xxx_enum[ATT_RIGHT_DIF_2R_3R_ENUM]), + SOC_ENUM("Right DIF_1L_1R input attenuation", + adc3xxx_enum[ATT_RIGHT_DIF_1L_1R_ENUM]), + SOC_ENUM("Right DIF_2L_3L input attenuation", + adc3xxx_enum[ATT_RIGHT_DIF_2L_3L_ENUM]), + SOC_DOUBLE_R_S_TLV("ADC Volume Control Volume", LADC_VOL, RADC_VOL, + 0, -24, 40, 6, 0, adc_tlv), + /* Empirically, the following doesn't work the way it's supposed + * to. Values 0, -0.1, -0.2 and -0.3 dB result in the same level, and + * -0.4 dB drops about 0.12 dB on a specific chip. + */ + SOC_DOUBLE_TLV("ADC Fine Volume Control Volume", ADC_FGA, 4, 0, 4, 1, + adc_fine_tlv), + SOC_SINGLE("Left ADC unselected CM bias", LEFT_PGA_SEL_2, 6, 1, 0), + SOC_SINGLE("Right ADC unselected CM bias", RIGHT_PGA_SEL_2, 6, 1, 0), + SOC_ENUM("Dither Control DC Offset", adc3xxx_enum[DITHER_DC_OFFSET_ENUM]), +}; + +/* Left input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1L switch", LEFT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2L switch", LEFT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3L switch", LEFT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L switch", LEFT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R switch", LEFT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R switch", LEFT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1R switch", LEFT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Right input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1R switch", RIGHT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2R switch", RIGHT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3R switch", RIGHT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R switch", RIGHT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R switch", RIGHT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L switch", RIGHT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1L switch", RIGHT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Left Digital Mic input for left ADC */ +static const struct snd_kcontrol_new left_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Left ADC switch", ADC_DIGITAL, 3, 0x1, 0), +}; + +/* Right Digital Mic input for Right ADC */ +static const struct snd_kcontrol_new right_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Right ADC switch", ADC_DIGITAL, 2, 0x1, 0), +}; + +/* adc3xxx Widget structure */ +static const struct snd_soc_dapm_widget adc3xxx_dapm_widgets[] = { + + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input Selection", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input Selection", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + /* PGA selection */ + SND_SOC_DAPM_PGA("Left PGA", LEFT_APGA_CTRL, 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", RIGHT_APGA_CTRL, 7, 1, NULL, 0), + + /* Digital Microphone Input Control for Left/Right ADC */ + SND_SOC_DAPM_MIXER("Left DMic Input", SND_SOC_NOPM, 0, 0, + &left_input_dmic_controls[0], + ARRAY_SIZE(left_input_dmic_controls)), + SND_SOC_DAPM_MIXER("Right DMic Input", SND_SOC_NOPM, 0, 0, + &right_input_dmic_controls[0], + ARRAY_SIZE(right_input_dmic_controls)), + + /* Left/Right ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC_DIGITAL, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC_DIGITAL, 6, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("IN_1L"), + SND_SOC_DAPM_INPUT("IN_1R"), + SND_SOC_DAPM_INPUT("IN_2L"), + SND_SOC_DAPM_INPUT("IN_2R"), + SND_SOC_DAPM_INPUT("IN_3L"), + SND_SOC_DAPM_INPUT("IN_3R"), + SND_SOC_DAPM_INPUT("DIFL_1L_1R"), + SND_SOC_DAPM_INPUT("DIFL_2L_3L"), + SND_SOC_DAPM_INPUT("DIFL_2R_3R"), + SND_SOC_DAPM_INPUT("DIFR_1L_1R"), + SND_SOC_DAPM_INPUT("DIFR_2L_3L"), + SND_SOC_DAPM_INPUT("DIFR_2R_3R"), + SND_SOC_DAPM_INPUT("DMic_L"), + SND_SOC_DAPM_INPUT("DMic_R"), +}; + +/* DAPM Routing related array declaration */ +static const struct snd_soc_dapm_route intercon[] = { +/* Left input selection from switches */ + {"Left Input Selection", "IN_1L switch", "IN_1L"}, + {"Left Input Selection", "IN_2L switch", "IN_2L"}, + {"Left Input Selection", "IN_3L switch", "IN_3L"}, + {"Left Input Selection", "DIF_2L_3L switch", "DIFL_2L_3L"}, + {"Left Input Selection", "DIF_1L_1R switch", "DIFL_1L_1R"}, + {"Left Input Selection", "DIF_2R_3R switch", "DIFL_2R_3R"}, + {"Left Input Selection", "IN_1R switch", "IN_1R"}, + +/* Left input selection to left PGA */ + {"Left PGA", NULL, "Left Input Selection"}, + +/* Left PGA to left ADC */ + {"Left ADC", NULL, "Left PGA"}, + +/* Right input selection from switches */ + {"Right Input Selection", "IN_1R switch", "IN_1R"}, + {"Right Input Selection", "IN_2R switch", "IN_2R"}, + {"Right Input Selection", "IN_3R switch", "IN_3R"}, + {"Right Input Selection", "DIF_2R_3R switch", "DIFR_2R_3R"}, + {"Right Input Selection", "DIF_1L_1R switch", "DIFR_1L_1R"}, + {"Right Input Selection", "DIF_2L_3L switch", "DIFR_2L_3L"}, + {"Right Input Selection", "IN_1L switch", "IN_1L"}, + +/* Right input selection to right PGA */ + {"Right PGA", NULL, "Right Input Selection"}, + +/* Right PGA to right ADC */ + {"Right ADC", NULL, "Right PGA"}, + +/* Left DMic Input selection from switch */ + {"Left DMic Input", "Left ADC switch", "DMic_L"}, + +/* Left DMic to left ADC */ + {"Left ADC", NULL, "Left DMic Input"}, + +/* Right DMic Input selection from switch */ + {"Right DMic Input", "Right ADC switch", "DMic_R"}, + +/* Right DMic to right ADC */ + {"Right ADC", NULL, "Right DMic Input"}, +}; + +/* GPIO control. These are only used when the corresponding GPIO pin is + * configured accordingly. + */ +static const struct snd_kcontrol_new adc3xxx_gpio1_out_control[] = { + SOC_SINGLE("GPIO1 Output", GPIO1_CTRL, GPIO_CTRL_OUTPUT_CTRL_SHIFT, 1, 0) +}; + +static const struct snd_kcontrol_new adc3xxx_gpio2_out_control[] = { + SOC_SINGLE("GPIO2 Output", GPIO2_CTRL, GPIO_CTRL_OUTPUT_CTRL_SHIFT, 1, 0) +}; + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_parse_gpio_mode + * Purpose : Parse a gpio mode property as found in devicetree, and write + * the resulting gpio mode to cfg, if available. + *---------------------------------------------------------------------------- + */ +static int adc3xxx_parse_gpio_mode(struct adc3xxx_priv *adc3xxx, + const char *propname, unsigned int *cfg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val & ~15 || val == 7 || val >= 11) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + if (val == ADC3XXX_GPIO_GPI) + dev_warn(dev, "GPIO Input read not yet implemented\n"); + *cfg = val + 1; /* 0 => not set up, all others shifted +1 */ + } + return 0; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_parse_pll_mode + * Purpose : Parse the clock-mode property as received from devicetree + * or via set_sysclk callback in the clk_id parameter. + *---------------------------------------------------------------------------- + */ +static int adc3xxx_parse_pll_mode(uint32_t val, unsigned int *pll_mode) +{ + if (val == ADC3XXX_PLL_ENABLE || val == ADC3XXX_PLL_BYPASS || + val == ADC3XXX_PLL_AUTO) { + *pll_mode = val; + return 0; + } else if (val == ADC3XXX_PLL_DONT_SET) + return 0; + + return -EINVAL; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_setup_pll + * Purpose : This function is to set up the adc3xxx PLL. + * + *---------------------------------------------------------------------------- + */ +static void adc3xxx_setup_pll(struct snd_soc_component *component, + int div_entry) +{ + int i = div_entry; + + /* P & R values */ + snd_soc_component_write(component, PLL_PROG_PR, + (adc3xxx_divs[i].pll_p << PLLP_SHIFT) | + (adc3xxx_divs[i].pll_r << PLLR_SHIFT)); + /* J value */ + snd_soc_component_write(component, PLL_PROG_J, + adc3xxx_divs[i].pll_j & PLLJ_MASK); + /* D value */ + snd_soc_component_write(component, PLL_PROG_D_LSB, + adc3xxx_divs[i].pll_d & PLLD_LSB_MASK); + snd_soc_component_write(component, PLL_PROG_D_MSB, + (adc3xxx_divs[i].pll_d >> 8) & PLLD_MSB_MASK); +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_hw_params + * Purpose : This function sets the sample rate and audio data word length. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct adc3xxx_priv *adc3xxx = snd_soc_component_get_drvdata(component); + int i, width = 16; + u8 iface_len, bdiv; + + i = adc3xxx_get_divs(component->dev, adc3xxx->sysclk, + params_rate(params), adc3xxx->pll_mode); + + if (i < 0) + return i; + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface_len = IFACE_16BITS; + width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_len = IFACE_20BITS; + width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_len = IFACE_24BITS; + width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_len = IFACE_32BITS; + width = 32; + break; + default: + dev_err(component->dev, "Unsupported serial data format\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, INTERFACE_CTRL_1, WLENGTH_MASK, iface_len); + + if (adc3xxx_divs[i].pll_p) { /* If PLL used for this mode */ + adc3xxx_setup_pll(component, i); + snd_soc_component_write(component, CLKGEN_MUX, USE_PLL); + adc3xxx->use_pll = 1; + } else { + snd_soc_component_write(component, CLKGEN_MUX, NO_PLL); + adc3xxx->use_pll = 0; + } + + /* Write divisor values; masks => bit 7 = 0 => divisor powered down */ + /* NADC */ + snd_soc_component_write(component, ADC_NADC, + adc3xxx_divs[i].nadc & NADC_MASK); + /* MADC */ + snd_soc_component_write(component, ADC_MADC, + adc3xxx_divs[i].madc & MADC_MASK); + /* AOSR */ + snd_soc_component_write(component, ADC_AOSR, + adc3xxx_divs[i].aosr & AOSR_MASK); + /* BDIV N Value */ + /* BCLK is set up to be derived from ADC_CLK */ + bdiv = (adc3xxx_divs[i].aosr * adc3xxx_divs[i].madc) / (2 * width); + /* BCLK enabled (when needed) in adc3xxx_set_bias_level() */ + snd_soc_component_write(component, BCLK_N_DIV, bdiv & BDIV_MASK); + + return 0; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_set_dai_sysclk + * Purpose : This function is to set the DAI system clock + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx_priv *adc3xxx = snd_soc_component_get_drvdata(component); + int ret; + + ret = adc3xxx_parse_pll_mode(clk_id, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + + adc3xxx->sysclk = freq; + dev_info(component->dev, "Set sysclk to %u Hz, %s\n", + freq, PLL_MODE_TEXT(adc3xxx->pll_mode)); + return 0; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_set_dai_fmt + * Purpose : This function is to set the DAI format + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx_priv *adc3xxx = snd_soc_component_get_drvdata(component); + u8 clkdir = 0, format = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + adc3xxx->master = 1; + clkdir = BCLK_MASTER | WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + adc3xxx->master = 0; + break; + default: + dev_err(component->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF: + format = FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF: + format = FORMAT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF: + format = FORMAT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF: + format = FORMAT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF: + format = FORMAT_LJF; + break; + default: + dev_err(component->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* set clock direction and format */ + return snd_soc_component_update_bits(component, INTERFACE_CTRL_1, + clkdir | format, + CLKDIR_MASK | FORMAT_MASK); +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_set_bias_level + * Purpose : This function is to get triggered when dapm events occurs. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adc3xxx_priv *adc3xxx = snd_soc_component_get_drvdata(component); + + if (snd_soc_component_get_bias_level(component) == level) + return 0; + + /* all power is driven by DAPM system */ + switch (level) { + case SND_SOC_BIAS_ON: + /* Enable pll when needed */ + if (adc3xxx->use_pll) + snd_soc_component_update_bits(component, PLL_PROG_PR, + ENABLE_PLL, ENABLE_PLL); + + /* Switch on NADC Divider */ + snd_soc_component_update_bits(component, ADC_NADC, + ENABLE_NADC, ENABLE_NADC); + + /* Switch on MADC Divider */ + snd_soc_component_update_bits(component, ADC_MADC, + ENABLE_MADC, ENABLE_MADC); + + if (adc3xxx->master) + /* Switch on BCLK_N Divider */ + snd_soc_component_update_bits(component, BCLK_N_DIV, + ENABLE_BCLK, ENABLE_BCLK); + else + /* Switch off BCLK_N Divider */ + snd_soc_component_update_bits(component, BCLK_N_DIV, + ENABLE_BCLK, 0); + + if (adc3xxx->use_pll) + /* 10msec delay needed after PLL power-up to allow + * PLL and dividers to stabilize (datasheet p13). + */ + usleep_range(10000, 20000); + + break; + + /* partial On */ + case SND_SOC_BIAS_PREPARE: + break; + + /* Off, with power */ + case SND_SOC_BIAS_STANDBY: + /* Disable pll if used */ + if (adc3xxx->use_pll) + snd_soc_component_update_bits(component, PLL_PROG_PR, + ENABLE_PLL, 0); + + /* Switch off NADC Divider */ + snd_soc_component_update_bits(component, ADC_NADC, + ENABLE_NADC, 0); + + /* Switch off MADC Divider */ + snd_soc_component_update_bits(component, ADC_MADC, + ENABLE_MADC, 0); + + /* Switch off BCLK_N Divider */ + if (adc3xxx->master) + snd_soc_component_update_bits(component, BCLK_N_DIV, + ENABLE_BCLK, 0); + break; + + /* Off without power */ + case SND_SOC_BIAS_OFF: + + /* power off Left/Right ADC channels */ + snd_soc_component_update_bits(component, ADC_DIGITAL, + LADC_PWR_ON | RADC_PWR_ON, 0); + /* Turn off PLLs if used */ + if (adc3xxx->use_pll) + snd_soc_component_update_bits(component, PLL_PROG_PR, + ENABLE_PLL, 0); + + /* Switch off NADC Divider */ + snd_soc_component_update_bits(component, ADC_NADC, + ENABLE_NADC, 0); + + /* Switch off MADC Divider */ + snd_soc_component_update_bits(component, ADC_MADC, + ENABLE_MADC, 0); + + /* Switch off BCLK_N Divider */ + if (adc3xxx->master) + snd_soc_component_update_bits(component, BCLK_N_DIV, + ENABLE_BCLK, 0); + break; + } + + return 0; +} + +static struct snd_soc_dai_ops adc3xxx_dai_ops = { + .hw_params = adc3xxx_hw_params, + .set_sysclk = adc3xxx_set_dai_sysclk, + .set_fmt = adc3xxx_set_dai_fmt, +}; + +struct snd_soc_dai_driver adc3xxx_dai = { + .name = "tlv320adc3xxx-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ADC3xxx_RATES, + .formats = ADC3xxx_FORMATS, + }, + .ops = &adc3xxx_dai_ops, +}; +EXPORT_SYMBOL_GPL(adc3xxx_dai); + +#define REGISTER_INIT(CODEC, MAP) \ + do { \ + int i; \ + for (i = 0; i < ARRAY_SIZE(MAP); i++) \ + snd_soc_component_write(CODEC, MAP[i].reg_offset, \ + MAP[i].reg_val); \ + } while (0) + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_init + * Purpose : This function is to initialize the adc3xxx driver + * register the mixer and dsp interfaces with the kernel. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_init(struct snd_soc_component *component) +{ + struct adc3xxx_priv *adc3xxx = snd_soc_component_get_drvdata(component); + + /* Issue software reset to adc3xxx */ + snd_soc_component_write(component, RESET, SOFT_RESET); + + adc3xxx_set_bias_level(component, SND_SOC_BIAS_STANDBY); + + REGISTER_INIT(component, adc3xxx_reg_init); + if (adc3xxx->type == ADC3101) + REGISTER_INIT(component, adc3101_reg_init); + + if (adc3xxx->gpio1_cfg) { + unsigned int cfg = adc3xxx->gpio1_cfg - 1; + + snd_soc_component_update_bits(component, + GPIO1_CTRL, GPIO_CTRL_CFG_MASK, + cfg << GPIO_CTRL_CFG_SHIFT); + if (cfg == ADC3XXX_GPIO_GPO) { + snd_soc_add_component_controls(component, + adc3xxx_gpio1_out_control, + ARRAY_SIZE(adc3xxx_gpio1_out_control)); + } + } + + if (adc3xxx->gpio2_cfg) { + unsigned int cfg = adc3xxx->gpio2_cfg - 1; + + snd_soc_component_update_bits(component, + GPIO2_CTRL, GPIO_CTRL_CFG_MASK, + cfg << GPIO_CTRL_CFG_SHIFT); + if (cfg == ADC3XXX_GPIO_GPO) { + snd_soc_add_component_controls(component, + adc3xxx_gpio2_out_control, + ARRAY_SIZE(adc3xxx_gpio2_out_control)); + } + } + + //adc3xxx_set_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_probe + * Purpose : This is first driver function called by the SoC core driver. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_probe(struct snd_soc_component *component) +{ + struct adc3xxx_priv *adc3xxx = snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = devm_gpio_request(component->dev, adc3xxx->rst_pin, "adc3xxx reset"); + if (ret < 0) { + dev_err(component->dev, "Failed to request rst_pin: %d\n", ret); + return ret; + } + + /* Reset adc3xxx codec */ + gpio_direction_output(adc3xxx->rst_pin, adc3xxx->rst_active); + usleep_range(2000, 100000); /* Requirement: > 10 ns (datasheet p13) */ + gpio_direction_output(adc3xxx->rst_pin, !adc3xxx->rst_active); + + ret = adc3xxx_init(component); + + if (ret < 0) { + dev_err(component->dev, "Failed to initialize ADC3xxx\n"); + devm_gpio_free(component->dev, adc3xxx->rst_pin); + } + + return ret; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_remove + * Purpose : to remove adc3xxx soc device + * + *---------------------------------------------------------------------------- + */ +static void adc3xxx_remove(struct snd_soc_component *component) +{ + adc3xxx_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_suspend + * Purpose : This function is to suspend the adc3xxx driver. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_suspend(struct snd_soc_component *component) +{ + adc3xxx_set_bias_level(component, SND_SOC_BIAS_OFF); + + return 0; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_resume + * Purpose : This function is to resume the ADC3xxx driver + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_resume(struct snd_soc_component *component) +{ + snd_soc_component_cache_sync(component); + adc3xxx_set_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_adc3xxx = { + .probe = adc3xxx_probe, + .remove = adc3xxx_remove, + .suspend = adc3xxx_suspend, + .resume = adc3xxx_resume, + .set_bias_level = adc3xxx_set_bias_level, + + .controls = adc3xxx_snd_controls, + .num_controls = ARRAY_SIZE(adc3xxx_snd_controls), + .dapm_widgets = adc3xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adc3xxx_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), +}; + + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_i2c_probe + * Purpose : This function attaches the i2c client and initializes + * adc3xxx CODEC. + * + *---------------------------------------------------------------------------- + */ +static int adc3xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + struct adc3xxx_priv *adc3xxx = NULL; + enum of_gpio_flags flags; + int rst_pin; + uint32_t val; + int ret; + + adc3xxx = devm_kzalloc(dev, sizeof(struct adc3xxx_priv), GFP_KERNEL); + if (!adc3xxx) + return -ENOMEM; + adc3xxx->dev = dev; + + adc3xxx->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(adc3xxx->mclk)) { + if (PTR_ERR(adc3xxx->mclk) != -ENOENT) + return PTR_ERR(adc3xxx->mclk); + /* Make a note that there is no mclk specified. */ + adc3xxx->mclk = NULL; + } else if (adc3xxx->mclk) { + ret = clk_prepare_enable(adc3xxx->mclk); + if (ret < 0) + return ret; + dev_info(dev, "Enabled MCLK, freq %lu Hz\n", clk_get_rate(adc3xxx->mclk)); + } + + if (!of_property_read_u32(np, "ti,pll-mode", &val)) { + ret = adc3xxx_parse_pll_mode(val, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + dev_info(dev, "PLL mode set in devicetree: %s\n", + PLL_MODE_TEXT(adc3xxx->pll_mode)); + } + + adc3xxx_parse_gpio_mode(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio1_cfg); + adc3xxx_parse_gpio_mode(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio2_cfg); + + adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); + if (IS_ERR(adc3xxx->regmap)) { + ret = PTR_ERR(adc3xxx->regmap); + return ret; + } + + rst_pin = of_get_gpio_flags(np, 0, &flags); + if (rst_pin < 0 || !gpio_is_valid(rst_pin)) + return -ENXIO; + + adc3xxx->rst_pin = rst_pin; + adc3xxx->rst_active = !(flags & OF_GPIO_ACTIVE_LOW); + adc3xxx->type = id->driver_data; + + i2c_set_clientdata(i2c, adc3xxx); + ret = snd_soc_register_component(dev, + &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec\n"); + + return ret; +} + +/* + *---------------------------------------------------------------------------- + * Function : adc3xxx_i2c_remove + * Purpose : This function removes the i2c client and unregisters + * adc3xxx CODEC. + * + *---------------------------------------------------------------------------- + */ +static int __exit adc3xxx_i2c_remove(struct i2c_client *client) +{ + struct adc3xxx_priv *adc3xxx = i2c_get_clientdata(client); + + if (adc3xxx->mclk) + clk_disable_unprepare(adc3xxx->mclk); + snd_soc_unregister_component(&client->dev); + return 0; +} + +static const struct of_device_id tlv320adc3xxx_of_match[] = { + { .compatible = "ti,tlv320adc3001",}, + { .compatible = "ti,tlv320adc3101",}, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adc3xxx_of_match); + + +static const struct i2c_device_id adc3xxx_i2c_id[] = { + {"tlv320adc3001", ADC3001}, + {"tlv320adc3101", ADC3101}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, adc3xxx_i2c_id); + +/* machine i2c codec control layer */ +static struct i2c_driver adc3xxx_i2c_driver = { + .driver = { + .name = "tlv320adc3xxx-codec", + .of_match_table = tlv320adc3xxx_of_match, + }, + .probe = adc3xxx_i2c_probe, + .remove = adc3xxx_i2c_remove, + .id_table = adc3xxx_i2c_id, +}; +#endif + +static int __init tlv320adc3xxx_init(void) +{ + return i2c_add_driver(&adc3xxx_i2c_driver); +} + +static void __exit tlv320adc3xxx_exit(void) +{ + i2c_del_driver(&adc3xxx_i2c_driver); +} + +module_init(tlv320adc3xxx_init); +module_exit(tlv320adc3xxx_exit); + +MODULE_DESCRIPTION("ASoC TLV320ADC3xxx codec driver"); +MODULE_AUTHOR(" shahina.s(a)mistralsolutions.com "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tlv320adc3xxx.h b/sound/soc/codecs/tlv320adc3xxx.h new file mode 100644 index 000000000000..21276ad71876 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.h @@ -0,0 +1,381 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * History: + * + * Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > + * Copyright: (C) 2010 Mistral Solutions Pvt Ltd. + * + * Author: Dongge wu <dgwu(a)ambarella.com> + * 2015/10/28 - [Dongge wu] Created file + * Copyright (C) 2014-2018, Ambarella, Inc. + * + * Author: Ricard Wanderlof <ricardw(a)axis.com> + * 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. + * 2021/09/03: Porting to Linux 5.4, with enhancements (see tlv320adc3xxx.c). + * Copyright (C) 2021 Axis Communications AB + * + */ + +#ifndef _ADC3xxx_H +#define _ADC3xxx_H + +/* 8 bit mask value */ +#define ADC3xxx_8BITS_MASK 0xFF + +/* Enable slave / master mode for codec */ +#define ADC3xxx_MCBSP_SLAVE //codec master +//#undef ADC3xxx_MCBSP_SLAVE + +#define ADC3xxx_PAGE_SIZE 128 +#define ADC3xxx_REG(page, reg) ((page * ADC3xxx_PAGE_SIZE) + reg) + + +/****************************************************************************/ +/* Page 0 Registers */ +/****************************************************************************/ + +/* Page select register */ +#define PAGE_SELECT ADC3xxx_REG(0, 0) +/* Software reset register */ +#define RESET ADC3xxx_REG(0, 1) + +/* 2-3 Reserved */ + +/* PLL programming register B */ +#define CLKGEN_MUX ADC3xxx_REG(0, 4) +/* PLL P and R-Val */ +#define PLL_PROG_PR ADC3xxx_REG(0, 5) +/* PLL J-Val */ +#define PLL_PROG_J ADC3xxx_REG(0, 6) +/* PLL D-Val MSB */ +#define PLL_PROG_D_MSB ADC3xxx_REG(0, 7) +/* PLL D-Val LSB */ +#define PLL_PROG_D_LSB ADC3xxx_REG(0, 8) + +/* 9-17 Reserved */ + +/* ADC NADC */ +#define ADC_NADC ADC3xxx_REG(0, 18) +/* ADC MADC */ +#define ADC_MADC ADC3xxx_REG(0, 19) +/* ADC AOSR */ +#define ADC_AOSR ADC3xxx_REG(0, 20) +/* ADC IADC */ +#define ADC_IADC ADC3xxx_REG(0, 21) + +/* 23-24 Reserved */ + +/* CLKOUT MUX */ +#define CLKOUT_MUX ADC3xxx_REG(0, 25) +/* CLOCKOUT M divider value */ +#define CLKOUT_M_DIV ADC3xxx_REG(0, 26) +/*Audio Interface Setting Register 1*/ +#define INTERFACE_CTRL_1 ADC3xxx_REG(0, 27) +/* Data Slot Offset (Ch_Offset_1) */ +#define CH_OFFSET_1 ADC3xxx_REG(0, 28) +/* ADC interface control 2 */ +#define INTERFACE_CTRL_2 ADC3xxx_REG(0, 29) +/* BCLK N Divider */ +#define BCLK_N_DIV ADC3xxx_REG(0, 30) +/* Secondary audio interface control 1 */ +#define INTERFACE_CTRL_3 ADC3xxx_REG(0, 31) +/* Secondary audio interface control 2 */ +#define INTERFACE_CTRL_4 ADC3xxx_REG(0, 32) +/* Secondary audio interface control 3 */ +#define INTERFACE_CTRL_5 ADC3xxx_REG(0, 33) +/* I2S sync */ +#define I2S_SYNC ADC3xxx_REG(0, 34) + +/* 35 Reserved */ + +/* ADC flag register */ +#define ADC_FLAG ADC3xxx_REG(0, 36) +/* Data slot offset 2 (Ch_Offset_2) */ +#define CH_OFFSET_2 ADC3xxx_REG(0, 37) +/* I2S TDM control register */ +#define I2S_TDM_CTRL ADC3xxx_REG(0, 38) + +/* 39-41 Reserved */ + +/* Interrupt flags (overflow) */ +#define INTR_FLAG_1 ADC3xxx_REG(0, 42) +/* Interrupt flags (overflow) */ +#define INTR_FLAG_2 ADC3xxx_REG(0, 43) + +/* 44 Reserved */ + +/* Interrupt flags ADC */ +#define INTR_FLAG_ADC1 ADC3xxx_REG(0, 45) + +/* 46 Reserved */ + +/* Interrupt flags ADC */ +#define INTR_FLAG_ADC2 ADC3xxx_REG(0, 47) +/* INT1 interrupt control */ +#define INT1_CTRL ADC3xxx_REG(0, 48) +/* INT2 interrupt control */ +#define INT2_CTRL ADC3xxx_REG(0, 49) + +/* 50 Reserved */ + +/* DMCLK/GPIO2 control */ +#define GPIO2_CTRL ADC3xxx_REG(0, 51) +/* DMDIN/GPIO1 control */ +#define GPIO1_CTRL ADC3xxx_REG(0, 52) +/* DOUT Control */ +#define DOUT_CTRL ADC3xxx_REG(0, 53) + +/* 54-56 Reserved */ + +/* ADC sync control 1 */ +#define SYNC_CTRL_1 ADC3xxx_REG(0, 57) +/* ADC sync control 2 */ +#define SYNC_CTRL_2 ADC3xxx_REG(0, 58) +/* ADC CIC filter gain control */ +#define CIC_GAIN_CTRL ADC3xxx_REG(0, 59) + +/* 60 Reserved */ + +/* ADC processing block selection */ +#define PRB_SELECT ADC3xxx_REG(0, 61) +/* Programmable instruction mode control bits */ +#define INST_MODE_CTRL ADC3xxx_REG(0, 62) + +/* 63-79 Reserved */ + +/* Digital microphone polarity control */ +#define MIC_POLARITY_CTRL ADC3xxx_REG(0, 80) +/* ADC Digital */ +#define ADC_DIGITAL ADC3xxx_REG(0, 81) +/* ADC Fine Gain Adjust */ +#define ADC_FGA ADC3xxx_REG(0, 82) +/* Left ADC Channel Volume Control */ +#define LADC_VOL ADC3xxx_REG(0, 83) +/* Right ADC Channel Volume Control */ +#define RADC_VOL ADC3xxx_REG(0, 84) +/* ADC phase compensation */ +#define ADC_PHASE_COMP ADC3xxx_REG(0, 85) +/* Left Channel AGC Control Register 1 */ +#define LEFT_CHN_AGC_1 ADC3xxx_REG(0, 86) +/* Left Channel AGC Control Register 2 */ +#define LEFT_CHN_AGC_2 ADC3xxx_REG(0, 87) +/* Left Channel AGC Control Register 3 */ +#define LEFT_CHN_AGC_3 ADC3xxx_REG(0, 88) +/* Left Channel AGC Control Register 4 */ +#define LEFT_CHN_AGC_4 ADC3xxx_REG(0, 89) +/* Left Channel AGC Control Register 5 */ +#define LEFT_CHN_AGC_5 ADC3xxx_REG(0, 90) +/* Left Channel AGC Control Register 6 */ +#define LEFT_CHN_AGC_6 ADC3xxx_REG(0, 91) +/* Left Channel AGC Control Register 7 */ +#define LEFT_CHN_AGC_7 ADC3xxx_REG(0, 92) +/* Left AGC gain */ +#define LEFT_AGC_GAIN ADC3xxx_REG(0, 93) +/* Right Channel AGC Control Register 1 */ +#define RIGHT_CHN_AGC_1 ADC3xxx_REG(0, 94) +/* Right Channel AGC Control Register 2 */ +#define RIGHT_CHN_AGC_2 ADC3xxx_REG(0, 95) +/* Right Channel AGC Control Register 3 */ +#define RIGHT_CHN_AGC_3 ADC3xxx_REG(0, 96) +/* Right Channel AGC Control Register 4 */ +#define RIGHT_CHN_AGC_4 ADC3xxx_REG(0, 97) +/* Right Channel AGC Control Register 5 */ +#define RIGHT_CHN_AGC_5 ADC3xxx_REG(0, 98) +/* Right Channel AGC Control Register 6 */ +#define RIGHT_CHN_AGC_6 ADC3xxx_REG(0, 99) +/* Right Channel AGC Control Register 7 */ +#define RIGHT_CHN_AGC_7 ADC3xxx_REG(0, 100) +/* Right AGC gain */ +#define RIGHT_AGC_GAIN ADC3xxx_REG(0, 101) + +/* 102-127 Reserved */ + +/****************************************************************************/ +/* Page 1 Registers */ +/****************************************************************************/ +/* 1-25 Reserved */ + +/* Dither control */ +#define DITHER_CTRL ADC3xxx_REG(1, 26) + +/* 27-50 Reserved */ + +/* MICBIAS Configuration Register */ +#define MICBIAS_CTRL ADC3xxx_REG(1, 51) +/* Left ADC input selection for Left PGA */ +#define LEFT_PGA_SEL_1 ADC3xxx_REG(1, 52) + +/* 53 Reserved */ + +/* Right ADC input selection for Left PGA */ +#define LEFT_PGA_SEL_2 ADC3xxx_REG(1, 54) +/* Right ADC input selection for right PGA */ +#define RIGHT_PGA_SEL_1 ADC3xxx_REG(1, 55) + +/* 56 Reserved */ + +/* Right ADC input selection for right PGA */ +#define RIGHT_PGA_SEL_2 ADC3xxx_REG(1, 57) + +/* 58 Reserved */ + +/* Left analog PGA settings */ +#define LEFT_APGA_CTRL ADC3xxx_REG(1, 59) +/* Right analog PGA settings */ +#define RIGHT_APGA_CTRL ADC3xxx_REG(1, 60) +/* ADC Low current Modes */ +#define LOW_CURRENT_MODES ADC3xxx_REG(1, 61) +/* ADC analog PGA flags */ +#define ANALOG_PGA_FLAGS ADC3xxx_REG(1, 62) + +/* 63-127 Reserved */ + +/****************************************************************************/ +/* Macros and definitions */ +/****************************************************************************/ + +#define ADC3xxx_RATES SNDRV_PCM_RATE_8000_96000 +#define ADC3xxx_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* bits defined for easy usage */ +#define D7 (0x01 << 7) +#define D6 (0x01 << 6) +#define D5 (0x01 << 5) +#define D4 (0x01 << 4) +#define D3 (0x01 << 3) +#define D2 (0x01 << 2) +#define D1 (0x01 << 1) +#define D0 (0x01 << 0) + +/****************************************************************************/ +/* ADC3xxx Register bits */ +/****************************************************************************/ +/* PLL Enable bits */ +#define ENABLE_PLL D7 +#define ENABLE_NADC D7 +#define ENABLE_MADC D7 +#define ENABLE_BCLK D7 + +/* Power bits */ +#define LADC_PWR_ON D7 +#define RADC_PWR_ON D6 + +#define SOFT_RESET D0 +#define BCLK_MASTER D3 +#define WCLK_MASTER D2 + +/* Interface register masks */ +#define FORMAT_MASK (D7|D6) +#define FORMAT_SHIFT 6 +#define WLENGTH_MASK (D5|D4) +#define WLENGTH_SHIFT 4 +#define CLKDIR_MASK (D3|D2) +#define CLKDIR_SHIFT 2 + +/* Interface register bit patterns */ +#define FORMAT_I2S (0 << FORMAT_SHIFT) +#define FORMAT_DSP (1 << FORMAT_SHIFT) +#define FORMAT_RJF (2 << FORMAT_SHIFT) +#define FORMAT_LJF (3 << FORMAT_SHIFT) + +#define IFACE_16BITS (0 << WLENGTH_SHIFT) +#define IFACE_20BITS (1 << WLENGTH_SHIFT) +#define IFACE_24BITS (2 << WLENGTH_SHIFT) +#define IFACE_32BITS (3 << WLENGTH_SHIFT) + +/* PLL P/R bit offsets */ +#define PLLP_SHIFT 4 +#define PLLR_SHIFT 0 +#define PLL_PR_MASK 0x7F +#define PLLJ_MASK 0x3F +#define PLLD_MSB_MASK 0x3F +#define PLLD_LSB_MASK 0xFF +#define NADC_MASK 0x7F +#define MADC_MASK 0x7F +#define AOSR_MASK 0xFF +#define IADC_MASK 0xFF +#define BDIV_MASK 0x7F + +/* PLL_CLKIN bits */ +#define PLL_CLKIN_SHIFT 2 +#define PLL_CLKIN_MCLK 0x0 +#define PLL_CLKIN_BCLK 0x1 +#define PLL_CLKIN_ZERO 0x3 + +/* CODEC_CLKIN bits */ +#define CODEC_CLKIN_SHIFT 0 +#define CODEC_CLKIN_MCLK 0x0 +#define CODEC_CLKIN_BCLK 0x1 +#define CODEC_CLKIN_PLL_CLK 0x3 + +#define USE_PLL ((PLL_CLKIN_MCLK << PLL_CLKIN_SHIFT) | \ + (CODEC_CLKIN_PLL_CLK << CODEC_CLKIN_SHIFT)) +#define NO_PLL ((PLL_CLKIN_ZERO << PLL_CLKIN_SHIFT) | \ + (CODEC_CLKIN_MCLK << CODEC_CLKIN_SHIFT)) + +/* Analog PGA control bits */ +#define LPGA_MUTE D7 +#define RPGA_MUTE D7 + +#define LPGA_GAIN_MASK 0x7F +#define RPGA_GAIN_MASK 0x7F + +/* ADC current modes */ +#define ADC_LOW_CURR_MODE D0 + +/* Left ADC Input selection bits */ +#define LCH_SEL1_SHIFT 0 +#define LCH_SEL2_SHIFT 2 +#define LCH_SEL3_SHIFT 4 +#define LCH_SEL4_SHIFT 6 + +#define LCH_SEL1X_SHIFT 0 +#define LCH_SEL2X_SHIFT 2 +#define LCH_SEL3X_SHIFT 4 +#define LCH_COMMON_MODE D6 +#define BYPASS_LPGA D7 + +/* Right ADC Input selection bits */ +#define RCH_SEL1_SHIFT 0 +#define RCH_SEL2_SHIFT 2 +#define RCH_SEL3_SHIFT 4 +#define RCH_SEL4_SHIFT 6 + +#define RCH_SEL1X_SHIFT 0 +#define RCH_SEL2X_SHIFT 2 +#define RCH_SEL3X_SHIFT 4 +#define RCH_COMMON_MODE D6 +#define BYPASS_RPGA D7 + +/* MICBIAS control bits */ +#define MICBIAS1_SHIFT 5 +#define MICBIAS2_SHIFT 3 + +#define ADC_MAX_VOLUME 64 +#define ADC_POS_VOL 24 + +/* GPIO control bits (GPIO1_CTRL and GPIO2_CTRL) */ +#define GPIO_CTRL_CFG_MASK (D2 | D3 | D4 | D5) +#define GPIO_CTRL_CFG_SHIFT 2 +#define GPIO_CTRL_OUTPUT_CTRL D0 +#define GPIO_CTRL_OUTPUT_CTRL_SHIFT 0 +#define GPIO_CTRL_INPUT_VALUE D1 +#define GPIO_CTRL_INPUT_VALUE_SHIFT 1 + +/****************** RATES TABLE FOR ADC3xxx ************************/ +struct adc3xxx_rate_divs { + u32 mclk; + u32 rate; + u8 pll_p; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 nadc; + u8 madc; + u8 aosr; +}; + +#endif /* _ADC3xxx_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 8

[v6 1/2] ASoC: dt-bindings: max98520: add initial bindings
by George Song 26 Oct '21

26 Oct '21

add initial bindings for max98520 audio amplifier Signed-off-by: George Song <george.song(a)maximintegrated.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/maxim,max98520.yaml | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..b6509cb2c8e0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; + -- 2.25.1

1 1

[PATCH 0/2] sound: fix USB message timeouts
by Johan Hovold 26 Oct '21

26 Oct '21

A number of drivers throughout the tree were incorrectly specifying USB message timeout values in jiffies instead of milliseconds. This series fixes the two sound drivers that got it wrong. Johan Johan Hovold (2): sound: 6fire: fix control and bulk message timeouts sound: line6: fix control and interrupt message timeouts sound/usb/6fire/comm.c | 2 +- sound/usb/6fire/firmware.c | 6 +++--- sound/usb/line6/driver.c | 14 +++++++------- sound/usb/line6/driver.h | 2 +- sound/usb/line6/podhd.c | 6 +++--- sound/usb/line6/toneport.c | 2 +- 6 files changed, 16 insertions(+), 16 deletions(-) -- 2.32.0

2 3

[PATCH] sound/isa/gus: fix null pointer dereference on pointer block
by Chengfeng Ye 26 Oct '21

26 Oct '21

The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index a1c770d826dd..6d664dd8dde0 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.17.1

2 1

[PATCH next] ALSA: mixer: fix deadlock in snd_mixer_oss_set_volume
by Pavel Skripkin 26 Oct '21

26 Oct '21

In commit 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots") added mutex protection in snd_mixer_oss_set_volume(). Second mutex_lock() in same function looks like typo, fix it. Reported-by: syzbot+ace149a75a9a0a399ac7(a)syzkaller.appspotmail.com Fixes: 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots") Signed-off-by: Pavel Skripkin <paskripkin(a)gmail.com> --- sound/core/oss/mixer_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index d5ddc154a735..9620115cfdc0 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -313,7 +313,7 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, pslot->volume[1] = right; result = (left & 0xff) | ((right & 0xff) << 8); unlock: - mutex_lock(&mixer->reg_mutex); + mutex_unlock(&mixer->reg_mutex); return result; } -- 2.33.1

2 1

[PATCH] ASoC: qcom: common: Respect status = "disabled" on DAI link nodes
by Stephan Gerhold 25 Oct '21

25 Oct '21

At the moment, the DAI link nodes in the device tree always have to be specified completely in each device tree. However, the available interfaces (e.g. Primary/Secondary/Tertiary/Quaternary MI2S) are common for all devices of a SoC, so the majority of the definitions can be placed in a common device tree include to reduce boilerplate. Make it possible to define such stubs in device tree includes by respecting the "status" property for the DAI link nodes. This is a trivial change that just requires switching to the _available_ OF functions that check the "status" property additionally. This allows defining a stub like: sound_dai_quaternary: dai-link-quaternary { link-name = "Quaternary MI2S"; status = "disabled"; /* Needs extra codec configuration */ cpu { sound-dai = <&q6afedai QUATERNARY_MI2S_RX>; }; platform { sound-dai = <&q6routing>; }; }; where the codec would be filled in by the device-specific device tree. For existing device trees this change does not make any difference. A missing "status" property is treated like status = "okay". Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 09af00700700..2e1c618f7529 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -44,7 +44,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) return ret; /* Populate links */ - num_links = of_get_child_count(dev->of_node); + num_links = of_get_available_child_count(dev->of_node); /* Allocate the DAI link array */ card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL); @@ -54,7 +54,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) card->num_links = num_links; link = card->dai_link; - for_each_child_of_node(dev->of_node, np) { + for_each_available_child_of_node(dev->of_node, np) { dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); if (!dlc) { ret = -ENOMEM; -- 2.33.1

3 2

[PATCH v3] ASoC: dt-bindings: lpass: add binding headers for digital codecs
by Srinivasa Rao Mandadapu 25 Oct '21

25 Oct '21

Add header defining for lpass internal digital codecs rx,tx and va dai node id's. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v2: -- Rebased to latest sound next and created new patch. Changes since v1: -- Add missing dai node ID's include/dt-bindings/sound/qcom,lpass.h | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/include/dt-bindings/sound/qcom,lpass.h b/include/dt-bindings/sound/qcom,lpass.h index 187af45..a9404c3 100644 --- a/include/dt-bindings/sound/qcom,lpass.h +++ b/include/dt-bindings/sound/qcom,lpass.h @@ -11,9 +11,35 @@ #define LPASS_DP_RX 5 #define LPASS_CDC_DMA_RX0 6 -#define LPASS_CDC_DMA_TX3 7 -#define LPASS_CDC_DMA_VA0 8 -#define LPASS_MAX_PORTS 9 +#define LPASS_CDC_DMA_RX1 7 +#define LPASS_CDC_DMA_RX2 8 +#define LPASS_CDC_DMA_RX3 9 +#define LPASS_CDC_DMA_RX4 10 +#define LPASS_CDC_DMA_RX5 11 +#define LPASS_CDC_DMA_RX6 12 +#define LPASS_CDC_DMA_RX7 13 +#define LPASS_CDC_DMA_RX8 14 +#define LPASS_CDC_DMA_RX9 15 + +#define LPASS_CDC_DMA_TX0 16 +#define LPASS_CDC_DMA_TX1 17 +#define LPASS_CDC_DMA_TX2 18 +#define LPASS_CDC_DMA_TX3 19 +#define LPASS_CDC_DMA_TX4 20 +#define LPASS_CDC_DMA_TX5 21 +#define LPASS_CDC_DMA_TX6 22 +#define LPASS_CDC_DMA_TX7 23 +#define LPASS_CDC_DMA_TX8 24 + +#define LPASS_CDC_DMA_VA_TX0 25 +#define LPASS_CDC_DMA_VA_TX1 26 +#define LPASS_CDC_DMA_VA_TX2 27 +#define LPASS_CDC_DMA_VA_TX3 28 +#define LPASS_CDC_DMA_VA_TX4 29 +#define LPASS_CDC_DMA_VA_TX5 30 +#define LPASS_CDC_DMA_VA_TX6 31 +#define LPASS_CDC_DMA_VA_TX7 32 +#define LPASS_CDC_DMA_VA_TX8 33 #define LPASS_MCLK0 0 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[PATCH] ASoC: topology: Fix stub for snd_soc_tplg_component_remove()
by Mark Brown 25 Oct '21

25 Oct '21

When removing the index argument from snd_soc_topology_component_remove() commit a5b8f71c5477f (ASoC: topology: Remove multistep topology loading) forgot to update the stub for !SND_SOC_TOPOLOGY use, causing build failures for anything that tries to make use of it. Fixes: a5b8f71c5477f (ASoC: topology: Remove multistep topology loading) Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-topology.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 7f33de8ffd95..b4b896f83b94 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -188,8 +188,7 @@ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, #else -static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp, - u32 index) +static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { return 0; } -- 2.30.2

1 1

[PATCH 5.14 018/169] ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers
by Greg Kroah-Hartman 25 Oct '21

25 Oct '21

From: Peter Rosin <peda(a)axentia.se> [ Upstream commit 3f4b57ad07d9237acf1b8cff3f8bf530cacef87a ] Commit 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") breaks the TSE-850 device, which is using a pcm5142 in I2S and CBM_CFS mode (maybe not relevant). Without this fix, the result is: pcm512x 0-004c: Failed to set data format: -16 And after that, no sound. This fix is not 100% correct. The datasheet of at least the pcm5142 states that four bits (0xcc) in the I2S_1 register are "RSV" ("Reserved. Do not access.") and no hint is given as to what the initial values are supposed to be. So, specifying defaults for these bits is wrong. But perhaps better than a broken driver? Fixes: 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kirill Marinushkin <kmarinushkin(a)birdec.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Peter Rosin <peda(a)axentia.se> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/2d221984-7a2e-7006-0f8a-ffb5f64ee885@axentia.se Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc..60dee41816dc 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -116,6 +116,8 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_FS_SPEED_MODE, 0x00 }, { PCM512x_IDAC_1, 0x01 }, { PCM512x_IDAC_2, 0x00 }, + { PCM512x_I2S_1, 0x02 }, + { PCM512x_I2S_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) -- 2.33.0

1 0

[PATCH AUTOSEL 5.10 01/13] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
by Sasha Levin 25 Oct '21

25 Oct '21

From: Yang Yingliang <yangyingliang(a)huawei.com> [ Upstream commit c448b7aa3e66042fc0f849d9a0fb90d1af82e948 ] 'component' is allocated in snd_soc_register_component(), but component->list is not initalized, this may cause snd_soc_del_component_unlocked() deref null ptr in the error handing case. KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007] RIP: 0010:__list_del_entry_valid+0x81/0xf0 Call Trace: snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core] snd_soc_add_component.cold+0x54/0x6c [snd_soc_core] snd_soc_register_component+0x70/0x90 [snd_soc_core] devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core] tas2552_probe+0x265/0x320 [snd_soc_tas2552] ? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552] i2c_device_probe+0xa31/0xbe0 Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize(). Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e677422c1058..133296596864 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2454,6 +2454,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); -- 2.33.0

1 1

[PATCH AUTOSEL 5.14 05/18] ALSA: hda/realtek: Fixes HP Spectre x360 15-eb1xxx speakers
by Sasha Levin 25 Oct '21

25 Oct '21

From: Davide Baldo <davide(a)baldo.me> [ Upstream commit d94befbb5ae379f6dfd4fa6d460eacc09fa7b9c3 ] In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and rear speakers are silent, this patch fixes that by overriding the pin layout and by initializing the amplifier which needs a GPIO pin to be set to 1 then 0, similar to the existing HP Spectre x360 14 model. In order to have volume control, both front and rear speakers were forced to use the DAC1. This patch also correctly map the mute LED but since there is no microphone on/off switch exposed by the alsa subsystem it never turns on by itself. There are still known audio issues in this laptop: headset microphone doesn't work, the button to mute/unmute microphone is not yet mapped, the LED of the mute/unmute speakers doesn't seems to be exposed via GPIO and never turns on. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213953 Signed-off-by: Davide Baldo <davide(a)baldo.me> Link: https://lore.kernel.org/r/20211015072121.5287-1-davide@baldo.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8e6ff50f0f94..a628b2c050bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6418,6 +6418,44 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +/* GPIO1 = amplifier on/off + * GPIO3 = mic mute LED + */ +static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* front/high speakers */ + { 0x17, 0x90170130 }, /* back/bass speakers */ + { } + }; + + //enable micmute led + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->micmute_led_polarity = 1; + /* needed for amp of back speakers */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + snd_hda_apply_pincfgs(codec, pincfgs); + /* share DAC to have unified volume control */ + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp of back speakers */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6570,6 +6608,7 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -8263,6 +8302,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360, }, + [ALC285_FIXUP_HP_SPECTRE_X360_EB1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_eb1 + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -8597,6 +8640,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), @@ -9018,6 +9063,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 03/18] ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets
by Sasha Levin 25 Oct '21

25 Oct '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit aa18457c4af7a9dad1f2b150b11beae1d8ab57aa ] Ensure the default 0dB playback path is always used. The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was spurious, and resulted in a -6dB attenuation being accidentally inserted into the playback path. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20211011144903.28915-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 99c022be94a6..8838b9a0de8e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -901,7 +901,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -972,20 +971,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 02/18] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
by Sasha Levin 25 Oct '21

25 Oct '21

From: Yang Yingliang <yangyingliang(a)huawei.com> [ Upstream commit c448b7aa3e66042fc0f849d9a0fb90d1af82e948 ] 'component' is allocated in snd_soc_register_component(), but component->list is not initalized, this may cause snd_soc_del_component_unlocked() deref null ptr in the error handing case. KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007] RIP: 0010:__list_del_entry_valid+0x81/0xf0 Call Trace: snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core] snd_soc_add_component.cold+0x54/0x6c [snd_soc_core] snd_soc_register_component+0x70/0x90 [snd_soc_core] devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core] tas2552_probe+0x265/0x320 [snd_soc_tas2552] ? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552] i2c_device_probe+0xa31/0xbe0 Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize(). Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 583f2381cfc8..e926985bb2f8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,6 +2599,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); -- 2.33.0

1 0

[PATCH] ASoC: doc: update codec example code
by John Keeping 25 Oct '21

25 Oct '21

As the codec API has evolved the documentation has not kept up and still uses old fields that have been removed. Update the examples to represent the current API. Signed-off-by: John Keeping <john(a)metanate.com> --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index 8a9737eb7597..57df149acafc 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -40,7 +40,7 @@ e.g. .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, - .digital_mute = wm8731_mute, + .mute_stream = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, }; @@ -60,7 +60,7 @@ e.g. .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, .ops = &wm8731_dai_ops, - .symmetric_rates = 1, + .symmetric_rate = 1, }; @@ -177,10 +177,10 @@ when the mute is applied or freed. i.e. :: - static int wm8974_mute(struct snd_soc_dai *dai, int mute) + static int wm8974_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; - u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; + u16 mute_reg = snd_soc_component_read(component, WM8974_DAC) & 0xffbf; if (mute) snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); -- 2.33.1

2 1

[PATCH v1 1/2] ASoC: tegra: Restore AC97 support
by Dmitry Osipenko 25 Oct '21

25 Oct '21

The device-tree of AC97 codecs need to be parsed differently from I2S codecs, plus codec device may need to be created. This was missed by the patch that unified machine drivers into a single driver, fix it. It should restore audio on Toradex Colibri board. Cc: <stable(a)vger.kernel.org> Fixes: cc8f70f56039 ("ASoC: tegra: Unify ASoC machine drivers") Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- sound/soc/tegra/tegra_asoc_machine.c | 59 +++++++++++++++++++++++----- sound/soc/tegra/tegra_asoc_machine.h | 1 + 2 files changed, 50 insertions(+), 10 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index 3cbe6ef1cf9f..d92fabff08bc 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -341,9 +341,34 @@ tegra_machine_parse_phandle(struct device *dev, const char *name) return np; } +static void tegra_machine_unregister_codec(void *pdev) +{ + platform_device_unregister(pdev); +} + +static int tegra_machine_register_codec(struct device *dev, const char *name) +{ + struct platform_device *pdev; + int err; + + if (!name) + return 0; + + pdev = platform_device_register_simple(name, -1, NULL, 0); + if (IS_ERR(pdev)) + return PTR_ERR(pdev); + + err = devm_add_action_or_reset(dev, tegra_machine_unregister_codec, + pdev); + if (err) + return err; + + return 0; +} + int tegra_asoc_machine_probe(struct platform_device *pdev) { - struct device_node *np_codec, *np_i2s; + struct device_node *np_codec, *np_i2s, *np_ac97; const struct tegra_asoc_data *asoc; struct device *dev = &pdev->dev; struct tegra_machine *machine; @@ -404,17 +429,30 @@ int tegra_asoc_machine_probe(struct platform_device *pdev) return err; } - np_codec = tegra_machine_parse_phandle(dev, "nvidia,audio-codec"); - if (IS_ERR(np_codec)) - return PTR_ERR(np_codec); + if (asoc->set_ac97) { + err = tegra_machine_register_codec(dev, asoc->codec_dev_name); + if (err) + return err; + + np_ac97 = tegra_machine_parse_phandle(dev, "nvidia,ac97-controller"); + if (IS_ERR(np_ac97)) + return PTR_ERR(np_ac97); - np_i2s = tegra_machine_parse_phandle(dev, "nvidia,i2s-controller"); - if (IS_ERR(np_i2s)) - return PTR_ERR(np_i2s); + card->dai_link->cpus->of_node = np_ac97; + card->dai_link->platforms->of_node = np_ac97; + } else { + np_codec = tegra_machine_parse_phandle(dev, "nvidia,audio-codec"); + if (IS_ERR(np_codec)) + return PTR_ERR(np_codec); - card->dai_link->cpus->of_node = np_i2s; - card->dai_link->codecs->of_node = np_codec; - card->dai_link->platforms->of_node = np_i2s; + np_i2s = tegra_machine_parse_phandle(dev, "nvidia,i2s-controller"); + if (IS_ERR(np_i2s)) + return PTR_ERR(np_i2s); + + card->dai_link->cpus->of_node = np_i2s; + card->dai_link->codecs->of_node = np_codec; + card->dai_link->platforms->of_node = np_i2s; + } if (asoc->add_common_controls) { card->controls = tegra_machine_controls; @@ -589,6 +627,7 @@ static struct snd_soc_card snd_soc_tegra_wm9712 = { static const struct tegra_asoc_data tegra_wm9712_data = { .card = &snd_soc_tegra_wm9712, .add_common_dapm_widgets = true, + .codec_dev_name = "wm9712-codec", .set_ac97 = true, }; diff --git a/sound/soc/tegra/tegra_asoc_machine.h b/sound/soc/tegra/tegra_asoc_machine.h index 8ee0ec814f67..d6a8d1320551 100644 --- a/sound/soc/tegra/tegra_asoc_machine.h +++ b/sound/soc/tegra/tegra_asoc_machine.h @@ -13,6 +13,7 @@ struct snd_soc_pcm_runtime; struct tegra_asoc_data { unsigned int (*mclk_rate)(unsigned int srate); + const char *codec_dev_name; struct snd_soc_card *card; unsigned int mclk_id; bool hp_jack_gpio_active_low; -- 2.33.1

2 2

[PATCH] ASoC: fix unmet dependency on GPIOLIB for SND_SOC_MAX98357A
by Julian Braha 25 Oct '21

25 Oct '21

When SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH is selected, and GPIOLIB is not selected, Kbuild gives the following warnings: WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_DA7219_MAX98357A_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_DA7219_MAX98357A_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] WARNING: unmet direct dependencies detected for SND_SOC_INTEL_DA7219_MAX98357A_GENERIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && SND_SOC_INTEL_KBL [=y] && I2C [=y] && ACPI [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) This is because SND_SOC_DMIC and SND_SOC_MAX98357A are selected by SND_SOC_INTEL_DA7219_MAX98357A_GENERIC, which is also selected by SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH. However, the selectors do not depend on or select GPIOLIB, despite SND_SOC_DMIC and SND_SOC_MAX98357A depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/intel/boards/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 61b71d6c44cf..87db6a80876a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -371,7 +371,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI + depends on I2C && ACPI && GPIOLIB depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC help -- 2.30.2

2 1

[PATCH] ASoC: cs35l41: Make cs35l41_remove() return void
by Uwe Kleine-König 25 Oct '21

25 Oct '21

Up to now cs35l41_remove() returns zero unconditionally. Make it return void instead which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c, platform and spi remove callbacks is ignored anyway. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/cs35l41-i2c.c | 4 +++- sound/soc/codecs/cs35l41-spi.c | 4 +++- sound/soc/codecs/cs35l41.c | 4 +--- sound/soc/codecs/cs35l41.h | 2 +- 4 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs35l41-i2c.c b/sound/soc/codecs/cs35l41-i2c.c index 2f3d1bd8e046..d5fa8d2c4a70 100644 --- a/sound/soc/codecs/cs35l41-i2c.c +++ b/sound/soc/codecs/cs35l41-i2c.c @@ -75,7 +75,9 @@ static int cs35l41_i2c_remove(struct i2c_client *client) { struct cs35l41_private *cs35l41 = i2c_get_clientdata(client); - return cs35l41_remove(cs35l41); + cs35l41_remove(cs35l41); + + return 0; } #ifdef CONFIG_OF diff --git a/sound/soc/codecs/cs35l41-spi.c b/sound/soc/codecs/cs35l41-spi.c index eac64779eea8..90a921f726c3 100644 --- a/sound/soc/codecs/cs35l41-spi.c +++ b/sound/soc/codecs/cs35l41-spi.c @@ -100,7 +100,9 @@ static int cs35l41_spi_remove(struct spi_device *spi) { struct cs35l41_private *cs35l41 = spi_get_drvdata(spi); - return cs35l41_remove(cs35l41); + cs35l41_remove(cs35l41); + + return 0; } #ifdef CONFIG_OF diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index b16eb6610c0e..94ed21d7676f 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1433,13 +1433,11 @@ int cs35l41_probe(struct cs35l41_private *cs35l41, return ret; } -int cs35l41_remove(struct cs35l41_private *cs35l41) +void cs35l41_remove(struct cs35l41_private *cs35l41) { regmap_write(cs35l41->regmap, CS35L41_IRQ1_MASK1, 0xFFFFFFFF); regulator_bulk_disable(CS35L41_NUM_SUPPLIES, cs35l41->supplies); gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); - - return 0; } MODULE_DESCRIPTION("ASoC CS35L41 driver"); diff --git a/sound/soc/codecs/cs35l41.h b/sound/soc/codecs/cs35l41.h index 0e2639d6ef19..6cffe8a55beb 100644 --- a/sound/soc/codecs/cs35l41.h +++ b/sound/soc/codecs/cs35l41.h @@ -770,6 +770,6 @@ struct cs35l41_private { int cs35l41_probe(struct cs35l41_private *cs35l41, struct cs35l41_platform_data *pdata); -int cs35l41_remove(struct cs35l41_private *cs35l41); +void cs35l41_remove(struct cs35l41_private *cs35l41); #endif /*__CS35L41_H__*/ -- 2.30.2

2 1

[PATCH] ASoc: wm8731: Drop empty spi_driver remove callback
by Uwe Kleine-König 25 Oct '21

25 Oct '21

A driver with a remove callback that just returns 0 behaves identically to a driver with no remove callback at all. So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/wm8731.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index dcee7b2bd3d7..86b1f6eaa599 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -713,18 +713,12 @@ static int wm8731_spi_probe(struct spi_device *spi) return 0; } -static int wm8731_spi_remove(struct spi_device *spi) -{ - return 0; -} - static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", .of_match_table = wm8731_of_match, }, .probe = wm8731_spi_probe, - .remove = wm8731_spi_remove, }; #endif /* CONFIG_SPI_MASTER */ -- 2.30.2

3 2

[PATCH] ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler
by Richard Fitzgerald 25 Oct '21

25 Oct '21

The interrupt handling code was getting the struct device* from a struct snd_soc_component* stored in struct cs42l42_private. If the interrupt was asserted before ASoC calls component_probe() the snd_soc_component* will be NULL. The stored snd_soc_component* is not actually used for anything other than indirectly getting the struct device*. Remove it, and store the struct device* in struct cs42l42_private. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 28 +++++++++------------------- sound/soc/codecs/cs42l42.h | 2 +- 2 files changed, 10 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 0dbe4e23194b..a8fff274ec63 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -526,17 +526,7 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ return 0; } -static int cs42l42_component_probe(struct snd_soc_component *component) -{ - struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - - cs42l42->component = component; - - return 0; -} - static const struct snd_soc_component_driver soc_component_dev_cs42l42 = { - .probe = cs42l42_component_probe, .set_jack = cs42l42_set_jack, .dapm_widgets = cs42l42_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l42_dapm_widgets), @@ -1207,7 +1197,7 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) */ if (cs42l42->hs_type == CS42L42_PLUG_INVALID || cs42l42->hs_type == CS42L42_PLUG_HEADPHONE) { - dev_dbg(cs42l42->component->dev, "Running Manual Detection Fallback\n"); + dev_dbg(cs42l42->dev, "Running Manual Detection Fallback\n"); cs42l42_manual_hs_type_detect(cs42l42); } @@ -1506,19 +1496,19 @@ static int cs42l42_handle_button_press(struct cs42l42_private *cs42l42) switch (bias_level) { case 1: /* Function C button press */ bias_level = SND_JACK_BTN_2; - dev_dbg(cs42l42->component->dev, "Function C button press\n"); + dev_dbg(cs42l42->dev, "Function C button press\n"); break; case 2: /* Function B button press */ bias_level = SND_JACK_BTN_1; - dev_dbg(cs42l42->component->dev, "Function B button press\n"); + dev_dbg(cs42l42->dev, "Function B button press\n"); break; case 3: /* Function D button press */ bias_level = SND_JACK_BTN_3; - dev_dbg(cs42l42->component->dev, "Function D button press\n"); + dev_dbg(cs42l42->dev, "Function D button press\n"); break; case 4: /* Function A button press */ bias_level = SND_JACK_BTN_0; - dev_dbg(cs42l42->component->dev, "Function A button press\n"); + dev_dbg(cs42l42->dev, "Function A button press\n"); break; default: bias_level = 0; @@ -1592,7 +1582,6 @@ static const struct cs42l42_irq_params irq_params_table[] = { static irqreturn_t cs42l42_irq_thread(int irq, void *data) { struct cs42l42_private *cs42l42 = (struct cs42l42_private *)data; - struct snd_soc_component *component = cs42l42->component; unsigned int stickies[12]; unsigned int masks[12]; unsigned int current_plug_status; @@ -1639,7 +1628,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) default: break; } - dev_dbg(component->dev, "Auto detect done (%d)\n", cs42l42->hs_type); + dev_dbg(cs42l42->dev, "Auto detect done (%d)\n", cs42l42->hs_type); } } @@ -1673,7 +1662,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); - dev_dbg(component->dev, "Unplug event\n"); + dev_dbg(cs42l42->dev, "Unplug event\n"); } break; @@ -1689,7 +1678,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) CS42L42_M_HSBIAS_HIZ_MASK)) { if (current_button_status & CS42L42_M_DETECT_TF_MASK) { - dev_dbg(component->dev, "Button released\n"); + dev_dbg(cs42l42->dev, "Button released\n"); report = 0; } else if (current_button_status & CS42L42_M_DETECT_FT_MASK) { report = cs42l42_handle_button_press(cs42l42); @@ -2043,6 +2032,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, if (!cs42l42) return -ENOMEM; + cs42l42->dev = &i2c_client->dev; i2c_set_clientdata(i2c_client, cs42l42); cs42l42->regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index dd4744de9e0a..f45bcc9a3a62 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -833,7 +833,7 @@ static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { struct cs42l42_private { struct regmap *regmap; - struct snd_soc_component *component; + struct device *dev; struct regulator_bulk_data supplies[CS42L42_NUM_SUPPLIES]; struct gpio_desc *reset_gpio; struct completion pdn_done; -- 2.11.0

2 1

[PATCH] ASoC: wm8962: Convert to devm_clk_get_optional()
by Geert Uytterhoeven 25 Oct '21

25 Oct '21

Use the existing devm_clk_get_optional() helper instead of building a similar construct on top of devm_clk_get() that fails to handle all errors but -EPROBE_DEFER. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Compile-tested only. --- sound/soc/codecs/wm8962.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ba16bdf9e478ca5f..a5584ba962dcf005 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3538,9 +3538,8 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, pdata->gpio_init[i] = 0x0; } - pdata->mclk = devm_clk_get(&i2c->dev, NULL); - - return 0; + pdata->mclk = devm_clk_get_optional(&i2c->dev, NULL); + return PTR_ERR_OR_ZERO(pdata->mclk); } static int wm8962_i2c_probe(struct i2c_client *i2c, @@ -3572,14 +3571,6 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, return ret; } - /* Mark the mclk pointer to NULL if no mclk assigned */ - if (IS_ERR(wm8962->pdata.mclk)) { - /* But do not ignore the request for probe defer */ - if (PTR_ERR(wm8962->pdata.mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; - wm8962->pdata.mclk = NULL; - } - for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; -- 2.25.1

3 2

[PATCH] ASoC: amd: acp: Add acp_machine struct for renoir platform.
by Ajit Kumar Pandey 25 Oct '21

25 Oct '21

Add acpi_mach struct for renoir platform to select machine driver based on codec and amp ACPI id. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/amd/acp/acp-renoir.c | 19 +++++++++++++++++++ sound/soc/amd/acp/amd.h | 1 - 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index 82faae1b110b..9b321a055b52 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -25,6 +25,25 @@ #define DRV_NAME "acp_asoc_renoir" +static struct snd_soc_acpi_codecs amp_rt1019 = { + .num_codecs = 1, + .codecs = {"10EC1019"} +}; + +static struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[] = { + { + .id = "10EC5682", + .drv_name = "rn_rt5682_rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + { + .id = "AMDI1019", + .drv_name = "renoir-acp", + }, + {}, +}; + static struct snd_soc_dai_driver acp_renoir_dai[] = { { .name = "acp-i2s-sp", diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 3532f4d3ccff..8eee3d34774b 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -102,7 +102,6 @@ int acp_platform_register(struct device *dev); int acp_platform_unregister(struct device *dev); int acp_machine_select(struct acp_dev_data *adata); -extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[]; static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int direction) { -- 2.25.1

2 1

[PATCH] ASoc: wm8900: Drop empty spi_driver remove callback
by Uwe Kleine-König 25 Oct '21

25 Oct '21

A driver with a remove callback that just returns 0 behaves identically to a driver with no remove callback at all. So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/wm8900.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index a9a6d766a176..bf3a4415a85f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1252,17 +1252,11 @@ static int wm8900_spi_probe(struct spi_device *spi) return ret; } -static int wm8900_spi_remove(struct spi_device *spi) -{ - return 0; -} - static struct spi_driver wm8900_spi_driver = { .driver = { .name = "wm8900", }, .probe = wm8900_spi_probe, - .remove = wm8900_spi_remove, }; #endif /* CONFIG_SPI_MASTER */ -- 2.30.2

3 2

[PATCH] ALSA: firewire-motu: fix invalid memory access when operating hwdep character device
by Takashi Sakamoto 25 Oct '21

25 Oct '21

ALSA firewire-motu driver recently got support for event notification via ALSA HwDep interface for register DSP models. However, when polling ALSA HwDep cdev, the driver can cause null pointer dereference for the other models due to accessing to unallocated memory or uninitialized memory. This commit fixes the bug by check the type of model before accessing to the memory. Fixes: 634ec0b2906e ("ALSA: firewire-motu: notify event for parameter change in register DSP model") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-hwdep.c | 72 ++++++++++++++++++++------------ 1 file changed, 45 insertions(+), 27 deletions(-) diff --git a/sound/firewire/motu/motu-hwdep.c b/sound/firewire/motu/motu-hwdep.c index 9c2e457ce692..ae2d01ddc8d3 100644 --- a/sound/firewire/motu/motu-hwdep.c +++ b/sound/firewire/motu/motu-hwdep.c @@ -16,6 +16,47 @@ #include "motu.h" +static bool has_dsp_event(struct snd_motu *motu) +{ + if (motu->spec->flags & SND_MOTU_SPEC_REGISTER_DSP) + return (snd_motu_register_dsp_message_parser_count_event(motu) > 0); + else + return false; +} + +// NOTE: Take care of page fault due to accessing to userspace. +static long copy_dsp_event_to_user(struct snd_motu *motu, char __user *buf, long count, + struct snd_firewire_event_motu_register_dsp_change *event) +{ + if (motu->spec->flags & SND_MOTU_SPEC_REGISTER_DSP) { + size_t consumed = 0; + u32 __user *ptr; + u32 ev; + + // Header is filled later. + consumed += sizeof(*event); + + while (consumed < count && + snd_motu_register_dsp_message_parser_copy_event(motu, &ev)) { + ptr = (u32 __user *)(buf + consumed); + if (put_user(ev, ptr)) + return -EFAULT; + consumed += sizeof(ev); + } + + event->type = SNDRV_FIREWIRE_EVENT_MOTU_REGISTER_DSP_CHANGE; + event->count = (consumed - sizeof(*event)) / 4; + if (copy_to_user(buf, &event, sizeof(*event))) + return -EFAULT; + + count = consumed; + } else { + count = 0; + } + + return count; +} + static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, loff_t *offset) { @@ -25,8 +66,7 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, spin_lock_irq(&motu->lock); - while (!motu->dev_lock_changed && motu->msg == 0 && - snd_motu_register_dsp_message_parser_count_event(motu) == 0) { + while (!motu->dev_lock_changed && motu->msg == 0 && !has_dsp_event(motu)) { prepare_to_wait(&motu->hwdep_wait, &wait, TASK_INTERRUPTIBLE); spin_unlock_irq(&motu->lock); schedule(); @@ -55,31 +95,10 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, count = min_t(long, count, sizeof(event)); if (copy_to_user(buf, &event, count)) return -EFAULT; - } else if (snd_motu_register_dsp_message_parser_count_event(motu) > 0) { - size_t consumed = 0; - u32 __user *ptr; - u32 ev; - + } else if (has_dsp_event(motu)) { spin_unlock_irq(&motu->lock); - // Header is filled later. - consumed += sizeof(event.motu_register_dsp_change); - - while (consumed < count && - snd_motu_register_dsp_message_parser_copy_event(motu, &ev)) { - ptr = (u32 __user *)(buf + consumed); - if (put_user(ev, ptr)) - return -EFAULT; - consumed += sizeof(ev); - } - - event.motu_register_dsp_change.type = SNDRV_FIREWIRE_EVENT_MOTU_REGISTER_DSP_CHANGE; - event.motu_register_dsp_change.count = - (consumed - sizeof(event.motu_register_dsp_change)) / 4; - if (copy_to_user(buf, &event, sizeof(event.motu_register_dsp_change))) - return -EFAULT; - - count = consumed; + count = copy_dsp_event_to_user(motu, buf, count, &event.motu_register_dsp_change); } return count; @@ -94,8 +113,7 @@ static __poll_t hwdep_poll(struct snd_hwdep *hwdep, struct file *file, poll_wait(file, &motu->hwdep_wait, wait); spin_lock_irq(&motu->lock); - if (motu->dev_lock_changed || motu->msg || - snd_motu_register_dsp_message_parser_count_event(motu) > 0) + if (motu->dev_lock_changed || motu->msg || has_dsp_event(motu)) events = EPOLLIN | EPOLLRDNORM; else events = 0; -- 2.30.2

3 3

[PATCH v3 0/2] Add code to manage DSP clocks and provide dts-binding document
by YC Hung 25 Oct '21

25 Oct '21

From: "yc.hung" <yc.hung(a)mediatek.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and device Tree communities the it will be merged to SOF tree and then merged into ALSA tree. It provides two patches, one is for mt8195 dsp clocks related. Another is for mt8195 dsp dts binding decription. YC Hung (2): ASoC: SOF: mediatek: Add mt8195 dsp clock support dt-bindings: dsp: mediatek: Add mt8195 DSP binding support .../bindings/dsp/mtk,mt8195-dsp.yaml | 139 +++++++++++++++ sound/soc/sof/mediatek/adsp_helper.h | 2 +- sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 163 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 28 +++ sound/soc/sof/mediatek/mt8195/mt8195.c | 22 ++- 6 files changed, 352 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h -- 2.18.0

2 3

[PATCH 0/3] Rockchip S/PDIF Reciever Driver
by Yunhao Tian 25 Oct '21

25 Oct '21

From: Yunhao Tian <t123yh.xyz(a)gmail.com> Hello, This patch series adds support for S/PDIF receiver found on RK3308. The code is based on Rockchip Kernel 4.4. Several ALSA controls such as synchronization status, sample rate and channel status bits are added. Also some basic error handling such as parity error or BMC decode error are performed. The implementation took stm32-spdifrx and mchp-spdifrx as reference. The driver is tested to be working on my custom board based on Firefly CORE-3308Y, with a TOSLINK receiver. This is my first time submitting a new driver to kernel, so I'll appreciate it if I can be corrected for anything wrong. Yunhao Tian (3): ASoC: rockchip: add support for spdifrx receiver ASoC: dt-bindings: rockchip: add spdifrx bindings arm64: dts: rockchip: add spdifrx support to rk3308.dtsi .../bindings/sound/rockchip-spdifrx.yaml | 80 +++ MAINTAINERS | 7 + arch/arm64/boot/dts/rockchip/rk3308.dtsi | 16 + sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdifrx.c | 660 ++++++++++++++++++ sound/soc/rockchip/rockchip_spdifrx.h | 90 +++ 7 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdifrx.yaml create mode 100644 sound/soc/rockchip/rockchip_spdifrx.c create mode 100644 sound/soc/rockchip/rockchip_spdifrx.h -- 2.25.1

2 4

[v5 1/2] ASoC: dt-bindings: max98520: add initial bindings
by George Song 25 Oct '21

25 Oct '21

add initial bindings for max98520 audio amplifier Signed-off-by: George Song <george.song(a)maximintegrated.com> --- .../bindings/sound/maxim,max98520.yaml | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..b6509cb2c8e0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; + -- 2.25.1

2 2

Mute LED on HP OMEN 15 needs a fixup in hda_intel
by Влад Макаров 24 Oct '21

24 Oct '21

Greetings everyone I am not sure if this is the right place to ask, but I don't know where else to look for help. I have a laptop (HP Omen 15 2020 amd) which has a mute button (Fn + F5) which works great, however, the LED indicator on it doesn't light up. With the help of this article <https://asus-linux.org/blog/sound-2021-01-11/> I managed to figure out that I can trigger the LED with the following command: sudo hda-verb /dev/snd/hwC1D0 0x20 0x500 0x0B && sudo hda-verb /dev/snd/hwC1D0 0x20 0x400 0x7778 And disable it with sudo hda-verb /dev/snd/hwC1D0 0x20 0x500 0x0B && sudo hda-verb /dev/snd/hwC1D0 0x20 0x400 0x777 I think that a fix should be added to patch_realtek.c <https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/tree/sound…> (to change those values when mute button is pressed) but this requires knowledge of C and kernel stuff. I don't have much experience with those so I would be very grateful if anyone could help me. I have also filed a kernel bug report <https://bugzilla.kernel.org/show_bug.cgi?id=214735> (with some useful attachments) but nobody responded there. Thanks in advance

1 0

[syzbot] possible deadlock in snd_mixer_oss_ioctl1
by syzbot 24 Oct '21

24 Oct '21

Hello, syzbot found the following issue on: HEAD commit: cf6c9d12750c Add linux-next specific files for 20211022 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=15d3d4c4b00000 kernel config: https://syzkaller.appspot.com/x/.config?x=e694204255ade3a3 dashboard link: https://syzkaller.appspot.com/bug?extid=ace149a75a9a0a399ac7 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=17c0530cb00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1129c952b00000 The issue was bisected to: commit 411cef6adfb38a5bb6bd9af3941b28198e7fb680 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Oct 20 16:48:46 2021 +0000 ALSA: mixer: oss: Fix racy access to slots bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=167f6454b00000 final oops: https://syzkaller.appspot.com/x/report.txt?x=157f6454b00000 console output: https://syzkaller.appspot.com/x/log.txt?x=117f6454b00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+ace149a75a9a0a399ac7(a)syzkaller.appspotmail.com Fixes: 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots") ============================================ WARNING: possible recursive locking detected 5.15.0-rc6-next-20211022-syzkaller #0 Not tainted -------------------------------------------- syz-executor206/6529 is trying to acquire lock: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:316 [inline] ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_ioctl1+0x630/0x19a0 sound/core/oss/mixer_oss.c:375 but task is already holding lock: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:300 [inline] ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_ioctl1+0x4b1/0x19a0 sound/core/oss/mixer_oss.c:375 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(&mixer->reg_mutex); lock(&mixer->reg_mutex); *** DEADLOCK *** May be due to missing lock nesting notation 1 lock held by syz-executor206/6529: #0: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:300 [inline] #0: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_ioctl1+0x4b1/0x19a0 sound/core/oss/mixer_oss.c:375 stack backtrace: CPU: 0 PID: 6529 Comm: syz-executor206 Not tainted 5.15.0-rc6-next-20211022-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 print_deadlock_bug kernel/locking/lockdep.c:2956 [inline] check_deadlock kernel/locking/lockdep.c:2999 [inline] validate_chain kernel/locking/lockdep.c:3788 [inline] __lock_acquire.cold+0x149/0x3ab kernel/locking/lockdep.c:5027 lock_acquire kernel/locking/lockdep.c:5637 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5602 __mutex_lock_common kernel/locking/mutex.c:607 [inline] __mutex_lock+0x12f/0x12f0 kernel/locking/mutex.c:740 snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:316 [inline] snd_mixer_oss_ioctl1+0x630/0x19a0 sound/core/oss/mixer_oss.c:375 snd_mixer_oss_ioctl+0x40/0x50 sound/core/oss/mixer_oss.c:390 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x7f0c06979fe9 Code: 28 c3 e8 2a 14 00 00 66 2e 0f 1f 84 00 00 00 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 c0 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007ffea97dd558 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 00007f0c06979fe9 RDX: 0000000020000080 RSI: 00000000c0044d00 RDI: 0000000000000003 RBP: 00007f0c0693dfd0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f0c0693e060 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 </TASK> --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

Where did snd_pcm_parse_control_id() go?
by GitHub issues - opened 24 Oct '21

24 Oct '21

alsa-project/alsa-lib issue #186 was opened from Saur2000: It seems `snd_pcm_parse_control_id()` was made private in 1.2.5 without any prior deprecation. It is still present in `pcm_external.h`, and our code now fails to link with: ``` ld: src/plugin_control.o: in function `plugin_control_load_user_control': src/plugin_control.c:254: undefined reference to `snd_pcm_parse_control_id' ``` What are we supposed to use instead? And why is the function still in the header file if it is not in the library? Issue URL : https://github.com/alsa-project/alsa-lib/issues/186 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] sound/usb: fix null pointer dereference on pointer cs_desc
by Chengfeng Ye 24 Oct '21

24 Oct '21

The pointer cs_desc return from snd_usb_find_clock_source could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Fixes: 9ec73005 ("ALSA: usb-audio: Refactoring UAC2/3 clock setup code") Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> --- sound/usb/clock.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 81d5ce07d548..98345a695dcc 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -496,6 +496,10 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, union uac23_clock_source_desc *cs_desc; cs_desc = snd_usb_find_clock_source(chip, clock, fmt->protocol); + + if (!cs_desc) + return 0; + if (fmt->protocol == UAC_VERSION_3) bmControls = le32_to_cpu(cs_desc->v3.bmControls); else -- 2.17.1

1 0

[PATCH V2 00/13] Add Yellow Carp platform ASoC driver
by Vijendar Mukunda 23 Oct '21

23 Oct '21

Yellow Carp platform is new APU series based on acp6.x design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Yellow Carp APU with DMIC endpoint support. changes since v1: - fixed SPDX commenting style in acp6x pci driver. - fixed Copyright commenting style in acp6x pci driver, pdm dma driver and machine driver files. - fixed local variable check logic in acp6x_power_on() function. - fixed Kconfig indentation issue - removed irq_flag local variable in irq handler. - add dmi info check in machine driver probe Vijendar Mukunda (13): ASoC: amd: add Yellow Carp ACP6x IP register header ASoC: amd: add Yellow Carp ACP PCI driver ASoC: amd: add acp6x init/de-init functions ASoC: amd: add platform devices for acp6x pdm driver and dmic driver ASoC: amd: add acp6x pdm platform driver ASoC: amd: add acp6x irq handler ASoC: amd: add acp6x pdm driver dma ops ASoC: amd: add acp6x pci driver pm ops ASoC: amd: add acp6x pdm driver pm ops ASoC: amd: enable Yellow carp acp6x drivers build ASoC: amd: create platform device for acp6x machine driver ASoC: amd: add YC machine driver using dmic ASoC: amd: enable Yellow Carp platform machine driver build sound/soc/amd/Kconfig | 21 + sound/soc/amd/Makefile | 1 + sound/soc/amd/yc/Makefile | 9 + sound/soc/amd/yc/acp6x-mach.c | 194 ++++++++++ sound/soc/amd/yc/acp6x-pdm-dma.c | 448 ++++++++++++++++++++++ sound/soc/amd/yc/acp6x.h | 107 ++++++ sound/soc/amd/yc/acp6x_chip_offset_byte.h | 444 +++++++++++++++++++++ sound/soc/amd/yc/pci-acp6x.c | 338 ++++++++++++++++ 8 files changed, 1562 insertions(+) create mode 100644 sound/soc/amd/yc/Makefile create mode 100644 sound/soc/amd/yc/acp6x-mach.c create mode 100644 sound/soc/amd/yc/acp6x-pdm-dma.c create mode 100644 sound/soc/amd/yc/acp6x.h create mode 100644 sound/soc/amd/yc/acp6x_chip_offset_byte.h create mode 100644 sound/soc/amd/yc/pci-acp6x.c -- 2.25.1

2 14

[PATCH v3 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 23 Oct '21

23 Oct '21

changes since v2: - Fixes Kconfig description and indent. Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 51 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 340 +++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 600 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 315 ++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 131 ++++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1993 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

3 13

[PATCH v4 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 23 Oct '21

23 Oct '21

changes since v3: - rebase and fixes merge conflict. - Fixed kernel autobot warning. Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 51 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 340 +++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 600 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 315 ++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 131 ++++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1993 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

2 9

[PATCH v3 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 23 Oct '21

23 Oct '21

Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/common.c | 39 ++ sound/soc/qcom/common.h | 1 + sound/soc/qcom/lpass-cdc-dma.c | 195 +++++++ sound/soc/qcom/lpass-cpu.c | 245 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 513 +++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 599 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 157 ++++++ 11 files changed, 1859 insertions(+), 79 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 11

[PATCH 0/2] ASoC: meson: axg: fix TDM channel order sync
by Jerome Brunet 23 Oct '21

23 Oct '21

On the Amlogic AXG series, the TODDR FIFO may get out of sync with the TDM decoder if the decoder is started before the FIFO. The channel appears shifted in memory in an unpredictable way. To fix this, the trick is to start the FIFO before the TDM decoder. This way the FIFO is already waiting when the 1st channel is produced and it is correctly placed in memory. Jerome Brunet (2): ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++++++----- 2 files changed, 22 insertions(+), 5 deletions(-) -- 2.33.0

2 3

[PATCH] ASoC: rt5682s: Downsizing the DAC volume scale
by derek.fang＠realtek.com 23 Oct '21

23 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> Use 0.75db/step of DAC volume instead of 1.5 to get a more smooth volume curve. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 9dd1796f3339..f6435b233a51 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1003,7 +1003,7 @@ static int rt5682s_set_jack_detect(struct snd_soc_component *component, return 0; } -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9562, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); @@ -1011,7 +1011,7 @@ static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, - RT5682S_L_VOL_SFT + 2, RT5682S_R_VOL_SFT + 2, 63, 0, dac_vol_tlv), + RT5682S_L_VOL_SFT + 1, RT5682S_R_VOL_SFT + 1, 127, 0, dac_vol_tlv), /* CBJ Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682S_REC_MIXER, -- 2.17.1

2 1

[PATCH] ASoC: cs42l42: Remove unused runtime_suspend/runtime_resume callbacks
by Richard Fitzgerald 23 Oct '21

23 Oct '21

The driver has runtime_suspend and runtime_resume callbacks, but pm_runtime is never enabled so these functions won't be called. They could not be used anyway because the runtime_suspend would cause jack detect to stop working. These functions are unused - delete them. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 51 +--------------------------------------------- 1 file changed, 1 insertion(+), 50 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5e4d6791756b..0dbe4e23194b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -25,7 +25,6 @@ #include <linux/regulator/consumer.h> #include <linux/gpio/consumer.h> #include <linux/of_device.h> -#include <linux/pm_runtime.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2175,59 +2174,12 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client) if (i2c_client->irq) devm_free_irq(&i2c_client->dev, i2c_client->irq, cs42l42); - pm_runtime_suspend(&i2c_client->dev); - pm_runtime_disable(&i2c_client->dev); - - return 0; -} - -#ifdef CONFIG_PM -static int cs42l42_runtime_suspend(struct device *dev) -{ - struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); - - regcache_cache_only(cs42l42->regmap, true); - regcache_mark_dirty(cs42l42->regmap); - - /* Hold down reset */ gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); - - /* remove power */ - regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), - cs42l42->supplies); + regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), cs42l42->supplies); return 0; } -static int cs42l42_runtime_resume(struct device *dev) -{ - struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); - int ret; - - /* Enable power */ - ret = regulator_bulk_enable(ARRAY_SIZE(cs42l42->supplies), - cs42l42->supplies); - if (ret != 0) { - dev_err(dev, "Failed to enable supplies: %d\n", - ret); - return ret; - } - - gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); - usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); - - regcache_cache_only(cs42l42->regmap, false); - regcache_sync(cs42l42->regmap); - - return 0; -} -#endif - -static const struct dev_pm_ops cs42l42_runtime_pm = { - SET_RUNTIME_PM_OPS(cs42l42_runtime_suspend, cs42l42_runtime_resume, - NULL) -}; - #ifdef CONFIG_OF static const struct of_device_id cs42l42_of_match[] = { { .compatible = "cirrus,cs42l42", }, @@ -2254,7 +2206,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l42_id); static struct i2c_driver cs42l42_i2c_driver = { .driver = { .name = "cs42l42", - .pm = &cs42l42_runtime_pm, .of_match_table = of_match_ptr(cs42l42_of_match), .acpi_match_table = ACPI_PTR(cs42l42_acpi_match), }, -- 2.11.0

2 1

[PATCH] ASoC: fix unmet dependency on GPIOLIB
by Julian Braha 23 Oct '21

23 Oct '21

When SND_SOC_SC7180 or SND_SOC_STORM is selected, and GPIOLIB is not selected, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_STORM [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] - SND_SOC_SC7180 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && I2C [=y] This is because SND_SOC_MAX98357A is selected by SND_SOC_STORM and SND_SOC_SC7180, but these config options do not select or depend on GPIOLIB, despite SND_SOC_MAX98357A depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/qcom/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index cc7c1de2f1d9..8c5c1ed54bc7 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -38,6 +38,7 @@ config SND_SOC_LPASS_SC7180 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" + depends on GPIOLIB select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -141,7 +142,7 @@ config SND_SOC_SM8250 config SND_SOC_SC7180 tristate "SoC Machine driver for SC7180 boards" - depends on I2C + depends on I2C && GPIOLIB select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A -- 2.25.1

2 1

wrong test in alsa-info.sh
by GitHub issues - opened 22 Oct '21

22 Oct '21

alsa-project/alsa-utils issue #115 was opened from mhampl: In line 691 of alsa_info.sh script (version 0.5.0) a test for $TEMPDIR/alsa-usbstream.tmp is needed instead of $TEMPDIR/lsusb.tmp Issue URL : https://github.com/alsa-project/alsa-utils/issues/115 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: cs42l51: improve error handling in cs42l51_remove()
by Uwe Kleine-König 22 Oct '21

22 Oct '21

When disabling a regulator fails while the device goes away, it doesn't make sense to reenable the already disabled regulators, so use regulator_bulk_force_disable() instead of regulator_bulk_disable(). In the error case returning an error code is also hardly sensible because the only effect is that the i2c core emits a generic error message. Replace this by a more specific message and make sure the i2c remove callback returns 0 instead. While at it also change cs42l51_remove() to return void as it doesn't yield any meaningful return value any more. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, a bit unsure if regulator_bulk_force_disable() is indeed the right function here, its documentation specifies a different usecase. My motivation for this change was to make it obvious in cs42l51_i2c_remove() that there is no error to handle to eventually make i2c remove callbacks return void, too. Best regards Uwe sound/soc/codecs/cs42l51-i2c.c | 4 +++- sound/soc/codecs/cs42l51.c | 11 ++++++++--- sound/soc/codecs/cs42l51.h | 2 +- 3 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 70260e0a8f09..3cb21a2ba29f 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -31,7 +31,9 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c, static int cs42l51_i2c_remove(struct i2c_client *i2c) { - return cs42l51_remove(&i2c->dev); + cs42l51_remove(&i2c->dev); + + return 0; } static const struct dev_pm_ops cs42l51_pm_ops = { diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..f3667693e2ea 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -793,14 +793,19 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(cs42l51_probe); -int cs42l51_remove(struct device *dev) +void cs42l51_remove(struct device *dev) { struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + int ret; gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); - return regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), - cs42l51->supplies); + ret = regulator_bulk_force_disable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); + if (ret) + dev_warn(dev, "Failed to disable all regulators (%pe)\n", + ERR_PTR(ret)); + } EXPORT_SYMBOL_GPL(cs42l51_remove); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 9d06cf7f8876..a79343e8a54e 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -13,7 +13,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); -int cs42l51_remove(struct device *dev); +void cs42l51_remove(struct device *dev); int __maybe_unused cs42l51_suspend(struct device *dev); int __maybe_unused cs42l51_resume(struct device *dev); extern const struct of_device_id cs42l51_of_match[]; -- 2.30.2

2 5

[PATCH] ASoC: Intel: Unify HDAudio-ext bus initialization
by Cezary Rojewski 21 Oct '21

21 Oct '21

HDAudio-extended bus initialization parts are scattered throughout Intel ADSP drivers code. Gather them up in snd_hda_ext_bus_init() to provide unified initialization point. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/hdaudio_ext.h | 9 ++++++--- sound/hda/ext/hdac_ext_bus.c | 29 ++++++++++++++++++++--------- sound/soc/intel/skylake/skl.c | 9 +-------- sound/soc/sof/intel/hda-bus.c | 16 +++++++++------- sound/soc/sof/intel/hda.c | 5 +++-- sound/soc/sof/intel/hda.h | 3 ++- 6 files changed, 41 insertions(+), 30 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 375581634143..d1f6e9f7c057 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -2,11 +2,14 @@ #ifndef __SOUND_HDAUDIO_EXT_H #define __SOUND_HDAUDIO_EXT_H +#include <linux/pci.h> #include <sound/hdaudio.h> +#include <sound/hda_codec.h> -int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_bus_ops *ops, - const struct hdac_ext_bus_ops *ext_ops); +int snd_hda_ext_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const struct hdac_bus_ops *ops, + const struct hdac_ext_bus_ops *ext_ops, + const char *model); void snd_hdac_ext_bus_exit(struct hdac_bus *bus); int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 765c40a6ccba..a89e2e80ea4c 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -10,15 +10,17 @@ */ #include <linux/module.h> +#include <linux/pci.h> #include <linux/slab.h> #include <linux/io.h> #include <sound/hdaudio_ext.h> +#include <sound/hda_codec.h> MODULE_DESCRIPTION("HDA extended core"); MODULE_LICENSE("GPL v2"); /** - * snd_hdac_ext_bus_init - initialize a HD-audio extended bus + * snd_hda_ext_bus_init - initialize a HD-audio extended bus * @bus: the pointer to HDAC bus object * @dev: device pointer * @ops: bus verb operators @@ -26,28 +28,37 @@ MODULE_LICENSE("GPL v2"); * * Returns 0 if successful, or a negative error code. */ -int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_bus_ops *ops, - const struct hdac_ext_bus_ops *ext_ops) +int snd_hda_ext_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const struct hdac_bus_ops *ops, + const struct hdac_ext_bus_ops *ext_ops, + const char *model) { + struct hdac_bus *base = &bus->core; int ret; - ret = snd_hdac_bus_init(bus, dev, ops); + ret = snd_hdac_bus_init(base, &pdev->dev, ops); if (ret < 0) return ret; - bus->ext_ops = ext_ops; + base->ext_ops = ext_ops; /* FIXME: * Currently only one bus is supported, if there is device with more * buses, bus->idx should be greater than 0, but there needs to be a * reliable way to always assign same number. */ - bus->idx = 0; - bus->cmd_dma_state = true; + base->idx = 0; + base->cmd_dma_state = true; + base->use_posbuf = 1; + base->bdl_pos_adj = 0; + base->sync_write = 1; + bus->pci = pdev; + bus->modelname = model; + bus->mixer_assigned = -1; + mutex_init(&bus->prepare_mutex); return 0; } -EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); +EXPORT_SYMBOL_GPL(snd_hda_ext_bus_init); /** * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5b1a15e39912..95de41d14e56 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -886,16 +886,9 @@ static int skl_create(struct pci_dev *pci, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif - snd_hdac_ext_bus_init(bus, &pci->dev, NULL, ext_ops); - bus->use_posbuf = 1; + snd_hda_ext_bus_init(hbus, pci, NULL, ext_ops, "sklbus"); skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); - bus->bdl_pos_adj = 0; - - mutex_init(&hbus->prepare_mutex); - hbus->pci = pci; - hbus->mixer_assigned = -1; - hbus->modelname = "sklbus"; *rskl = skl; diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 30025d3c16b6..5d5081f80e88 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -8,6 +8,7 @@ // Authors: Keyon Jie <yang.jie(a)linux.intel.com> #include <linux/io.h> +#include <linux/pci.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include "../sof-priv.h" @@ -53,24 +54,25 @@ static const struct hdac_bus_ops bus_core_ops = { /* * This can be used for both with/without hda link support. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev) +void sof_hda_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const char *model) { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - snd_hdac_ext_bus_init(bus, dev, &bus_core_ops, sof_hda_ext_ops); + snd_hda_ext_bus_init(bus, pdev, &bus_core_ops, sof_hda_ext_ops, model); #else /* CONFIG_SND_SOC_SOF_HDA */ memset(bus, 0, sizeof(*bus)); - bus->dev = dev; + bus->core.dev = &pdev->dev; - INIT_LIST_HEAD(&bus->stream_list); + INIT_LIST_HEAD(&bus->core.stream_list); - bus->irq = -1; + bus->core.irq = -1; /* * There is only one HDA bus atm. keep the index as 0. * Need to fix when there are more than one HDA bus. */ - bus->idx = 0; + bus->core.idx = 0; - spin_lock_init(&bus->reg_lock); + spin_lock_init(&bus->core.reg_lock); #endif /* CONFIG_SND_SOC_SOF_HDA */ } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index fbc2421c77f8..03a68d286c7c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -609,8 +609,9 @@ static int hda_init(struct snd_sof_dev *sdev) bus = sof_to_bus(sdev); /* HDA bus init */ - sof_hda_bus_init(bus, &pci->dev); + sof_hda_bus_init(hbus, pci, hda_model); +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) bus->use_posbuf = 1; bus->bdl_pos_adj = 0; bus->sync_write = 1; @@ -619,7 +620,7 @@ static int hda_init(struct snd_sof_dev *sdev) hbus->pci = pci; hbus->mixer_assigned = -1; hbus->modelname = hda_model; - +#endif /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); bus->remap_addr = pci_ioremap_bar(pci, 0); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 1195018a1f4f..a4ec88f36512 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -635,7 +635,8 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev); /* * HDA bus operations. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev); +void sof_hda_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const char *model); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* -- 2.25.1

3 11

[PATCH] Revert "ASoC: rockchip: add config for rockchip dmaengine pcm register"
by Sugar Zhang 21 Oct '21

21 Oct '21

This reverts commit 75b31192fe6ad20b42276b20ee3bdf1493216d63. The original purpose of customized pcm was to config prealloc buffer size flexibly. but, we can do the same thing by soc-generic-dmaengine-pcm. And the generic one can generated the better config by querying DMA capabilities from dmaengine driver rather than the Hard-Coded one. e.g. the customized one: static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, ... the generic one: ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause && dma_caps.cmd_resume) hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; ... So, let's revert back to use the generic dmaengine pcm. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/Makefile | 3 +-- sound/soc/rockchip/rockchip_i2s.c | 3 +-- sound/soc/rockchip/rockchip_pcm.c | 44 --------------------------------------- sound/soc/rockchip/rockchip_pcm.h | 11 ---------- 4 files changed, 2 insertions(+), 59 deletions(-) delete mode 100644 sound/soc/rockchip/rockchip_pcm.c delete mode 100644 sound/soc/rockchip/rockchip_pcm.h diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 65e814d..05b078e 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,11 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o -snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 7e89f5b..a6d7656 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -20,7 +20,6 @@ #include <sound/dmaengine_pcm.h> #include "rockchip_i2s.h" -#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -756,7 +755,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = rockchip_pcm_platform_register(&pdev->dev); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); goto err_suspend; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c deleted file mode 100644 index 02254e4..0000000 --- a/sound/soc/rockchip/rockchip_pcm.c +++ /dev/null @@ -1,44 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#include <linux/device.h> -#include <linux/init.h> -#include <linux/module.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/dmaengine_pcm.h> - -#include "rockchip_pcm.h" - -static const struct snd_pcm_hardware snd_rockchip_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 52, - .buffer_bytes_max = 64 * 1024, - .fifo_size = 32, -}; - -static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { - .pcm_hardware = &snd_rockchip_hardware, - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .prealloc_buffer_size = 32 * 1024, -}; - -int rockchip_pcm_platform_register(struct device *dev) -{ - return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); -} -EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); - -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h deleted file mode 100644 index 7f00e2c..0000000 --- a/sound/soc/rockchip/rockchip_pcm.h +++ /dev/null @@ -1,11 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#ifndef _ROCKCHIP_PCM_H -#define _ROCKCHIP_PCM_H - -int rockchip_pcm_platform_register(struct device *dev); - -#endif -- 2.7.4

3 2

[PATCH 0/2] ASoC: qcom: sm8250: add support for TX and RX Macro dais
by Srinivas Kandagatla 21 Oct '21

21 Oct '21

This patchset adds support for testing WCD938X connected via TX and RX Macros on SM8250 MTP. Srinivas Kandagatla (2): ASoC: qcom: sm8250: add support for TX and RX Macro dais ASoC: qcom: sm8250: Add Jack support sound/soc/qcom/sm8250.c | 79 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 79 insertions(+) -- 2.21.0

2 3

a52 plugin not working with PipeWire
by GitHub issues - edited 21 Oct '21

21 Oct '21

alsa-project/alsa-plugins issue #37 was edited from sjnewbury: Using the lastest code since v1.2.5 I can successfully get AC3 output through HDMI when using speaker-test. Defaults use a huge buffer and gives terrible latency, but it works. PulseAudio also works fine with fairly low-latency using "tsched=1 tsched_buffer_size=32768 tsched_buffer_watermark=8192 fixed_latency_range=yes mmap=1". This doesn't need the rate kludge any more. When I try pipewire using mmap I get the below messages in the log, no audio: `spa.alsa: alsa-pcm a52hdmi: mmap_commit wrote 512 instead of 1024` With mmap disabled, no errors are reported, but still no sound. Is there a configuration for pipewire which works with the a52 plugin, a pipewire bug, or is this another a52 plugin bug? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/37 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

a52 plugin not working with PipeWire
by GitHub issues - opened 21 Oct '21

21 Oct '21

alsa-project/alsa-plugins issue #37 was opened from sjnewbury: Using the lastest code since v1.2.5 I can successfully get AC3 output through HDMI when using speaker-test. Defaults uses a huge buffer and gives terrible latency, but it works. PulseAudio also works fine with fairly low-latency using "tsched=1 tsched_buffer_size=32768 tsched_buffer_watermark=8192 fixed_latency_range=yes mmap=1". This doesn't need the rate kludge any more. When I try pipewire using mmap I get the below messages in the log, no audio: `spa.alsa: alsa-pcm a52hdmi: mmap_commit wrote 512 instead of 1024` With mmap disabled, no errors are reported, but still no sound. Is there a configuration for pipewire which works with the a52 plugin, a pipewire bug, or is this another a52 plugin bug? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/37 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: hda/realtek: Fix mic mute LED for the HP Spectre x360 14
by Johnathon Clark 21 Oct '21

21 Oct '21

On the 'HP Spectre x360 Convertible 14-ea0xx' the microphone mute led is controlled by GPIO 0x04. The speaker mute LED does not seem to be exposed by GPIO and is there not set. Signed-off-by: Johnathon Clark <john.clark(a)cantab.net> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 965b096f416f..79c6fb39864d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4355,6 +4355,17 @@ static void alc287_fixup_hp_gpio_led(struct hda_codec *codec, alc_fixup_hp_gpio_led(codec, action, 0x10, 0); } +static void alc245_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->micmute_led_polarity = 1; + } + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); +} + /* turn on/off mic-mute LED per capture hook via VREF change */ static int vref_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -6709,6 +6720,7 @@ enum { ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, ALC287_FIXUP_HP_GPIO_LED, ALC256_FIXUP_HP_HEADSET_MIC, + ALC245_FIXUP_HP_GPIO_LED, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, @@ -7333,6 +7345,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC245_FIXUP_HP_X360_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_x360_amp, + .chained = true, + .chain_id = ALC245_FIXUP_HP_GPIO_LED }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, @@ -8432,6 +8446,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_tongfang_reset_persistent_settings, }, + [ALC245_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_gpio_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { -- 2.33.1

2 1

[PATCH v2 0/2] Add code to manage DSP clocks and provide dts-binding document
by YC Hung 21 Oct '21

21 Oct '21

From: "yc.hung" <yc.hung(a)mediatek.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and device Tree communities the it will be merged to SOF tree and then merged into ALSA tree. It provides two patches, one is for mt8195 dsp clocks related. Another is for mt8195 dsp dts binding decription. YC Hung (2): ASoC: SOF: mediatek: Add mt8195 dsp clock support dt-bindings: dsp: mediatek: Add mt8195 DSP binding support .../bindings/dsp/mtk,mt8195-dsp.yaml | 139 +++++++++++++++ sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 164 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 29 ++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 23 ++- 5 files changed, 354 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h -- 2.18.0

1 2

[PATCH AUTOSEL 4.4 6/7] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 4727f5b80e76..4ee3458ad810 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -288,8 +288,9 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

1 0

[PATCH AUTOSEL 4.9 6/7] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 00c6af2ae1c2..f0e112906c68 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -389,8 +389,9 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

1 0

[PATCH AUTOSEL 4.14 8/9] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 778b42ba90b8..5ae72134159a 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -389,8 +389,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

1 0

[PATCH AUTOSEL 4.19 09/10] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 74244d8e2909..a65e8c0c630d 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -390,8 +390,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

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[PATCH AUTOSEL 5.4 09/11] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 7e7be8e4dcf9..87ba66dcfd47 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -395,8 +395,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

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[PATCH AUTOSEL 5.10 10/14] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index b98449fd92f3..522d1897659c 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -421,8 +421,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

1 0

[PATCH AUTOSEL 5.10 02/14] ALSA: hda: intel: Allow repeatedly probing on codec configuration errors
by Sasha Levin 21 Oct '21

21 Oct '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c0f1886de7e173865f1a0fa7680a1c07954a987f ] It seems that a few recent AMD systems show the codec configuration errors at the early boot, while loading the driver at a later stage works magically. Although the root cause of the error isn't clear, it's certainly not bad to allow retrying the codec probe in such a case if that helps. This patch adds the capability for retrying the probe upon codec probe errors on the certain AMD platforms. The probe_work is changed to a delayed work, and at the secondary call, it'll jump to the codec probing. Note that, not only adding the re-probing, this includes the behavior changes in the codec configuration function. Namely, snd_hda_codec_configure() won't unregister the codec at errors any longer. Instead, its caller, azx_codec_configure() unregisters the codecs with the probe failures *if* any codec has been successfully configured. If all codec probe failed, it doesn't unregister but let it re-probed -- which is the most case we're seeing and this patch tries to improve. Even if the driver doesn't re-probe or give up, it will go to the "free-all" error path, hence the leftover codecs shall be disabled / deleted in anyway. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190801 Link: https://lore.kernel.org/r/20211006141940.2897-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_bind.c | 20 +++++++++++--------- sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_controller.c | 24 ++++++++++++++++-------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 29 +++++++++++++++++++++++------ sound/pci/hda/hda_intel.h | 4 +++- 7 files changed, 56 insertions(+), 25 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 73827b7d17e0..cf530e9fb5f5 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -225,6 +225,7 @@ struct hda_codec { #endif /* misc flags */ + unsigned int configured:1; /* codec was configured */ unsigned int in_freeing:1; /* being released */ unsigned int registered:1; /* codec was registered */ unsigned int display_power_control:1; /* needs display power */ diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 17a25e453f60..4efbcc41fdfb 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -301,29 +301,31 @@ int snd_hda_codec_configure(struct hda_codec *codec) { int err; + if (codec->configured) + return 0; + if (is_generic_config(codec)) codec->probe_id = HDA_CODEC_ID_GENERIC; else codec->probe_id = 0; - err = snd_hdac_device_register(&codec->core); - if (err < 0) - return err; + if (!device_is_registered(&codec->core.dev)) { + err = snd_hdac_device_register(&codec->core); + if (err < 0) + return err; + } if (!codec->preset) codec_bind_module(codec); if (!codec->preset) { err = codec_bind_generic(codec); if (err < 0) { - codec_err(codec, "Unable to bind the codec\n"); - goto error; + codec_dbg(codec, "Unable to bind the codec\n"); + return err; } } + codec->configured = 1; return 0; - - error: - snd_hdac_device_unregister(&codec->core); - return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_configure); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4cec1bd77e6f..6dece719be66 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -791,6 +791,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->nids); remove_conn_list(codec); snd_hdac_regmap_exit(&codec->core); + codec->configured = 0; } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index b972d59eb1ec..3de7dc34def2 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include <sound/initval.h> #include "hda_controller.h" +#include "hda_local.h" #define CREATE_TRACE_POINTS #include "hda_controller_trace.h" @@ -1259,17 +1260,24 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs); int azx_codec_configure(struct azx *chip) { struct hda_codec *codec, *next; + int success = 0; - /* use _safe version here since snd_hda_codec_configure() deregisters - * the device upon error and deletes itself from the bus list. - */ - list_for_each_codec_safe(codec, next, &chip->bus) { - snd_hda_codec_configure(codec); + list_for_each_codec(codec, &chip->bus) { + if (!snd_hda_codec_configure(codec)) + success++; } - if (!azx_bus(chip)->num_codecs) - return -ENODEV; - return 0; + if (success) { + /* unregister failed codecs if any codec has been probed */ + list_for_each_codec_safe(codec, next, &chip->bus) { + if (!codec->configured) { + codec_err(codec, "Unable to configure, disabling\n"); + snd_hdac_device_unregister(&codec->core); + } + } + } + + return success ? 0 : -ENODEV; } EXPORT_SYMBOL_GPL(azx_codec_configure); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 68f9668788ea..324cba13c7ba 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_RETRY_PROBE (1 << 27) /* retry probe if no codec is configured */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4c8b281c3992..8bc27e7c0590 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -341,7 +341,8 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ - AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_RETRY_PROBE) /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ @@ -1758,7 +1759,7 @@ static void azx_check_snoop_available(struct azx *chip) static void azx_probe_work(struct work_struct *work) { - struct hda_intel *hda = container_of(work, struct hda_intel, probe_work); + struct hda_intel *hda = container_of(work, struct hda_intel, probe_work.work); azx_probe_continue(&hda->chip); } @@ -1867,7 +1868,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, } /* continue probing in work context as may trigger request module */ - INIT_WORK(&hda->probe_work, azx_probe_work); + INIT_DELAYED_WORK(&hda->probe_work, azx_probe_work); *rchip = chip; @@ -2202,7 +2203,7 @@ static int azx_probe(struct pci_dev *pci, #endif if (schedule_probe) - schedule_work(&hda->probe_work); + schedule_delayed_work(&hda->probe_work, 0); dev++; if (chip->disabled) @@ -2290,6 +2291,11 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + if (chip->disabled || hda->init_failed) + return -EIO; + if (hda->probe_retry) + goto probe_retry; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; @@ -2351,10 +2357,20 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif + + probe_retry: if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); - if (err < 0) + if (err) { + if ((chip->driver_caps & AZX_DCAPS_RETRY_PROBE) && + ++hda->probe_retry < 60) { + schedule_delayed_work(&hda->probe_work, + msecs_to_jiffies(1000)); + return 0; /* keep things up */ + } + dev_err(chip->card->dev, "Cannot probe codecs, giving up\n"); goto out_free; + } } err = snd_card_register(chip->card); @@ -2384,6 +2400,7 @@ static int azx_probe_continue(struct azx *chip) display_power(chip, false); complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; + hda->probe_retry = 0; return 0; } @@ -2409,7 +2426,7 @@ static void azx_remove(struct pci_dev *pci) * device during cancel_work_sync() call. */ device_unlock(&pci->dev); - cancel_work_sync(&hda->probe_work); + cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); snd_card_free(card); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 3fb119f09040..0f39418f9328 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -14,7 +14,7 @@ struct hda_intel { /* sync probing */ struct completion probe_wait; - struct work_struct probe_work; + struct delayed_work probe_work; /* card list (for power_save trigger) */ struct list_head list; @@ -30,6 +30,8 @@ struct hda_intel { unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ + + int probe_retry; /* being probe-retry */ }; #endif -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 19/26] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 062da7a7a586..f7bd6e2db085 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -421,8 +421,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 10/26] ALSA: hda: intel: Allow repeatedly probing on codec configuration errors
by Sasha Levin 21 Oct '21

21 Oct '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c0f1886de7e173865f1a0fa7680a1c07954a987f ] It seems that a few recent AMD systems show the codec configuration errors at the early boot, while loading the driver at a later stage works magically. Although the root cause of the error isn't clear, it's certainly not bad to allow retrying the codec probe in such a case if that helps. This patch adds the capability for retrying the probe upon codec probe errors on the certain AMD platforms. The probe_work is changed to a delayed work, and at the secondary call, it'll jump to the codec probing. Note that, not only adding the re-probing, this includes the behavior changes in the codec configuration function. Namely, snd_hda_codec_configure() won't unregister the codec at errors any longer. Instead, its caller, azx_codec_configure() unregisters the codecs with the probe failures *if* any codec has been successfully configured. If all codec probe failed, it doesn't unregister but let it re-probed -- which is the most case we're seeing and this patch tries to improve. Even if the driver doesn't re-probe or give up, it will go to the "free-all" error path, hence the leftover codecs shall be disabled / deleted in anyway. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190801 Link: https://lore.kernel.org/r/20211006141940.2897-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_bind.c | 20 +++++++++++--------- sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_controller.c | 24 ++++++++++++++++-------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 29 +++++++++++++++++++++++------ sound/pci/hda/hda_intel.h | 4 +++- 7 files changed, 56 insertions(+), 25 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 2e8d51937acd..47d2cad21b56 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -225,6 +225,7 @@ struct hda_codec { #endif /* misc flags */ + unsigned int configured:1; /* codec was configured */ unsigned int in_freeing:1; /* being released */ unsigned int registered:1; /* codec was registered */ unsigned int display_power_control:1; /* needs display power */ diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index e8dee24c309d..50a58fb5ad9c 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -304,29 +304,31 @@ int snd_hda_codec_configure(struct hda_codec *codec) { int err; + if (codec->configured) + return 0; + if (is_generic_config(codec)) codec->probe_id = HDA_CODEC_ID_GENERIC; else codec->probe_id = 0; - err = snd_hdac_device_register(&codec->core); - if (err < 0) - return err; + if (!device_is_registered(&codec->core.dev)) { + err = snd_hdac_device_register(&codec->core); + if (err < 0) + return err; + } if (!codec->preset) codec_bind_module(codec); if (!codec->preset) { err = codec_bind_generic(codec); if (err < 0) { - codec_err(codec, "Unable to bind the codec\n"); - goto error; + codec_dbg(codec, "Unable to bind the codec\n"); + return err; } } + codec->configured = 1; return 0; - - error: - snd_hdac_device_unregister(&codec->core); - return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_configure); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7a717e151156..8afcce6478cd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -791,6 +791,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->nids); remove_conn_list(codec); snd_hdac_regmap_exit(&codec->core); + codec->configured = 0; } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ca2f2ecd1488..5a49ee4f6ce0 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include <sound/initval.h> #include "hda_controller.h" +#include "hda_local.h" #define CREATE_TRACE_POINTS #include "hda_controller_trace.h" @@ -1259,17 +1260,24 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs); int azx_codec_configure(struct azx *chip) { struct hda_codec *codec, *next; + int success = 0; - /* use _safe version here since snd_hda_codec_configure() deregisters - * the device upon error and deletes itself from the bus list. - */ - list_for_each_codec_safe(codec, next, &chip->bus) { - snd_hda_codec_configure(codec); + list_for_each_codec(codec, &chip->bus) { + if (!snd_hda_codec_configure(codec)) + success++; } - if (!azx_bus(chip)->num_codecs) - return -ENODEV; - return 0; + if (success) { + /* unregister failed codecs if any codec has been probed */ + list_for_each_codec_safe(codec, next, &chip->bus) { + if (!codec->configured) { + codec_err(codec, "Unable to configure, disabling\n"); + snd_hdac_device_unregister(&codec->core); + } + } + } + + return success ? 0 : -ENODEV; } EXPORT_SYMBOL_GPL(azx_codec_configure); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 68f9668788ea..324cba13c7ba 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_RETRY_PROBE (1 << 27) /* retry probe if no codec is configured */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0062c18b646a..89f135a6a1f6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -307,7 +307,8 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ - AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_RETRY_PROBE) /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ @@ -1730,7 +1731,7 @@ static void azx_check_snoop_available(struct azx *chip) static void azx_probe_work(struct work_struct *work) { - struct hda_intel *hda = container_of(work, struct hda_intel, probe_work); + struct hda_intel *hda = container_of(work, struct hda_intel, probe_work.work); azx_probe_continue(&hda->chip); } @@ -1839,7 +1840,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, } /* continue probing in work context as may trigger request module */ - INIT_WORK(&hda->probe_work, azx_probe_work); + INIT_DELAYED_WORK(&hda->probe_work, azx_probe_work); *rchip = chip; @@ -2170,7 +2171,7 @@ static int azx_probe(struct pci_dev *pci, #endif if (schedule_probe) - schedule_work(&hda->probe_work); + schedule_delayed_work(&hda->probe_work, 0); dev++; if (chip->disabled) @@ -2256,6 +2257,11 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + if (chip->disabled || hda->init_failed) + return -EIO; + if (hda->probe_retry) + goto probe_retry; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; @@ -2317,10 +2323,20 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif + + probe_retry: if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); - if (err < 0) + if (err) { + if ((chip->driver_caps & AZX_DCAPS_RETRY_PROBE) && + ++hda->probe_retry < 60) { + schedule_delayed_work(&hda->probe_work, + msecs_to_jiffies(1000)); + return 0; /* keep things up */ + } + dev_err(chip->card->dev, "Cannot probe codecs, giving up\n"); goto out_free; + } } err = snd_card_register(chip->card); @@ -2350,6 +2366,7 @@ static int azx_probe_continue(struct azx *chip) display_power(chip, false); complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; + hda->probe_retry = 0; return 0; } @@ -2375,7 +2392,7 @@ static void azx_remove(struct pci_dev *pci) * device during cancel_work_sync() call. */ device_unlock(&pci->dev); - cancel_work_sync(&hda->probe_work); + cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); snd_card_free(card); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 3fb119f09040..0f39418f9328 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -14,7 +14,7 @@ struct hda_intel { /* sync probing */ struct completion probe_wait; - struct work_struct probe_work; + struct delayed_work probe_work; /* card list (for power_save trigger) */ struct list_head list; @@ -30,6 +30,8 @@ struct hda_intel { unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ + + int probe_retry; /* being probe-retry */ }; #endif -- 2.33.0

1 0

Re: linux-next: Tree for Oct 20 [sound/soc/codecs/snd-soc-cs35l41-spi.ko]
by Randy Dunlap 20 Oct '21

20 Oct '21

On 10/20/21 12:39 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20211019: > on i386: WARNING: unmet direct dependencies detected for SND_SOC_CS35L41_SPI Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SPI_MASTER [=n] Selected by [m]: - SND_SOC_AMD_VANGOGH_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP5x [=m] && I2C [=y] KCONFIG_SEED=0x4F150562 and then ERROR: modpost: "__devm_regmap_init_spi" [sound/soc/codecs/snd-soc-cs35l41-spi.ko] undefined! ERROR: modpost: "__spi_register_driver" [sound/soc/codecs/snd-soc-cs35l41-spi.ko] undefined! ERROR: modpost: "spi_setup" [sound/soc/codecs/snd-soc-cs35l41-spi.ko] undefined! Full randconfig file is attached. -- ~Randy

1 0

[PATCH] ALSA: mixer: oss: Fix racy access to slots
by Takashi Iwai 20 Oct '21

20 Oct '21

The OSS mixer can reassign the mapping slots dynamically via proc file. Although the addition and deletion of those slots are protected by mixer->reg_mutex, the access to slots aren't, hence this may cause UAF when the slots in use are deleted concurrently. This patch applies the mixer->reg_mutex in all appropriate code paths (i.e. the ioctl functions) that may access slots. Reported-by: syzbot+9988f17cf72a1045a189(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/00000000000036adc005ceca9175@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/mixer_oss.c | 44 ++++++++++++++++++++++++++++---------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 6a5abdd4271b..d5ddc154a735 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -130,11 +130,13 @@ static int snd_mixer_oss_devmask(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; if (pslot->put_volume || pslot->put_recsrc) result |= 1 << chn; } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -146,11 +148,13 @@ static int snd_mixer_oss_stereodevs(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; if (pslot->put_volume && pslot->stereo) result |= 1 << chn; } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -161,6 +165,7 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ result = mixer->mask_recsrc; } else { @@ -172,6 +177,7 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) result |= 1 << chn; } } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -182,12 +188,12 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ - int err; unsigned int index; - err = mixer->get_recsrc(fmixer, &index); - if (err < 0) - return err; + result = mixer->get_recsrc(fmixer, &index); + if (result < 0) + goto unlock; result = 1 << index; } else { struct snd_mixer_oss_slot *pslot; @@ -202,7 +208,10 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) } } } - return mixer->oss_recsrc = result; + mixer->oss_recsrc = result; + unlock: + mutex_unlock(&mixer->reg_mutex); + return result; } static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsrc) @@ -215,6 +224,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); if (mixer->get_recsrc && mixer->put_recsrc) { /* exclusive input */ if (recsrc & ~mixer->oss_recsrc) recsrc &= ~mixer->oss_recsrc; @@ -240,6 +250,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr } } } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -251,6 +262,7 @@ static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot) if (mixer == NULL || slot > 30) return -EIO; + mutex_lock(&mixer->reg_mutex); pslot = &mixer->slots[slot]; left = pslot->volume[0]; right = pslot->volume[1]; @@ -258,15 +270,21 @@ static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot) result = pslot->get_volume(fmixer, pslot, &left, &right); if (!pslot->stereo) right = left; - if (snd_BUG_ON(left < 0 || left > 100)) - return -EIO; - if (snd_BUG_ON(right < 0 || right > 100)) - return -EIO; + if (snd_BUG_ON(left < 0 || left > 100)) { + result = -EIO; + goto unlock; + } + if (snd_BUG_ON(right < 0 || right > 100)) { + result = -EIO; + goto unlock; + } if (result >= 0) { pslot->volume[0] = left; pslot->volume[1] = right; result = (left & 0xff) | ((right & 0xff) << 8); } + unlock: + mutex_unlock(&mixer->reg_mutex); return result; } @@ -279,6 +297,7 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, if (mixer == NULL || slot > 30) return -EIO; + mutex_lock(&mixer->reg_mutex); pslot = &mixer->slots[slot]; if (left > 100) left = 100; @@ -289,10 +308,13 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, if (pslot->put_volume) result = pslot->put_volume(fmixer, pslot, left, right); if (result < 0) - return result; + goto unlock; pslot->volume[0] = left; pslot->volume[1] = right; - return (left & 0xff) | ((right & 0xff) << 8); + result = (left & 0xff) | ((right & 0xff) << 8); + unlock: + mutex_lock(&mixer->reg_mutex); + return result; } static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int cmd, unsigned long arg) -- 2.26.2

2 1

WC vs UC mappings in snd_dma_sg_alloc()
by sanekf＠nxt.ru 20 Oct '21

20 Oct '21

Hi I've stumbled across this code in sound/core/sgbuf.c: 66 static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) 67 { < ... > 80 if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { 81 type = SNDRV_DMA_TYPE_DEV_WC; 82 #ifdef pgprot_noncached 83 prot = pgprot_noncached(PAGE_KERNEL); 84 #endif 85 } < ... > 131 area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); Does not this violate x86 rules about using the same memory mapping type for all mappings? It seems that the following patch should fix it (only compile tested - my x86 PCs are either without Linux or without sound, and probably in practice this might not trigger any problems since both WC and UC-minus are incoherent types): ----------------------------- ALSA: memalloc: duly use pgprot_writecombine() for WC mapping x86 has strict rules about not having memory type aliasing (Documentation/x86/pat.rst). snd_dma_sg_alloc() violates them by mapping first as WC (with set_memory_wc()) and then as UC- (with pgprot_noncached() + vmap()). Switching to pgprot_writecombine() should fix this. Signed-off-by: Aleksandr Fedorov <halcien(a)gmail.com> diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 8352a5cdb19f..670b30c3b6e5 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -79,9 +79,7 @@ static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) return NULL; if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { type = SNDRV_DMA_TYPE_DEV_WC; -#ifdef pgprot_noncached - prot = pgprot_noncached(PAGE_KERNEL); -#endif + prot = pgprot_writecombine(PAGE_KERNEL); } sgbuf->dev = dmab->dev.dev; pages = snd_sgbuf_aligned_pages(size);

2 1

[PATCH v3 0/2] arm: ep93xx: CCF conversion
by Alexander Sverdlin 20 Oct '21

20 Oct '21

This series series of patches converts ep93xx to Common Clock Framework. It consists of preparation patches to use clk_prepare_enable where it is needed, instead of clk_enable used in ep93xx drivers prior to CCF and a patch converting mach-ep93xx/clock.c to CCF. Link: https://lore.kernel.org/patchwork/cover/1445563/ Link: https://lore.kernel.org/patchwork/patch/1435884/ v2->v3: Rebased the series onto v5.15-rc5 v1->v2: - added SoB Alexander Sverdlin (1): ASoC: cirrus: i2s: Prepare clock before using it Nikita Shubin (1): ep93xx: clock: convert in-place to COMMON_CLK arch/arm/Kconfig | 2 +- arch/arm/mach-ep93xx/clock.c | 975 +++++++++++++++++++--------------- arch/arm/mach-ep93xx/core.c | 2 +- arch/arm/mach-ep93xx/soc.h | 42 +- sound/soc/cirrus/ep93xx-i2s.c | 12 +- 5 files changed, 587 insertions(+), 446 deletions(-) -- 2.33.0

2 2

[PATCH v3 0/6] Multiple headphone codec driver support
by Brent Lu 20 Oct '21

20 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched V3 Changes: - upstreamd from SOF github, PR#3200 - use new compatiable IDs to shrink the enumerate table of BYT and CHT - add 'const' to snd_soc_acpi_codecs structures Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s Pierre-Louis Bossart (3): ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs include/sound/soc-acpi.h | 3 + sound/soc/intel/boards/sof_rt5682.c | 34 ++------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++- .../intel/common/soc-acpi-intel-bxt-match.c | 2 +- .../intel/common/soc-acpi-intel-byt-match.c | 68 +++++++----------- .../intel/common/soc-acpi-intel-cht-match.c | 69 +++++++------------ .../intel/common/soc-acpi-intel-cml-match.c | 8 +-- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-jsl-match.c | 43 ++++-------- .../intel/common/soc-acpi-intel-kbl-match.c | 12 ++-- .../intel/common/soc-acpi-intel-skl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++- sound/soc/soc-acpi.c | 24 ++++++- 13 files changed, 119 insertions(+), 170 deletions(-) -- 2.25.1

3 8

[syzbot] KASAN: use-after-free Read in snd_mixer_oss_get_volume1
by syzbot 20 Oct '21

20 Oct '21

Hello, syzbot found the following issue on: HEAD commit: 8fe31e0995f0 Merge tag 'gpio-fixes-for-v5.15-rc6' of git:/.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=16a58d94b00000 kernel config: https://syzkaller.appspot.com/x/.config?x=bab9d35f204746a7 dashboard link: https://syzkaller.appspot.com/bug?extid=9988f17cf72a1045a189 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 userspace arch: i386 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=10251d58b00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17039278b00000 Bisection is inconclusive: the issue happens on the oldest tested release. bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=167855f4b00000 final oops: https://syzkaller.appspot.com/x/report.txt?x=157855f4b00000 console output: https://syzkaller.appspot.com/x/log.txt?x=117855f4b00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+9988f17cf72a1045a189(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_mixer_oss_get_volume1+0x574/0x610 sound/core/oss/mixer_oss.c:600 Read of size 4 at addr ffff888070afc984 by task syz-executor166/6533 CPU: 0 PID: 6533 Comm: syz-executor166 Not tainted 5.15.0-rc5-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 print_address_description.constprop.0.cold+0x6c/0x309 mm/kasan/report.c:256 __kasan_report mm/kasan/report.c:442 [inline] kasan_report.cold+0x83/0xdf mm/kasan/report.c:459 snd_mixer_oss_get_volume1+0x574/0x610 sound/core/oss/mixer_oss.c:600 snd_mixer_oss_get_volume sound/core/oss/mixer_oss.c:258 [inline] snd_mixer_oss_ioctl1+0x249/0x1880 sound/core/oss/mixer_oss.c:358 snd_mixer_oss_ioctl_compat+0x45/0x60 sound/core/oss/mixer_oss.c:391 __do_compat_sys_ioctl+0x1c7/0x290 fs/ioctl.c:972 do_syscall_32_irqs_on arch/x86/entry/common.c:112 [inline] __do_fast_syscall_32+0x65/0xf0 arch/x86/entry/common.c:178 do_fast_syscall_32+0x2f/0x70 arch/x86/entry/common.c:203 entry_SYSENTER_compat_after_hwframe+0x4d/0x5c RIP: 0023:0xf7f42549 Code: 03 74 c0 01 10 05 03 74 b8 01 10 06 03 74 b4 01 10 07 03 74 b0 01 10 08 03 74 d8 01 00 00 00 00 00 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 8d b4 26 00 00 00 00 8d b4 26 00 00 00 00 RSP: 002b:00000000ffa708bc EFLAGS: 00000246 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 0000000080254d18 RDX: 0000000000000000 RSI: 0000000003700000 RDI: 0000000001000000 RBP: 0000000000000003 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Allocated by task 6526: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:38 kasan_set_track mm/kasan/common.c:46 [inline] set_alloc_info mm/kasan/common.c:434 [inline] ____kasan_kmalloc mm/kasan/common.c:513 [inline] ____kasan_kmalloc mm/kasan/common.c:472 [inline] __kasan_kmalloc+0xa4/0xd0 mm/kasan/common.c:522 kmalloc include/linux/slab.h:591 [inline] snd_mixer_oss_build_input+0x656/0xe30 sound/core/oss/mixer_oss.c:1093 snd_mixer_oss_proc_write+0x384/0x5c0 sound/core/oss/mixer_oss.c:1233 snd_info_text_entry_release+0xfa/0x260 sound/core/info.c:408 close_pdeo.part.0+0xdc/0x2e0 fs/proc/inode.c:244 close_pdeo fs/proc/inode.c:563 [inline] proc_reg_release+0x2e9/0x360 fs/proc/inode.c:562 __fput+0x288/0x9f0 fs/file_table.c:280 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 exit_task_work include/linux/task_work.h:32 [inline] do_exit+0xbae/0x2a30 kernel/exit.c:825 do_group_exit+0x125/0x310 kernel/exit.c:922 __do_sys_exit_group kernel/exit.c:933 [inline] __se_sys_exit_group kernel/exit.c:931 [inline] __ia32_sys_exit_group+0x3a/0x50 kernel/exit.c:931 do_syscall_32_irqs_on arch/x86/entry/common.c:112 [inline] __do_fast_syscall_32+0x65/0xf0 arch/x86/entry/common.c:178 do_fast_syscall_32+0x2f/0x70 arch/x86/entry/common.c:203 entry_SYSENTER_compat_after_hwframe+0x4d/0x5c Freed by task 6531: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:38 kasan_set_track+0x1c/0x30 mm/kasan/common.c:46 kasan_set_free_info+0x20/0x30 mm/kasan/generic.c:360 ____kasan_slab_free mm/kasan/common.c:366 [inline] ____kasan_slab_free mm/kasan/common.c:328 [inline] __kasan_slab_free+0xff/0x130 mm/kasan/common.c:374 kasan_slab_free include/linux/kasan.h:230 [inline] slab_free_hook mm/slub.c:1700 [inline] slab_free_freelist_hook+0x81/0x190 mm/slub.c:1725 slab_free mm/slub.c:3483 [inline] kfree+0xe4/0x530 mm/slub.c:4543 snd_mixer_oss_slot_free+0xf3/0x130 sound/core/oss/mixer_oss.c:951 mixer_slot_clear+0x5a/0xb0 sound/core/oss/mixer_oss.c:959 snd_mixer_oss_build_input+0x7f0/0xe30 sound/core/oss/mixer_oss.c:1101 snd_mixer_oss_proc_write+0x384/0x5c0 sound/core/oss/mixer_oss.c:1233 snd_info_text_entry_release+0xfa/0x260 sound/core/info.c:408 close_pdeo.part.0+0xdc/0x2e0 fs/proc/inode.c:244 close_pdeo fs/proc/inode.c:563 [inline] proc_reg_release+0x2e9/0x360 fs/proc/inode.c:562 __fput+0x288/0x9f0 fs/file_table.c:280 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 exit_task_work include/linux/task_work.h:32 [inline] do_exit+0xbae/0x2a30 kernel/exit.c:825 do_group_exit+0x125/0x310 kernel/exit.c:922 __do_sys_exit_group kernel/exit.c:933 [inline] __se_sys_exit_group kernel/exit.c:931 [inline] __ia32_sys_exit_group+0x3a/0x50 kernel/exit.c:931 do_syscall_32_irqs_on arch/x86/entry/common.c:112 [inline] __do_fast_syscall_32+0x65/0xf0 arch/x86/entry/common.c:178 do_fast_syscall_32+0x2f/0x70 arch/x86/entry/common.c:203 entry_SYSENTER_compat_after_hwframe+0x4d/0x5c The buggy address belongs to the object at ffff888070afc980 which belongs to the cache kmalloc-96 of size 96 The buggy address is located 4 bytes inside of 96-byte region [ffff888070afc980, ffff888070afc9e0) The buggy address belongs to the page: page:ffffea0001c2bf00 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x70afc flags: 0xfff00000000200(slab|node=0|zone=1|lastcpupid=0x7ff) raw: 00fff00000000200 ffffea0000725740 0000000700000007 ffff888010c41780 raw: 0000000000000000 0000000000200020 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 0, migratetype Unmovable, gfp_mask 0x12c40(GFP_NOFS|__GFP_NOWARN|__GFP_NORETRY), pid 4486, ts 39889903663, free_ts 39887746705 prep_new_page mm/page_alloc.c:2424 [inline] get_page_from_freelist+0xa72/0x2f80 mm/page_alloc.c:4153 __alloc_pages+0x1b2/0x500 mm/page_alloc.c:5375 alloc_pages+0x1a7/0x300 mm/mempolicy.c:2197 alloc_slab_page mm/slub.c:1763 [inline] allocate_slab mm/slub.c:1900 [inline] new_slab+0x319/0x490 mm/slub.c:1963 ___slab_alloc+0x921/0xfe0 mm/slub.c:2994 __slab_alloc.constprop.0+0x4d/0xa0 mm/slub.c:3081 slab_alloc_node mm/slub.c:3172 [inline] slab_alloc mm/slub.c:3214 [inline] __kmalloc+0x305/0x320 mm/slub.c:4387 kmalloc include/linux/slab.h:596 [inline] kzalloc include/linux/slab.h:721 [inline] tomoyo_encode2.part.0+0xe9/0x3a0 security/tomoyo/realpath.c:45 tomoyo_encode2 security/tomoyo/realpath.c:31 [inline] tomoyo_encode+0x28/0x50 security/tomoyo/realpath.c:80 tomoyo_realpath_from_path+0x186/0x620 security/tomoyo/realpath.c:288 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x272/0x380 security/tomoyo/file.c:771 tomoyo_file_open security/tomoyo/tomoyo.c:311 [inline] tomoyo_file_open+0xa3/0xd0 security/tomoyo/tomoyo.c:306 security_file_open+0x45/0xb0 security/security.c:1634 do_dentry_open+0x353/0x11d0 fs/open.c:809 do_open fs/namei.c:3428 [inline] path_openat+0x1c9a/0x2740 fs/namei.c:3561 do_filp_open+0x1aa/0x400 fs/namei.c:3588 page last free stack trace: reset_page_owner include/linux/page_owner.h:24 [inline] free_pages_prepare mm/page_alloc.c:1338 [inline] free_pcp_prepare+0x2c5/0x780 mm/page_alloc.c:1389 free_unref_page_prepare mm/page_alloc.c:3315 [inline] free_unref_page+0x19/0x690 mm/page_alloc.c:3394 __vunmap+0x783/0xb70 mm/vmalloc.c:2621 free_work+0x58/0x70 mm/vmalloc.c:95 process_one_work+0x9bf/0x16b0 kernel/workqueue.c:2297 worker_thread+0x658/0x11f0 kernel/workqueue.c:2444 kthread+0x3e5/0x4d0 kernel/kthread.c:319 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 Memory state around the buggy address: ffff888070afc880: 00 00 00 00 00 00 00 00 00 00 fc fc fc fc fc fc ffff888070afc900: fa fb fb fb fb fb fb fb fb fb fb fb fc fc fc fc >ffff888070afc980: fa fb fb fb fb fb fb fb fb fb fb fb fc fc fc fc ^ ffff888070afca00: fa fb fb fb fb fb fb fb fb fb fb fb fc fc fc fc ffff888070afca80: fa fb fb fb fb fb fb fb fb fb fb fb fc fc fc fc ================================================================== ---------------- Code disassembly (best guess): 0: 03 74 c0 01 add 0x1(%rax,%rax,8),%esi 4: 10 05 03 74 b8 01 adc %al,0x1b87403(%rip) # 0x1b8740d a: 10 06 adc %al,(%rsi) c: 03 74 b4 01 add 0x1(%rsp,%rsi,4),%esi 10: 10 07 adc %al,(%rdi) 12: 03 74 b0 01 add 0x1(%rax,%rsi,4),%esi 16: 10 08 adc %cl,(%rax) 18: 03 74 d8 01 add 0x1(%rax,%rbx,8),%esi 1c: 00 00 add %al,(%rax) 1e: 00 00 add %al,(%rax) 20: 00 51 52 add %dl,0x52(%rcx) 23: 55 push %rbp 24: 89 e5 mov %esp,%ebp 26: 0f 34 sysenter 28: cd 80 int $0x80 * 2a: 5d pop %rbp <-- trapping instruction 2b: 5a pop %rdx 2c: 59 pop %rcx 2d: c3 retq 2e: 90 nop 2f: 90 nop 30: 90 nop 31: 90 nop 32: 8d b4 26 00 00 00 00 lea 0x0(%rsi,%riz,1),%esi 39: 8d b4 26 00 00 00 00 lea 0x0(%rsi,%riz,1),%esi --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[PATCH alsa-utils 1/2] alsamixer: Check the availability of mouse
by Takashi Iwai 20 Oct '21

20 Oct '21

Let's check the availiabiy via has_mouse(). Otherwise the program aborts unexpectedly just focusing on my rxvt terminal :-( Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsamixer/mainloop.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsamixer/mainloop.c b/alsamixer/mainloop.c index e67c0f821e8a..cdbe28e390fd 100644 --- a/alsamixer/mainloop.c +++ b/alsamixer/mainloop.c @@ -50,7 +50,7 @@ void initialize_curses(bool use_color, bool use_mouse) #endif window_size_changed(); /* update screen_lines/cols */ init_colors(use_color); - if (use_mouse) + if (use_mouse && has_mouse()) mousemask(ALL_MOUSE_EVENTS, NULL); snd_lib_error_set_handler(black_hole_error_handler); } -- 2.26.2

1 1

[PATCH] soundwire: qcom: add debugfs entry for soundwire register dump
by Srinivas Kandagatla 20 Oct '21

20 Oct '21

For debug purposes add an entry in debugfs to dump Qualcomm SoundWire Controller registers. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 0ef79d60e88e..a317bea2d42d 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -7,6 +7,7 @@ #include <linux/io.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/debugfs.h> #include <linux/of.h> #include <linux/of_irq.h> #include <linux/of_device.h> @@ -83,6 +84,7 @@ #define SWRM_DP_PORT_HCTRL_BANK(n, m) (0x1134 + 0x100 * (n - 1) + 0x40 * m) #define SWRM_DP_BLOCK_CTRL3_BANK(n, m) (0x1138 + 0x100 * (n - 1) + 0x40 * m) #define SWRM_DIN_DPn_PCM_PORT_CTRL(n) (0x1054 + 0x100 * (n - 1)) +#define SWR_MSTR_MAX_REG_ADDR (0x1740) #define SWRM_DP_PORT_CTRL_EN_CHAN_SHFT 0x18 #define SWRM_DP_PORT_CTRL_OFFSET2_SHFT 0x10 @@ -127,6 +129,9 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs; +#endif struct completion broadcast; struct completion enumeration; struct work_struct slave_work; @@ -1192,6 +1197,22 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) return 0; } +#ifdef CONFIG_DEBUG_FS +static int swrm_reg_show(struct seq_file *s_file, void *data) +{ + struct qcom_swrm_ctrl *swrm = s_file->private; + int reg, reg_val; + + for (reg = 0; reg <= SWR_MSTR_MAX_REG_ADDR; reg += 4) { + swrm->reg_read(swrm, reg, &reg_val); + seq_printf(s_file, "0x%.3x: 0x%.2x\n", reg, reg_val); + } + + return 0; +} +DEFINE_SHOW_ATTRIBUTE(swrm_reg); +#endif + static int qcom_swrm_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; @@ -1302,6 +1323,12 @@ static int qcom_swrm_probe(struct platform_device *pdev) (ctrl->version >> 24) & 0xff, (ctrl->version >> 16) & 0xff, ctrl->version & 0xffff); +#ifdef CONFIG_DEBUG_FS + ctrl->debugfs = debugfs_create_dir("qualcomm-sdw", ctrl->bus.debugfs); + debugfs_create_file("qualcomm-registers", 0400, ctrl->debugfs, ctrl, + &swrm_reg_fops); +#endif + return 0; err_master_add: -- 2.21.0

3 4

[PATCH] soundwire: bus: stop dereferencing invalid slave pointer
by Srinivas Kandagatla 20 Oct '21

20 Oct '21

Slave pointer is invalid after end of list iteration, using this would result in below Memory abort. Unable to handle kernel NULL pointer dereference at virtual address 0000000000000004 ... Call trace: __dev_printk+0x34/0x7c _dev_warn+0x6c/0x90 sdw_bus_exit_clk_stop+0x194/0x1d0 swrm_runtime_resume+0x13c/0x238 pm_generic_runtime_resume+0x2c/0x48 __rpm_callback+0x44/0x150 rpm_callback+0x6c/0x78 rpm_resume+0x314/0x558 rpm_resume+0x378/0x558 rpm_resume+0x378/0x558 __pm_runtime_resume+0x3c/0x88 Use bus->dev instead to print this error message. Fixes: b50bb8ba369cd ("soundwire: bus: handle -ENODATA errors in clock stop/start sequences") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 1b115734a8f6..67369e941d0d 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1110,7 +1110,7 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) if (!simple_clk_stop) { ret = sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); if (ret < 0) - dev_warn(&slave->dev, "clock stop deprepare wait failed:%d\n", ret); + dev_warn(bus->dev, "clock stop deprepare wait failed:%d\n", ret); } list_for_each_entry(slave, &bus->slaves, node) { -- 2.21.0

3 2

[PATCH 0/2] Add code to manage DSP clocks and provide dts-binding document
by YC Hung 20 Oct '21

20 Oct '21

From: "yc.hung" <yc.hung(a)mediatek.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and device Tree communities the it will be merged to SOF tree and then merged into ALSA tree. It provides two patches, one is for mt8195 dsp clocks related. Another is for mt8195 dsp dts binding decription. YC Hung (2): ASoC: SOF: mediatek: Add mt8195 dsp clock support dt-bindings: dsp: mediatek: Add mt8195 DSP binding support .../bindings/dsp/mtk,mt8195-dsp.yaml | 138 +++++++++++++++ sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 164 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 29 ++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 23 ++- 5 files changed, 353 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h -- 2.18.0

2 3

[RFC PATCH v2 0/3] Add support for Legion 7 16ACHg6 laptop
by Lucas Tanure 20 Oct '21

20 Oct '21

Hi, I would like to get some guidance about this solution to support the 16ACHg6 laptop. Hardware: - The 16ACHg6 laptop has two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - To load the firmware for CS35L41, this solution will require the wm_adsp library, which requires regmap, header definitions and register tables. - The HDA machine driver will find the registered ASoC by its dai name. - To minimize the duplication of the code, the HDA will call snd_soc_dai_ops from the ASoC driver. Notes: - This is a work in progress, so the code is not functional, its only intent is to demonstrate the overall solution Changes from V1: - Split into multiple patches, separating ACPI, ASoC and HDA patches - Removed CS35L41 exported functions, moved code to snd_soc_dai_ops - Patch realtek uses dai ops functions Lucas Tanure (3): sound: cs35l41: Allow HDA systems to use CS35l41 ASoC driver ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Speakers ACPI / scan: Create platform device for INT3515 ACPI nodes drivers/acpi/scan.c | 1 + drivers/platform/x86/i2c-multi-instantiate.c | 7 + include/sound/cs35l41.h | 1 + sound/pci/hda/patch_realtek.c | 109 ++++++++++++++- sound/soc/codecs/cs35l41.c | 139 +++++++++++++++++-- sound/soc/codecs/cs35l41.h | 1 + 6 files changed, 249 insertions(+), 9 deletions(-) -- 2.33.1

3 5

No output for AC3
by GitHub issues - opened 20 Oct '21

20 Oct '21

alsa-project/alsa-plugins issue #36 was opened from mizapf: Sorry if this is not the proper place to ask, I'm gratefully taking any other suggestions. I use a toslink table from my computer to a Logitech z906 system, which can decode AC3 (as proven in Windows which is installed in parallel). If I select "Digital Stereo (IEC958)", I'm getting stereo output. Also, I can play a special AC3 test file (avi) via VLC and get a proper decoding of 5.1 (obviously passed thru). I assume that if I select "Digital Surround 5.1 (IEC958/AC3)", the A52 plugin is used. However, I get nothing but silence from the 6 speakers when I do the test, and when I try any other application. I am using pulseaudio (probably as in most installations today); I don't know whether this is important. I thought that the A52 plugin must be configured in .asoundrc before I learned that this is already done in /etc/alsa/conf.d/60-a52-encoder.conf. Moreover, the manuals for A52 and configuration do not reveal a lot, in particular for troubleshooting. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/36 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/5] ASoC: cleanup / tidyup soc-pcm/core/component
by Kuninori Morimoto 20 Oct '21

20 Oct '21

Hi Mark These are not a big deal, but cleanup / tidyup patch for ALSA SoC. Kuninori Morimoto (5): ASoC: soc-pcm: tidyup soc_pcm_hw_clean() - step1 ASoC: soc-pcm: tidyup soc_pcm_hw_clean() - step2 ASoC: soc-component: add snd_soc_component_is_codec() ASoC: soc-core: tidyup empty function ASoC: soc-core: accept zero format at snd_soc_runtime_set_dai_fmt() include/sound/soc-component.h | 5 +++ sound/soc/generic/simple-card-utils.c | 4 +-- sound/soc/meson/meson-codec-glue.c | 3 -- sound/soc/soc-core.c | 51 ++++++++------------------- sound/soc/soc-pcm.c | 13 +++---- 5 files changed, 26 insertions(+), 50 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: dt-bindings: mediatek: rename reset controller headers in DT example
by Tzung-Bi Shih 20 Oct '21

20 Oct '21

Commit f07c776f6d7e ("arm64: dts: mediatek: Move reset controller constants into common location") moves the reset controller headers. However, it forgot to rename the DT example in mt8192-afe-pcm.yaml. Renames the DT example to pass dt_binding_check. Fixes: f07c776f6d7e ("arm64: dts: mediatek: Move reset controller constants into common location") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- The patch bases on next-20211018. In Mark's tree[1], the commit f07c776f6d7e hasn't shown up. In Matthias's tree[2], mt8192-afe-pcm.yaml hasn't applied. [1]: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/ [2]: https://git.kernel.org/pub/scm/linux/kernel/git/matthias.bgg/linux.git/ Also, I am not sure if the commit hash "f07c776f6d7e" would change or not after it applies to mainline. Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml index 5e9fe067f440..7a25bc9b8060 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -74,7 +74,7 @@ examples: #include <dt-bindings/interrupt-controller/arm-gic.h> #include <dt-bindings/interrupt-controller/irq.h> #include <dt-bindings/power/mt8192-power.h> - #include <dt-bindings/reset-controller/mt8192-resets.h> + #include <dt-bindings/reset/mt8192-resets.h> afe: mt8192-afe-pcm { compatible = "mediatek,mt8192-audio"; -- 2.33.0.1079.g6e70778dc9-goog

3 2

[PATCH] sound: soc: tlv320aic32x4: Make aic32x4_remove() return void
by Uwe Kleine-König 20 Oct '21

20 Oct '21

Up to now aic32x4_remove() returns zero unconditionally. Make it return void instead which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c and spi remove callbacks is ignored anyway. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/tlv320aic32x4-i2c.c | 4 +++- sound/soc/codecs/tlv320aic32x4-spi.c | 4 +++- sound/soc/codecs/tlv320aic32x4.c | 4 +--- sound/soc/codecs/tlv320aic32x4.h | 2 +- 4 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c index 04ad38311360..ed70e3d9baf2 100644 --- a/sound/soc/codecs/tlv320aic32x4-i2c.c +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -44,7 +44,9 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, static int aic32x4_i2c_remove(struct i2c_client *i2c) { - return aic32x4_remove(&i2c->dev); + aic32x4_remove(&i2c->dev); + + return 0; } static const struct i2c_device_id aic32x4_i2c_id[] = { diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index e81c72958a82..a8958cd1c692 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -48,7 +48,9 @@ static int aic32x4_spi_probe(struct spi_device *spi) static int aic32x4_spi_remove(struct spi_device *spi) { - return aic32x4_remove(&spi->dev); + aic32x4_remove(&spi->dev); + + return 0; } static const struct spi_device_id aic32x4_spi_id[] = { diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index d39c7d52ecfd..8f42fd7bc053 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1418,13 +1418,11 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL(aic32x4_probe); -int aic32x4_remove(struct device *dev) +void aic32x4_remove(struct device *dev) { struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev); aic32x4_disable_regulators(aic32x4); - - return 0; } EXPORT_SYMBOL(aic32x4_remove); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index e9fd2e55d6c3..4de5bd9e8cc5 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -18,7 +18,7 @@ enum aic32x4_type { extern const struct regmap_config aic32x4_regmap_config; int aic32x4_probe(struct device *dev, struct regmap *regmap); -int aic32x4_remove(struct device *dev); +void aic32x4_remove(struct device *dev); int aic32x4_register_clocks(struct device *dev, const char *mclk_name); /* tlv320aic32x4 register space (in decimal to match datasheet) */ -- 2.30.2

2 1

[PATCH] ASoC: rockchip: i2s-tdm: Fix refcount test
by Nicolas Frattaroli 20 Oct '21

20 Oct '21

During development of V5 of the i2s-tdm patch series, I replaced the atomic refcount with a regular integer, as it was only ever accessed within a spinlock. Foolishly, I got the semantics of atomic_dec_and_test wrong, which resulted in a test for 0 actually becoming a test for >0. The result was that setting the audio frequency broke; switching from 44100 Hz audio playback to 96000 Hz audio playback would garble the sound most unpleasantly. Fix this by checking for --refcount == 0, which is what it should have been all along. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 396277eaa417..5d3abbada72a 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -408,7 +408,7 @@ static void rockchip_snd_txrxctrl(struct snd_pcm_substream *substream, else rockchip_disable_rde(i2s_tdm->regmap); - if (--i2s_tdm->refcount) { + if (--i2s_tdm->refcount == 0) { rockchip_snd_xfer_clear(i2s_tdm, I2S_CLR_TXC | I2S_CLR_RXC); } -- 2.33.1

2 1

[PATCH] ASoC: meson: implement driver_name for snd_soc_card in meson-card-utils
by Christian Hewitt 20 Oct '21

20 Oct '21

Implement driver_name to provide an alternative to card_name for userspace configuration of Amlogic audio cards. Suggested-by: Matthias Reichl <hias(a)horus.com> Signed-off-by: Christian Hewitt <christianshewitt(a)gmail.com> --- The meson AIU/AXG drivers currently require alsa confs based on the board model name which is unique to each board. This is very flexible for audio use-cases, but not very maintainable for distros with a single use-case that need to support a large number of different Amlogic boards. Using a driver name to match the card conf instead of card/model name reduces the number of confs needed in the LibreELEC (Kodi) distro from 41 to 2. sound/soc/meson/meson-card-utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 415cc0046e4b..29b0174f4b5c 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -302,6 +302,7 @@ int meson_card_probe(struct platform_device *pdev) priv->card.owner = THIS_MODULE; priv->card.dev = dev; + priv->card.driver_name = dev->driver->name; priv->match_data = data; ret = snd_soc_of_parse_card_name(&priv->card, "model"); -- 2.17.1

3 2

[v5 2/2] ASoC: max98520: add max98520 audio amplifier driver
by George Song 20 Oct '21

20 Oct '21

add max98520 audio amplifier driver Signed-off-by: George Song <george.song(a)maximintegrated.com> --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98520.c | 769 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98520.h | 159 ++++++++ 4 files changed, 942 insertions(+) create mode 100644 sound/soc/codecs/max98520.c create mode 100644 sound/soc/codecs/max98520.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..7ad1e3850279 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -115,6 +115,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MAX98357A imply SND_SOC_MAX98371 imply SND_SOC_MAX98504 + imply SND_SOC_MAX98520 imply SND_SOC_MAX9867 imply SND_SOC_MAX98925 imply SND_SOC_MAX98926 @@ -922,6 +923,17 @@ config SND_SOC_MAX98927 tristate "Maxim Integrated MAX98927 Speaker Amplifier" depends on I2C +config SND_SOC_MAX98520 + tristate "Maxim Integrated MAX98520 Speaker Amplifier" + depends on I2C + help + Enable support for Maxim Integrated MAX98520 audio + amplifier, which implements a tripler charge pump + based boost converter and supports sample rates of + 8KHz to 192KHz. + + To compile this driver as a module, choose M here. + config SND_SOC_MAX98373 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..daf63e31fdd0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -123,6 +123,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max98927-objs := max98927.o +snd-soc-max98520-objs := max98520.o snd-soc-max98373-objs := max98373.o snd-soc-max98373-i2c-objs := max98373-i2c.o snd-soc-max98373-sdw-objs := max98373-sdw.o @@ -450,6 +451,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o +obj-$(CONFIG_SND_SOC_MAX98520) += snd-soc-max98520.o obj-$(CONFIG_SND_SOC_MAX98373) += snd-soc-max98373.o obj-$(CONFIG_SND_SOC_MAX98373_I2C) += snd-soc-max98373-i2c.o obj-$(CONFIG_SND_SOC_MAX98373_SDW) += snd-soc-max98373-sdw.o diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c new file mode 100644 index 000000000000..bb8649cd421c --- /dev/null +++ b/sound/soc/codecs/max98520.c @@ -0,0 +1,769 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2021, Maxim Integrated + +#include <linux/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98520.h" + +static struct reg_default max98520_reg[] = { + {MAX98520_R2000_SW_RESET, 0x00}, + {MAX98520_R2001_STATUS_1, 0x00}, + {MAX98520_R2002_STATUS_2, 0x00}, + {MAX98520_R2020_THERM_WARN_THRESH, 0x46}, + {MAX98520_R2021_THERM_SHDN_THRESH, 0x64}, + {MAX98520_R2022_THERM_HYSTERESIS, 0x02}, + {MAX98520_R2023_THERM_FOLDBACK_SET, 0x31}, + {MAX98520_R2027_THERM_FOLDBACK_EN, 0x01}, + {MAX98520_R2030_CLK_MON_CTRL, 0x00}, + {MAX98520_R2037_ERR_MON_CTRL, 0x01}, + {MAX98520_R2040_PCM_MODE_CFG, 0xC0}, + {MAX98520_R2041_PCM_CLK_SETUP, 0x04}, + {MAX98520_R2042_PCM_SR_SETUP, 0x08}, + {MAX98520_R2043_PCM_RX_SRC1, 0x00}, + {MAX98520_R2044_PCM_RX_SRC2, 0x00}, + {MAX98520_R204F_PCM_RX_EN, 0x00}, + {MAX98520_R2090_AMP_VOL_CTRL, 0x00}, + {MAX98520_R2091_AMP_PATH_GAIN, 0x03}, + {MAX98520_R2092_AMP_DSP_CFG, 0x02}, + {MAX98520_R2094_SSM_CFG, 0x01}, + {MAX98520_R2095_AMP_CFG, 0xF0}, + {MAX98520_R209F_AMP_EN, 0x00}, + {MAX98520_R20B0_ADC_SR, 0x00}, + {MAX98520_R20B1_ADC_RESOLUTION, 0x00}, + {MAX98520_R20B2_ADC_PVDD0_CFG, 0x02}, + {MAX98520_R20B3_ADC_THERMAL_CFG, 0x02}, + {MAX98520_R20B4_ADC_READBACK_CTRL, 0x00}, + {MAX98520_R20B5_ADC_READBACK_UPDATE, 0x00}, + {MAX98520_R20B6_ADC_PVDD_READBACK_MSB, 0x00}, + {MAX98520_R20B7_ADC_PVDD_READBACK_LSB, 0x00}, + {MAX98520_R20B8_ADC_TEMP_READBACK_MSB, 0x00}, + {MAX98520_R20B9_ADC_TEMP_READBACK_LSB, 0x00}, + {MAX98520_R20BA_ADC_LOW_PVDD_READBACK_MSB, 0xFF}, + {MAX98520_R20BB_ADC_LOW_READBACK_LSB, 0x01}, + {MAX98520_R20BC_ADC_HIGH_TEMP_READBACK_MSB, 0x00}, + {MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB, 0x00}, + {MAX98520_R20CF_MEAS_ADC_CFG, 0x00}, + {MAX98520_R20D0_DHT_CFG1, 0x00}, + {MAX98520_R20D1_LIMITER_CFG1, 0x08}, + {MAX98520_R20D2_LIMITER_CFG2, 0x00}, + {MAX98520_R20D3_DHT_CFG2, 0x14}, + {MAX98520_R20D4_DHT_CFG3, 0x02}, + {MAX98520_R20D5_DHT_CFG4, 0x04}, + {MAX98520_R20D6_DHT_HYSTERESIS_CFG, 0x07}, + {MAX98520_R20D8_DHT_EN, 0x00}, + {MAX98520_R210E_AUTO_RESTART_BEHAVIOR, 0x00}, + {MAX98520_R210F_GLOBAL_EN, 0x00}, + {MAX98520_R21FF_REVISION_ID, 0x00}, +}; + +static int max98520_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + unsigned int format = 0; + unsigned int invert = 0; + + dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(component->dev, "DAI invert mode unsupported\n"); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format = MAX98520_PCM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + format = MAX98520_PCM_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98520_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98520_PCM_FORMAT_TDM_MODE0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_FORMAT_MASK, + format << MAX98520_PCM_MODE_CFG_FORMAT_SHIFT); + + return 0; +} + +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, 320, +}; + +static int max98520_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} + +static int max98520_set_clock(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98520->ch_size; + int value; + + if (!max98520->tdm_mode) { + /* BCLK configuration */ + value = max98520_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_CLK_SETUP_BSEL_MASK, + value); + } + dev_dbg(component->dev, "%s tdm_mode:%d out\n", __func__, max98520->tdm_mode); + return 0; +} + +static int max98520_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + goto err; + } + + max98520->ch_size = snd_pcm_format_width(params_format(params)); + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(component->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98520_PCM_SR_8000; + break; + case 11025: + sampling_rate = MAX98520_PCM_SR_11025; + break; + case 12000: + sampling_rate = MAX98520_PCM_SR_12000; + break; + case 16000: + sampling_rate = MAX98520_PCM_SR_16000; + break; + case 22050: + sampling_rate = MAX98520_PCM_SR_22050; + break; + case 24000: + sampling_rate = MAX98520_PCM_SR_24000; + break; + case 32000: + sampling_rate = MAX98520_PCM_SR_32000; + break; + case 44100: + sampling_rate = MAX98520_PCM_SR_44100; + break; + case 48000: + sampling_rate = MAX98520_PCM_SR_48000; + break; + case 88200: + sampling_rate = MAX98520_PCM_SR_88200; + break; + case 96000: + sampling_rate = MAX98520_PCM_SR_96000; + break; + case 176400: + sampling_rate = MAX98520_PCM_SR_176400; + break; + case 192000: + sampling_rate = MAX98520_PCM_SR_192000; + break; + default: + dev_err(component->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + + dev_dbg(component->dev, " %s ch_size: %d, sampling rate : %d out\n", __func__, + snd_pcm_format_width(params_format(params)), params_rate(params)); + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98520->regmap, + MAX98520_R2042_PCM_SR_SETUP, + MAX98520_PCM_SR_MASK, + sampling_rate); + + return max98520_set_clock(component, params); +err: + dev_dbg(component->dev, "%s out error", __func__); + return -EINVAL; +} + +static int max98520_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + int bsel; + unsigned int chan_sz = 0; + + if (!tx_mask && !rx_mask && !slots && !slot_width) + max98520->tdm_mode = false; + else + max98520->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98520_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(component->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_update_bits(max98520->regmap, + MAX98520_R2044_PCM_RX_SRC2, + MAX98520_PCM_DMIX_CH0_SRC_MASK, + rx_mask); + regmap_update_bits(max98520->regmap, + MAX98520_R2044_PCM_RX_SRC2, + MAX98520_PCM_DMIX_CH1_SRC_MASK, + rx_mask << MAX98520_PCM_DMIX_CH1_SHIFT); + + return 0; +} + +#define MAX98520_RATES SNDRV_PCM_RATE_8000_192000 + +#define MAX98520_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops max98520_dai_ops = { + .set_fmt = max98520_dai_set_fmt, + .hw_params = max98520_dai_hw_params, + .set_tdm_slot = max98520_dai_tdm_slot, +}; + +static int max98520_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_dbg(component->dev, " AMP ON\n"); + + regmap_write(max98520->regmap, MAX98520_R209F_AMP_EN, 1); + regmap_write(max98520->regmap, MAX98520_R210F_GLOBAL_EN, 1); + usleep_range(30000, 31000); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, " AMP OFF\n"); + + regmap_write(max98520->regmap, MAX98520_R210F_GLOBAL_EN, 0); + regmap_write(max98520->regmap, MAX98520_R209F_AMP_EN, 0); + usleep_range(30000, 31000); + break; + default: + return 0; + } + return 0; +} + +static const char * const max98520_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98520_R2043_PCM_RX_SRC1, + 0, 3, max98520_switch_text); + +static const struct snd_kcontrol_new max98520_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_kcontrol_new max98520_left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("PCM_INPUT_CH0", MAX98520_R2044_PCM_RX_SRC2, 0, 0x0, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH1", MAX98520_R2044_PCM_RX_SRC2, 0, 0x1, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH2", MAX98520_R2044_PCM_RX_SRC2, 0, 0x2, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH3", MAX98520_R2044_PCM_RX_SRC2, 0, 0x3, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH4", MAX98520_R2044_PCM_RX_SRC2, 0, 0x4, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH5", MAX98520_R2044_PCM_RX_SRC2, 0, 0x5, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH6", MAX98520_R2044_PCM_RX_SRC2, 0, 0x6, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH7", MAX98520_R2044_PCM_RX_SRC2, 0, 0x7, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH8", MAX98520_R2044_PCM_RX_SRC2, 0, 0x8, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH9", MAX98520_R2044_PCM_RX_SRC2, 0, 0x9, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH10", MAX98520_R2044_PCM_RX_SRC2, 0, 0xa, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH11", MAX98520_R2044_PCM_RX_SRC2, 0, 0xb, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH12", MAX98520_R2044_PCM_RX_SRC2, 0, 0xc, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH13", MAX98520_R2044_PCM_RX_SRC2, 0, 0xd, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH14", MAX98520_R2044_PCM_RX_SRC2, 0, 0xe, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH15", MAX98520_R2044_PCM_RX_SRC2, 0, 0xf, 0), +}; + +static const struct snd_kcontrol_new max98520_right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("PCM_INPUT_CH0", MAX98520_R2044_PCM_RX_SRC2, 4, 0x0, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH1", MAX98520_R2044_PCM_RX_SRC2, 4, 0x1, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH2", MAX98520_R2044_PCM_RX_SRC2, 4, 0x2, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH3", MAX98520_R2044_PCM_RX_SRC2, 4, 0x3, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH4", MAX98520_R2044_PCM_RX_SRC2, 4, 0x4, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH5", MAX98520_R2044_PCM_RX_SRC2, 4, 0x5, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH6", MAX98520_R2044_PCM_RX_SRC2, 4, 0x6, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH7", MAX98520_R2044_PCM_RX_SRC2, 4, 0x7, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH8", MAX98520_R2044_PCM_RX_SRC2, 4, 0x8, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH9", MAX98520_R2044_PCM_RX_SRC2, 4, 0x9, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH10", MAX98520_R2044_PCM_RX_SRC2, 4, 0xa, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH11", MAX98520_R2044_PCM_RX_SRC2, 4, 0xb, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH12", MAX98520_R2044_PCM_RX_SRC2, 4, 0xc, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH13", MAX98520_R2044_PCM_RX_SRC2, 4, 0xd, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH14", MAX98520_R2044_PCM_RX_SRC2, 4, 0xe, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH15", MAX98520_R2044_PCM_RX_SRC2, 4, 0xf, 0), +}; + +static const struct snd_soc_dapm_widget max98520_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", + SND_SOC_NOPM, 0, 0, max98520_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98520_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input Selection", SND_SOC_NOPM, 0, 0, + &max98520_left_input_mixer_controls[0], + ARRAY_SIZE(max98520_left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input Selection", SND_SOC_NOPM, 0, 0, + &max98520_right_input_mixer_controls[0], + ARRAY_SIZE(max98520_right_input_mixer_controls)), +}; + +static const DECLARE_TLV_DB_SCALE(max98520_digital_tlv, -6300, 50, 1); +static const DECLARE_TLV_DB_SCALE(max98520_spk_tlv, -600, 300, 0); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_lim_thresh_tlv, + 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_hysteresis_tlv, + 0, 3, TLV_DB_SCALE_ITEM(100, 100, 0), + 4, 7, TLV_DB_SCALE_ITEM(600, 200, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_rotation_point_tlv, + 0, 1, TLV_DB_SCALE_ITEM(-1500, 300, 0), + 2, 4, TLV_DB_SCALE_ITEM(-1000, 200, 0), + 5, 10, TLV_DB_SCALE_ITEM(-500, 100, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_supply_hr_tlv, + 0, 16, TLV_DB_SCALE_ITEM(-2000, 250, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_max_atten_tlv, + 1, 20, TLV_DB_SCALE_ITEM(-2000, 100, 0), +); + +static const char * const max98520_dht_attack_rate_text[] = { + "20us", "40us", "80us", "160us", "320us", "640us", + "1.28ms", "2.56ms", "5.12ms", "10.24ms", "20.48ms", "40.96ms", + "81.92ms", "163.84ms" +}; + +static SOC_ENUM_SINGLE_DECL(max98520_dht_attack_rate_enum, + MAX98520_R20D4_DHT_CFG3, 0, + max98520_dht_attack_rate_text); + +static const char * const max98520_dht_release_rate_text[] = { + "2ms", "4ms", "8ms", "16ms", "32ms", "64ms", "128ms", "256ms", "512ms", + "1.024s", "2.048s", "4.096s", "8.192s", "16.384s" +}; + +static SOC_ENUM_SINGLE_DECL(max98520_dht_release_rate_enum, + MAX98520_R20D5_DHT_CFG4, 0, + max98520_dht_release_rate_text); + +static bool max98520_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98520_R2000_SW_RESET: + case MAX98520_R2027_THERM_FOLDBACK_EN: + case MAX98520_R2030_CLK_MON_CTRL: + case MAX98520_R2037_ERR_MON_CTRL: + case MAX98520_R204F_PCM_RX_EN: + case MAX98520_R209F_AMP_EN: + case MAX98520_R20CF_MEAS_ADC_CFG: + case MAX98520_R20D8_DHT_EN: + case MAX98520_R21FF_REVISION_ID: + case MAX98520_R2001_STATUS_1... MAX98520_R2002_STATUS_2: + case MAX98520_R2020_THERM_WARN_THRESH... MAX98520_R2023_THERM_FOLDBACK_SET: + case MAX98520_R2040_PCM_MODE_CFG... MAX98520_R2044_PCM_RX_SRC2: + case MAX98520_R2090_AMP_VOL_CTRL... MAX98520_R2092_AMP_DSP_CFG: + case MAX98520_R2094_SSM_CFG... MAX98520_R2095_AMP_CFG: + case MAX98520_R20B0_ADC_SR... MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB: + case MAX98520_R20D0_DHT_CFG1... MAX98520_R20D6_DHT_HYSTERESIS_CFG: + case MAX98520_R210E_AUTO_RESTART_BEHAVIOR... MAX98520_R210F_GLOBAL_EN: + case MAX98520_R2161_BOOST_TM1... MAX98520_R2163_BOOST_TM3: + return true; + default: + return false; + } +}; + +static bool max98520_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98520_R210F_GLOBAL_EN: + case MAX98520_R21FF_REVISION_ID: + case MAX98520_R2000_SW_RESET: + case MAX98520_R2001_STATUS_1 ... MAX98520_R2002_STATUS_2: + case MAX98520_R20B4_ADC_READBACK_CTRL + ... MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB: + return true; + default: + return false; + } +} + +static const struct snd_kcontrol_new max98520_snd_controls[] = { +/* Volume */ +SOC_SINGLE_TLV("Digital Volume", MAX98520_R2090_AMP_VOL_CTRL, + 0, 0x7F, 1, max98520_digital_tlv), +SOC_SINGLE_TLV("Speaker Volume", MAX98520_R2091_AMP_PATH_GAIN, + 0, 0x5, 0, max98520_spk_tlv), +/* Volume Ramp Up/Down Enable*/ +SOC_SINGLE("Ramp Up Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_VOL_RMPUP_SHIFT, 1, 0), +SOC_SINGLE("Ramp Down Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_VOL_RMPDN_SHIFT, 1, 0), +/* Clock Monitor Enable */ +SOC_SINGLE("CLK Monitor Switch", MAX98520_R2037_ERR_MON_CTRL, + MAX98520_CTRL_CMON_EN_SHIFT, 1, 0), +/* Clock Monitor Config */ +SOC_SINGLE("CLKMON Autorestart Switch", MAX98520_R2030_CLK_MON_CTRL, + MAX98520_CMON_AUTORESTART_SHIFT, 1, 0), +/* Dither Enable */ +SOC_SINGLE("Dither Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_DITH_EN_SHIFT, 1, 0), +/* DC Blocker Enable */ +SOC_SINGLE("DC Blocker Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_DCBLK_EN_SHIFT, 1, 0), +/* Speaker Safe Mode Enable */ +SOC_SINGLE("Speaker Safemode Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_SAFE_EN_SHIFT, 1, 0), +/* AMP SSM Enable */ +SOC_SINGLE("CP Bypass Switch", MAX98520_R2094_SSM_CFG, + MAX98520_SSM_RCVR_MODE_SHIFT, 1, 0), +/* Dynamic Headroom Tracking */ +SOC_SINGLE("DHT Switch", MAX98520_R20D8_DHT_EN, 0, 1, 0), +SOC_SINGLE("DHT Limiter Mode", MAX98520_R20D2_LIMITER_CFG2, + MAX98520_DHT_LIMITER_MODE_SHIFT, 1, 0), +SOC_SINGLE("DHT Hysteresis Switch", MAX98520_R20D6_DHT_HYSTERESIS_CFG, + MAX98520_DHT_HYSTERESIS_SWITCH_SHIFT, 1, 0), +SOC_SINGLE_TLV("DHT Rot Pnt", MAX98520_R20D0_DHT_CFG1, + MAX98520_DHT_VROT_PNT_SHIFT, 10, 1, max98520_dht_rotation_point_tlv), +SOC_SINGLE_TLV("DHT Supply Headroom", MAX98520_R20D1_LIMITER_CFG1, + MAX98520_DHT_SUPPLY_HR_SHIFT, 16, 0, max98520_dht_supply_hr_tlv), +SOC_SINGLE_TLV("DHT Limiter Threshold", MAX98520_R20D2_LIMITER_CFG2, + MAX98520_DHT_LIMITER_THRESHOLD_SHIFT, 0xF, 1, max98520_dht_lim_thresh_tlv), +SOC_SINGLE_TLV("DHT Max Attenuation", MAX98520_R20D3_DHT_CFG2, + MAX98520_DHT_MAX_ATTEN_SHIFT, 20, 1, max98520_dht_max_atten_tlv), +SOC_SINGLE_TLV("DHT Hysteresis", MAX98520_R20D6_DHT_HYSTERESIS_CFG, + MAX98520_DHT_HYSTERESIS_SHIFT, 0x7, 0, max98520_dht_hysteresis_tlv), +SOC_ENUM("DHT Attack Rate", max98520_dht_attack_rate_enum), +SOC_ENUM("DHT Release Rate", max98520_dht_release_rate_enum), +/* ADC configuration */ +SOC_SINGLE("ADC PVDD CH Switch", MAX98520_R20CF_MEAS_ADC_CFG, 0, 1, 0), +SOC_SINGLE("ADC PVDD FLT Switch", MAX98520_R20B2_ADC_PVDD0_CFG, MAX98520_FLT_EN_SHIFT, 1, 0), +SOC_SINGLE("ADC TEMP FLT Switch", MAX98520_R20B3_ADC_THERMAL_CFG, MAX98520_FLT_EN_SHIFT, 1, 0), +SOC_SINGLE("ADC PVDD MSB", MAX98520_R20B6_ADC_PVDD_READBACK_MSB, 0, 0xFF, 0), +SOC_SINGLE("ADC PVDD LSB", MAX98520_R20B7_ADC_PVDD_READBACK_LSB, 0, 0x01, 0), +SOC_SINGLE("ADC TEMP MSB", MAX98520_R20B8_ADC_TEMP_READBACK_MSB, 0, 0xFF, 0), +SOC_SINGLE("ADC TEMP LSB", MAX98520_R20B9_ADC_TEMP_READBACK_LSB, 0, 0x01, 0), +}; + +static const struct snd_soc_dapm_route max98520_audio_map[] = { + /* Plabyack */ + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98520_dai[] = { + { + .name = "max98520-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98520_RATES, + .formats = MAX98520_FORMATS, + }, + .ops = &max98520_dai_ops, + } + +}; + +static int max98520_probe(struct snd_soc_component *component) +{ + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + + /* Software Reset */ + regmap_write(max98520->regmap, MAX98520_R2000_SW_RESET, 1); + + /* L/R mono mix configuration : "DAI Sel" for 0x2043 */ + regmap_write(max98520->regmap, MAX98520_R2043_PCM_RX_SRC1, 0x2); + + /* PCM input channles configuration : "Left Input Selection" for 0x2044 */ + /* PCM input channles configuration : "Right Input Selection" for 0x2044 */ + regmap_write(max98520->regmap, MAX98520_R2044_PCM_RX_SRC2, 0x10); + + /* Enable DC blocker */ + regmap_update_bits(max98520->regmap, MAX98520_R2092_AMP_DSP_CFG, 1, 1); + /* Enable Clock Monitor Auto-restart */ + regmap_write(max98520->regmap, MAX98520_R2030_CLK_MON_CTRL, 0x1); + + /* set Rx Enable */ + regmap_update_bits(max98520->regmap, + MAX98520_R204F_PCM_RX_EN, + MAX98520_PCM_RX_EN_MASK, + 1); + + return 0; +} + +static int __maybe_unused max98520_suspend(struct device *dev) +{ + struct max98520_priv *max98520 = dev_get_drvdata(dev); + + regcache_cache_only(max98520->regmap, true); + regcache_mark_dirty(max98520->regmap); + return 0; +} + +static int __maybe_unused max98520_resume(struct device *dev) +{ + struct max98520_priv *max98520 = dev_get_drvdata(dev); + + regcache_cache_only(max98520->regmap, false); + regmap_write(max98520->regmap, MAX98520_R2000_SW_RESET, 1); + regcache_sync(max98520->regmap); + return 0; +} + +static const struct dev_pm_ops max98520_pm = { + SET_SYSTEM_SLEEP_PM_OPS(max98520_suspend, max98520_resume) +}; + +static const struct snd_soc_component_driver soc_codec_dev_max98520 = { + .probe = max98520_probe, + .controls = max98520_snd_controls, + .num_controls = ARRAY_SIZE(max98520_snd_controls), + .dapm_widgets = max98520_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98520_dapm_widgets), + .dapm_routes = max98520_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98520_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config max98520_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98520_R21FF_REVISION_ID, + .reg_defaults = max98520_reg, + .num_reg_defaults = ARRAY_SIZE(max98520_reg), + .readable_reg = max98520_readable_register, + .volatile_reg = max98520_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static void max98520_power_on(struct max98520_priv *max98520, bool poweron) +{ + if (max98520->reset_gpio) + gpiod_set_value_cansleep(max98520->reset_gpio, !poweron); +} + +static int max98520_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + int ret; + int reg = 0; + struct max98520_priv *max98520; + struct i2c_adapter *adapter = to_i2c_adapter(i2c->dev.parent); + + ret = i2c_check_functionality(adapter, I2C_FUNC_SMBUS_BYTE | I2C_FUNC_SMBUS_BYTE_DATA); + if (!ret) { + dev_err(&i2c->dev, "I2C check functionality failed\n"); + return -ENXIO; + } + + max98520 = devm_kzalloc(&i2c->dev, sizeof(*max98520), GFP_KERNEL); + + if (!max98520) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98520); + + /* regmap initialization */ + max98520->regmap = devm_regmap_init_i2c(i2c, &max98520_regmap); + if (IS_ERR(max98520->regmap)) { + ret = PTR_ERR(max98520->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + + /* Power on device */ + max98520->reset_gpio = devm_gpiod_get_optional(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (max98520->reset_gpio) { + if (IS_ERR(max98520->reset_gpio)) { + ret = PTR_ERR(max98520->reset_gpio); + dev_err(&i2c->dev, "Unable to request GPIO pin: %d.\n", ret); + return ret; + } + + max98520_power_on(max98520, 1); + } + + /* Check Revision ID */ + ret = regmap_read(max98520->regmap, MAX98520_R21FF_REVISION_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, + "Failed to read: 0x%02X\n", MAX98520_R21FF_REVISION_ID); + return ret; + } + dev_info(&i2c->dev, "MAX98520 revisionID: 0x%02X\n", reg); + + /* codec registration */ + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_max98520, + max98520_dai, ARRAY_SIZE(max98520_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static const struct i2c_device_id max98520_i2c_id[] = { + { "max98520", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98520_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98520_of_match[] = { + { .compatible = "maxim,max98520", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98520_of_match); +#endif + +static struct i2c_driver max98520_i2c_driver = { + .driver = { + .name = "max98520", + .of_match_table = of_match_ptr(max98520_of_match), + .pm = &max98520_pm, + }, + .probe = max98520_i2c_probe, + .id_table = max98520_i2c_id, +}; + +module_i2c_driver(max98520_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98520 driver"); +MODULE_AUTHOR("George Song <george.song(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/max98520.h b/sound/soc/codecs/max98520.h new file mode 100644 index 000000000000..89a95c25afcf --- /dev/null +++ b/sound/soc/codecs/max98520.h @@ -0,0 +1,159 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2021, Maxim Integrated. + */ + +#ifndef _MAX98520_H +#define _MAX98520_H + +#define MAX98520_R2000_SW_RESET 0x2000 +#define MAX98520_R2001_STATUS_1 0x2001 +#define MAX98520_R2002_STATUS_2 0x2002 +#define MAX98520_R2020_THERM_WARN_THRESH 0x2020 +#define MAX98520_R2021_THERM_SHDN_THRESH 0x2021 +#define MAX98520_R2022_THERM_HYSTERESIS 0x2022 +#define MAX98520_R2023_THERM_FOLDBACK_SET 0x2023 +#define MAX98520_R2027_THERM_FOLDBACK_EN 0x2027 +#define MAX98520_R2030_CLK_MON_CTRL 0x2030 +#define MAX98520_R2037_ERR_MON_CTRL 0x2037 +#define MAX98520_R2040_PCM_MODE_CFG 0x2040 +#define MAX98520_R2041_PCM_CLK_SETUP 0x2041 +#define MAX98520_R2042_PCM_SR_SETUP 0x2042 +#define MAX98520_R2043_PCM_RX_SRC1 0x2043 +#define MAX98520_R2044_PCM_RX_SRC2 0x2044 +#define MAX98520_R204F_PCM_RX_EN 0x204F +#define MAX98520_R2090_AMP_VOL_CTRL 0x2090 +#define MAX98520_R2091_AMP_PATH_GAIN 0x2091 +#define MAX98520_R2092_AMP_DSP_CFG 0x2092 +#define MAX98520_R2094_SSM_CFG 0x2094 +#define MAX98520_R2095_AMP_CFG 0x2095 +#define MAX98520_R209F_AMP_EN 0x209F +#define MAX98520_R20B0_ADC_SR 0x20B0 +#define MAX98520_R20B1_ADC_RESOLUTION 0x20B1 +#define MAX98520_R20B2_ADC_PVDD0_CFG 0x20B2 +#define MAX98520_R20B3_ADC_THERMAL_CFG 0x20B3 +#define MAX98520_R20B4_ADC_READBACK_CTRL 0x20B4 +#define MAX98520_R20B5_ADC_READBACK_UPDATE 0x20B5 +#define MAX98520_R20B6_ADC_PVDD_READBACK_MSB 0x20B6 +#define MAX98520_R20B7_ADC_PVDD_READBACK_LSB 0x20B7 +#define MAX98520_R20B8_ADC_TEMP_READBACK_MSB 0x20B8 +#define MAX98520_R20B9_ADC_TEMP_READBACK_LSB 0x20B9 +#define MAX98520_R20BA_ADC_LOW_PVDD_READBACK_MSB 0x20BA +#define MAX98520_R20BB_ADC_LOW_READBACK_LSB 0x20BB +#define MAX98520_R20BC_ADC_HIGH_TEMP_READBACK_MSB 0x20BC +#define MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB 0x20BD +#define MAX98520_R20CF_MEAS_ADC_CFG 0x20CF +#define MAX98520_R20D0_DHT_CFG1 0x20D0 +#define MAX98520_R20D1_LIMITER_CFG1 0x20D1 +#define MAX98520_R20D2_LIMITER_CFG2 0x20D2 +#define MAX98520_R20D3_DHT_CFG2 0x20D3 +#define MAX98520_R20D4_DHT_CFG3 0x20D4 +#define MAX98520_R20D5_DHT_CFG4 0x20D5 +#define MAX98520_R20D6_DHT_HYSTERESIS_CFG 0x20D6 +#define MAX98520_R20D8_DHT_EN 0x20D8 +#define MAX98520_R210E_AUTO_RESTART_BEHAVIOR 0x210E +#define MAX98520_R210F_GLOBAL_EN 0x210F +#define MAX98520_R2161_BOOST_TM1 0x2161 +#define MAX98520_R2162_BOOST_TM2 0x2162 +#define MAX98520_R2163_BOOST_TM3 0x2163 +#define MAX98520_R21FF_REVISION_ID 0x21FF + +/* MAX98520_R2030_CLK_MON_CTRL */ +#define MAX98520_CMON_AUTORESTART_SHIFT (0) + +/* MAX98520_R2037_ERR_MON_CTRL */ +#define MAX98520_CTRL_CMON_EN_SHIFT (0) + +/* MAX98520_R2040_PCM_MODE_CFG */ +#define MAX98520_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) +#define MAX98520_PCM_MODE_CFG_FORMAT_SHIFT (3) +#define MAX98520_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 2) +#define MAX98520_PCM_FORMAT_I2S (0x0 << 3) +#define MAX98520_PCM_FORMAT_LJ (0x1 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE0 (0x3 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE1 (0x4 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE2 (0x5 << 3) +#define MAX98520_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98520_R2041_PCM_CLK_SETUP */ +#define MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 4) +#define MAX98520_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98520_R2042_PCM_SR_SETUP */ +#define MAX98520_PCM_SR_SHIFT (0) +#define MAX98520_IVADC_SR_SHIFT (4) +#define MAX98520_PCM_SR_MASK (0xF << MAX98520_PCM_SR_SHIFT) +#define MAX98520_IVADC_SR_MASK (0xF << MAX98520_IVADC_SR_SHIFT) +#define MAX98520_PCM_SR_8000 (0x0) +#define MAX98520_PCM_SR_11025 (0x1) +#define MAX98520_PCM_SR_12000 (0x2) +#define MAX98520_PCM_SR_16000 (0x3) +#define MAX98520_PCM_SR_22050 (0x4) +#define MAX98520_PCM_SR_24000 (0x5) +#define MAX98520_PCM_SR_32000 (0x6) +#define MAX98520_PCM_SR_44100 (0x7) +#define MAX98520_PCM_SR_48000 (0x8) +#define MAX98520_PCM_SR_88200 (0x9) +#define MAX98520_PCM_SR_96000 (0xA) +#define MAX98520_PCM_SR_176400 (0xB) +#define MAX98520_PCM_SR_192000 (0xC) + +/* MAX98520_R2044_PCM_RX_SRC2 */ +#define MAX98520_PCM_DMIX_CH1_SHIFT (0xF << 0) +#define MAX98520_PCM_DMIX_CH0_SRC_MASK (0xF << 0) +#define MAX98520_PCM_DMIX_CH1_SRC_MASK (0xF << MAX98520_PCM_DMIX_CH1_SHIFT) + +/* MAX98520_R204F_PCM_RX_EN */ +#define MAX98520_PCM_RX_EN_MASK (0x1 << 0) +#define MAX98520_PCM_RX_BYP_EN_MASK (0x1 << 1) + +/* MAX98520_R2092_AMP_DSP_CFG */ +#define MAX98520_DSP_SPK_DCBLK_EN_SHIFT (0) +#define MAX98520_DSP_SPK_DITH_EN_SHIFT (1) +#define MAX98520_DSP_SPK_INVERT_SHIFT (2) +#define MAX98520_DSP_SPK_VOL_RMPUP_SHIFT (3) +#define MAX98520_DSP_SPK_VOL_RMPDN_SHIFT (4) +#define MAX98520_DSP_SPK_SAFE_EN_SHIFT (5) + +#define MAX98520_SPK_SAFE_EN_MASK (0x1 << MAX98520_DSP_SPK_SAFE_EN_SHIFT) + +/* MAX98520_R2094_SSM_CFG */ +#define MAX98520_SSM_EN_SHIFT (0) +#define MAX98520_SSM_MOD_SHIFT (1) +#define MAX98520_SSM_RCVR_MODE_SHIFT (3) + +/* MAX98520_R2095_AMP_CFG */ +#define MAX98520_CFG_DYN_MODE_SHIFT (4) +#define MAX98520_CFG_SPK_MODE_SHIFT (3) + +/* MAX98520_R20D0_DHT_CFG1 */ +#define MAX98520_DHT_VROT_PNT_SHIFT (0) + +/* MAX98520_R20D1_LIMITER_CFG1 */ +#define MAX98520_DHT_SUPPLY_HR_SHIFT (0) + +/* MAX98520_R20D2_DHT_CFG2 */ +#define MAX98520_DHT_LIMITER_MODE_SHIFT (0) +#define MAX98520_DHT_LIMITER_THRESHOLD_SHIFT (1) + +/* MAX98520_R20D3_DHT_CFG2 */ +#define MAX98520_DHT_MAX_ATTEN_SHIFT (0) + +/* MAX98520_R20D6_DHT_HYSTERESIS_CFG */ +#define MAX98520_DHT_HYSTERESIS_SWITCH_SHIFT (0) +#define MAX98520_DHT_HYSTERESIS_SHIFT (1) + +/* MAX98520_R20B2_ADC_PVDD0_CFG, MAX98520_R20B3_ADC_THERMAL_CFG */ +#define MAX98520_FLT_EN_SHIFT (4) + +struct max98520_priv { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + unsigned int ch_size; + bool tdm_mode; +}; +#endif + -- 2.25.1

1 0

[PATCH 00/11] ALSA: firewire-motu: add ioctl commands to retrieve information in messages delivered by isoc packet
by Takashi Sakamoto 20 Oct '21

20 Oct '21

Hi, The purpose of this patchset is to add message parser to ALSA firewire-motu driver so that userspace applications can read information in message delivered by isochronous packet as well as PCM frames. The message includes information about hardware meter and user action over hardware component such as knob, and MIDI message bytes. Models in MOTU FireWire series can be categorized to 4 groups in regard of message mechanism: Group 1. 828 and 896 * quadlet message to registered destination address Group 2. 828mk2, 896hd, Traveler, 8 pre, Ultralite, 4 pre, and Audio Express * quadlet message to registered destination address * message delivered by isochronous packet Group 3. 828mk3, 896mk3, Ultralite mk3, Traveler mk3, and Track 16 * quadlet message to registered destination address * message delivered by isochronous packet * block message to registered destination address, including command Group 4. V3HD/V4HD * quadlet message to registered destination address * block message to registered destination address The patchset is for message delivered by isochronous packet in group 2 and 3. In Group 2, the message includes information of hardware meter, information of user action over hardware component. The model in Group 2 is called as 'register DSP' in the patchset since parameters of DSP can be configured by asynchronous transaction for register access. In Group 3, the message includes information of hardware meter only. The model in Group 3 is called as 'command DSP' since software and device communicate with commands transferred by asynchronous transaction in regard of DSP parameters. Two types of message parser is going to be added so that the driver caches images for the information. The cache is available for userspace application by ioctl commands newly introduced. I note that no control element is added for the hardware meters and user actions. It's expected for userspace application to retrieve and parse the information of image then operate for user-defined control element set. Takashi Sakamoto (11): ALSA: firewire-motu: add message parser to gather meter information in register DSP model ALSA: firewire-motu: add message parser for meter information in command DSP model ALSA: firewire-motu: add ioctl command to read cached hardware meter ALSA: firewire-motu: parse messages for mixer source parameters in register-DSP model ALSA: firewire-motu: parse messages for mixer output parameters in register DSP model ALSA: firewire-motu: parse messages for output parameters in register DSP model ALSA: firewire-motu: parse messages for line input parameters in register DSP model ALSA: firewire-motu: parse messages for input parameters in register DSP model ALSA: firewire-motu: add ioctl command to read cached DSP parameters ALSA: firewire-motu: queue event for parameter change in register DSP model ALSA: firewire-motu: notify event for parameter change in register DSP model include/uapi/sound/firewire.h | 141 ++++++ sound/firewire/motu/Makefile | 3 +- sound/firewire/motu/amdtp-motu.c | 9 + .../motu/motu-command-dsp-message-parser.c | 178 ++++++++ sound/firewire/motu/motu-hwdep.c | 113 ++++- sound/firewire/motu/motu-protocol-v2.c | 14 +- sound/firewire/motu/motu-protocol-v3.c | 14 +- .../motu/motu-register-dsp-message-parser.c | 416 ++++++++++++++++++ sound/firewire/motu/motu-stream.c | 10 + sound/firewire/motu/motu.c | 10 + sound/firewire/motu/motu.h | 23 + 11 files changed, 911 insertions(+), 20 deletions(-) create mode 100644 sound/firewire/motu/motu-command-dsp-message-parser.c create mode 100644 sound/firewire/motu/motu-register-dsp-message-parser.c -- 2.30.2

2 19

[v4] ASoC: max98520: add max98520 audio amplifier driver
by George Song 20 Oct '21

20 Oct '21

add max98520 audio amplifier driver Signed-off-by: George Song <george.song(a)maximintegrated.com> --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98520.c | 798 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98520.h | 159 +++++++ 4 files changed, 971 insertions(+) create mode 100644 sound/soc/codecs/max98520.c create mode 100644 sound/soc/codecs/max98520.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..e8817029c671 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -115,6 +115,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MAX98357A imply SND_SOC_MAX98371 imply SND_SOC_MAX98504 + imply SND_SOC_MAX98520 imply SND_SOC_MAX9867 imply SND_SOC_MAX98925 imply SND_SOC_MAX98926 @@ -922,6 +923,17 @@ config SND_SOC_MAX98927 tristate "Maxim Integrated MAX98927 Speaker Amplifier" depends on I2C +config SND_SOC_MAX98520 + tristate "Maxim Integrated MAX98520 Speaker Amplifier" + depends on I2C + help + Enable support for Maxim Integrated MAX98520 audio + amplifier, which implements a tripler charge pump + based boost converter and support sample rates of + 8KHz to 192KHz. + + To compile this driver as a module, choose M here. + config SND_SOC_MAX98373 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..daf63e31fdd0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -123,6 +123,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max98927-objs := max98927.o +snd-soc-max98520-objs := max98520.o snd-soc-max98373-objs := max98373.o snd-soc-max98373-i2c-objs := max98373-i2c.o snd-soc-max98373-sdw-objs := max98373-sdw.o @@ -450,6 +451,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o +obj-$(CONFIG_SND_SOC_MAX98520) += snd-soc-max98520.o obj-$(CONFIG_SND_SOC_MAX98373) += snd-soc-max98373.o obj-$(CONFIG_SND_SOC_MAX98373_I2C) += snd-soc-max98373-i2c.o obj-$(CONFIG_SND_SOC_MAX98373_SDW) += snd-soc-max98373-sdw.o diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c new file mode 100644 index 000000000000..3f87410d0608 --- /dev/null +++ b/sound/soc/codecs/max98520.c @@ -0,0 +1,798 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2021, Maxim Integrated + +#include <linux/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98520.h" + +static struct reg_default max98520_reg[] = { + {MAX98520_R2000_SW_RESET, 0x00}, + {MAX98520_R2001_STATUS_1, 0x00}, + {MAX98520_R2002_STATUS_2, 0x00}, + {MAX98520_R2020_THERM_WARN_THRESH, 0x46}, + {MAX98520_R2021_THERM_SHDN_THRESH, 0x64}, + {MAX98520_R2022_THERM_HYSTERESIS, 0x02}, + {MAX98520_R2023_THERM_FOLDBACK_SET, 0x31}, + {MAX98520_R2027_THERM_FOLDBACK_EN, 0x01}, + {MAX98520_R2030_CLK_MON_CTRL, 0x00}, + {MAX98520_R2037_ERR_MON_CTRL, 0x01}, + {MAX98520_R2040_PCM_MODE_CFG, 0xC0}, + {MAX98520_R2041_PCM_CLK_SETUP, 0x04}, + {MAX98520_R2042_PCM_SR_SETUP, 0x08}, + {MAX98520_R2043_PCM_RX_SRC1, 0x00}, + {MAX98520_R2044_PCM_RX_SRC2, 0x00}, + {MAX98520_R204F_PCM_RX_EN, 0x00}, + {MAX98520_R2090_AMP_VOL_CTRL, 0x00}, + {MAX98520_R2091_AMP_PATH_GAIN, 0x03}, + {MAX98520_R2092_AMP_DSP_CFG, 0x02}, + {MAX98520_R2094_SSM_CFG, 0x01}, + {MAX98520_R2095_AMP_CFG, 0xF0}, + {MAX98520_R209F_AMP_EN, 0x00}, + {MAX98520_R20B0_ADC_SR, 0x00}, + {MAX98520_R20B1_ADC_RESOLUTION, 0x00}, + {MAX98520_R20B2_ADC_PVDD0_CFG, 0x02}, + {MAX98520_R20B3_ADC_THERMAL_CFG, 0x02}, + {MAX98520_R20B4_ADC_READBACK_CTRL, 0x00}, + {MAX98520_R20B5_ADC_READBACK_UPDATE, 0x00}, + {MAX98520_R20B6_ADC_PVDD_READBACK_MSB, 0x00}, + {MAX98520_R20B7_ADC_PVDD_READBACK_LSB, 0x00}, + {MAX98520_R20B8_ADC_TEMP_READBACK_MSB, 0x00}, + {MAX98520_R20B9_ADC_TEMP_READBACK_LSB, 0x00}, + {MAX98520_R20BA_ADC_LOW_PVDD_READBACK_MSB, 0xFF}, + {MAX98520_R20BB_ADC_LOW_READBACK_LSB, 0x01}, + {MAX98520_R20BC_ADC_HIGH_TEMP_READBACK_MSB, 0x00}, + {MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB, 0x00}, + {MAX98520_R20CF_MEAS_ADC_CFG, 0x00}, + {MAX98520_R20D0_DHT_CFG1, 0x00}, + {MAX98520_R20D1_LIMITER_CFG1, 0x08}, + {MAX98520_R20D2_LIMITER_CFG2, 0x00}, + {MAX98520_R20D3_DHT_CFG2, 0x14}, + {MAX98520_R20D4_DHT_CFG3, 0x02}, + {MAX98520_R20D5_DHT_CFG4, 0x04}, + {MAX98520_R20D6_DHT_HYSTERESIS_CFG, 0x07}, + {MAX98520_R20D8_DHT_EN, 0x00}, + {MAX98520_R210E_AUTO_RESTART_BEHAVIOR, 0x00}, + {MAX98520_R210F_GLOBAL_EN, 0x00}, + {MAX98520_R21FF_REVISION_ID, 0x00}, +}; + +static int max98520_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + unsigned int format = 0; + unsigned int invert = 0; + + dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(component->dev, "DAI invert mode unsupported\n"); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format = MAX98520_PCM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + format = MAX98520_PCM_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98520_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98520_PCM_FORMAT_TDM_MODE0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_FORMAT_MASK, + format << MAX98520_PCM_MODE_CFG_FORMAT_SHIFT); + + return 0; +} + +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, 320, +}; + +static int max98520_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} + +static int max98520_set_clock(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98520->ch_size; + int value; + + if (!max98520->tdm_mode) { + /* BCLK configuration */ + value = max98520_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_CLK_SETUP_BSEL_MASK, + value); + } + dev_dbg(component->dev, "%s tdm_mode:%d out\n", __func__, max98520->tdm_mode); + return 0; +} + +static int max98520_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + goto err; + } + + max98520->ch_size = snd_pcm_format_width(params_format(params)); + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(component->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98520_PCM_SR_8000; + break; + case 11025: + sampling_rate = MAX98520_PCM_SR_11025; + break; + case 12000: + sampling_rate = MAX98520_PCM_SR_12000; + break; + case 16000: + sampling_rate = MAX98520_PCM_SR_16000; + break; + case 22050: + sampling_rate = MAX98520_PCM_SR_22050; + break; + case 24000: + sampling_rate = MAX98520_PCM_SR_24000; + break; + case 32000: + sampling_rate = MAX98520_PCM_SR_32000; + break; + case 44100: + sampling_rate = MAX98520_PCM_SR_44100; + break; + case 48000: + sampling_rate = MAX98520_PCM_SR_48000; + break; + case 88200: + sampling_rate = MAX98520_PCM_SR_88200; + break; + case 96000: + sampling_rate = MAX98520_PCM_SR_96000; + break; + case 176400: + sampling_rate = MAX98520_PCM_SR_176400; + break; + case 192000: + sampling_rate = MAX98520_PCM_SR_192000; + break; + default: + dev_err(component->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + + dev_dbg(component->dev, " %s ch_size: %d, sampling rate : %d out\n", __func__, + snd_pcm_format_width(params_format(params)), params_rate(params)); + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98520->regmap, + MAX98520_R2042_PCM_SR_SETUP, + MAX98520_PCM_SR_MASK, + sampling_rate); + + return max98520_set_clock(component, params); +err: + dev_dbg(component->dev, "%s out error", __func__); + return -EINVAL; +} + +static int max98520_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + int bsel = 0; + unsigned int chan_sz = 0; + + if (!tx_mask && !rx_mask && !slots && !slot_width) + max98520->tdm_mode = false; + else + max98520->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98520_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(component->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_update_bits(max98520->regmap, + MAX98520_R2044_PCM_RX_SRC2, + MAX98520_PCM_DMIX_CH0_SRC_MASK, + rx_mask); + regmap_update_bits(max98520->regmap, + MAX98520_R2044_PCM_RX_SRC2, + MAX98520_PCM_DMIX_CH1_SRC_MASK, + rx_mask << MAX98520_PCM_DMIX_CH1_SHIFT); + + return 0; +} + +#define MAX98520_RATES SNDRV_PCM_RATE_8000_192000 + +#define MAX98520_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops max98520_dai_ops = { + .set_fmt = max98520_dai_set_fmt, + .hw_params = max98520_dai_hw_params, + .set_tdm_slot = max98520_dai_tdm_slot, +}; + +static int max98520_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_dbg(component->dev, " AMP ON\n"); + + regmap_write(max98520->regmap, MAX98520_R209F_AMP_EN, 1); + regmap_write(max98520->regmap, MAX98520_R210F_GLOBAL_EN, 1); + usleep_range(30000, 31000); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, " AMP OFF\n"); + + regmap_write(max98520->regmap, MAX98520_R210F_GLOBAL_EN, 0); + regmap_write(max98520->regmap, MAX98520_R209F_AMP_EN, 0); + usleep_range(30000, 31000); + break; + default: + return 0; + } + return 0; +} + +static const char * const max98520_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98520_R2043_PCM_RX_SRC1, + 0, 3, max98520_switch_text); + +static const struct snd_kcontrol_new max98520_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_kcontrol_new max98520_left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("PCM_INPUT_CH0", MAX98520_R2044_PCM_RX_SRC2, 0, 0x0, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH1", MAX98520_R2044_PCM_RX_SRC2, 0, 0x1, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH2", MAX98520_R2044_PCM_RX_SRC2, 0, 0x2, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH3", MAX98520_R2044_PCM_RX_SRC2, 0, 0x3, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH4", MAX98520_R2044_PCM_RX_SRC2, 0, 0x4, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH5", MAX98520_R2044_PCM_RX_SRC2, 0, 0x5, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH6", MAX98520_R2044_PCM_RX_SRC2, 0, 0x6, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH7", MAX98520_R2044_PCM_RX_SRC2, 0, 0x7, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH8", MAX98520_R2044_PCM_RX_SRC2, 0, 0x8, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH9", MAX98520_R2044_PCM_RX_SRC2, 0, 0x9, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH10", MAX98520_R2044_PCM_RX_SRC2, 0, 0xa, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH11", MAX98520_R2044_PCM_RX_SRC2, 0, 0xb, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH12", MAX98520_R2044_PCM_RX_SRC2, 0, 0xc, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH13", MAX98520_R2044_PCM_RX_SRC2, 0, 0xd, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH14", MAX98520_R2044_PCM_RX_SRC2, 0, 0xe, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH15", MAX98520_R2044_PCM_RX_SRC2, 0, 0xf, 0), +}; + +static const struct snd_kcontrol_new max98520_right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("PCM_INPUT_CH0", MAX98520_R2044_PCM_RX_SRC2, 4, 0x0, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH1", MAX98520_R2044_PCM_RX_SRC2, 4, 0x1, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH2", MAX98520_R2044_PCM_RX_SRC2, 4, 0x2, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH3", MAX98520_R2044_PCM_RX_SRC2, 4, 0x3, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH4", MAX98520_R2044_PCM_RX_SRC2, 4, 0x4, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH5", MAX98520_R2044_PCM_RX_SRC2, 4, 0x5, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH6", MAX98520_R2044_PCM_RX_SRC2, 4, 0x6, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH7", MAX98520_R2044_PCM_RX_SRC2, 4, 0x7, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH8", MAX98520_R2044_PCM_RX_SRC2, 4, 0x8, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH9", MAX98520_R2044_PCM_RX_SRC2, 4, 0x9, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH10", MAX98520_R2044_PCM_RX_SRC2, 4, 0xa, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH11", MAX98520_R2044_PCM_RX_SRC2, 4, 0xb, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH12", MAX98520_R2044_PCM_RX_SRC2, 4, 0xc, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH13", MAX98520_R2044_PCM_RX_SRC2, 4, 0xd, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH14", MAX98520_R2044_PCM_RX_SRC2, 4, 0xe, 0), + SOC_DAPM_SINGLE("PCM_INPUT_CH15", MAX98520_R2044_PCM_RX_SRC2, 4, 0xf, 0), +}; + +static const struct snd_soc_dapm_widget max98520_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", + SND_SOC_NOPM, 0, 0, max98520_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98520_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input Selection", SND_SOC_NOPM, 0, 0, + &max98520_left_input_mixer_controls[0], + ARRAY_SIZE(max98520_left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input Selection", SND_SOC_NOPM, 0, 0, + &max98520_right_input_mixer_controls[0], + ARRAY_SIZE(max98520_right_input_mixer_controls)), +}; + +static const DECLARE_TLV_DB_SCALE(max98520_digital_tlv, -6300, 50, 1); +static const DECLARE_TLV_DB_SCALE(max98520_spk_tlv, -600, 300, 0); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_lim_thresh_tlv, + 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_hysteresis_tlv, + 0, 3, TLV_DB_SCALE_ITEM(100, 100, 0), + 4, 7, TLV_DB_SCALE_ITEM(600, 200, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_rotation_point_tlv, + 0, 1, TLV_DB_SCALE_ITEM(-1500, 300, 0), + 2, 4, TLV_DB_SCALE_ITEM(-1000, 200, 0), + 5, 10, TLV_DB_SCALE_ITEM(-500, 100, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_supply_hr_tlv, + 0, 16, TLV_DB_SCALE_ITEM(-2000, 250, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_max_atten_tlv, + 1, 20, TLV_DB_SCALE_ITEM(-2000, 100, 0), +); + +static const char * const max98520_dht_attack_rate_text[] = { + "20us", "40us", "80us", "160us", "320us", "640us", + "1.28ms", "2.56ms", "5.12ms", "10.24ms", "20.48ms", "40.96ms", + "81.92ms", "163.84ms" +}; + +static SOC_ENUM_SINGLE_DECL(max98520_dht_attack_rate_enum, + MAX98520_R20D4_DHT_CFG3, 0, + max98520_dht_attack_rate_text); + +static const char * const max98520_dht_release_rate_text[] = { + "2ms", "4ms", "8ms", "16ms", "32ms", "64ms", "128ms", "256ms", "512ms", + "1.024s", "2.048s", "4.096s", "8.192s", "16.384s" +}; + +static SOC_ENUM_SINGLE_DECL(max98520_dht_release_rate_enum, + MAX98520_R20D5_DHT_CFG4, 0, + max98520_dht_release_rate_text); + +static bool max98520_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98520_R2000_SW_RESET: + case MAX98520_R2027_THERM_FOLDBACK_EN: + case MAX98520_R2030_CLK_MON_CTRL: + case MAX98520_R2037_ERR_MON_CTRL: + case MAX98520_R204F_PCM_RX_EN: + case MAX98520_R209F_AMP_EN: + case MAX98520_R20CF_MEAS_ADC_CFG: + case MAX98520_R20D8_DHT_EN: + case MAX98520_R21FF_REVISION_ID: + case MAX98520_R2001_STATUS_1... MAX98520_R2002_STATUS_2: + case MAX98520_R2020_THERM_WARN_THRESH... MAX98520_R2023_THERM_FOLDBACK_SET: + case MAX98520_R2040_PCM_MODE_CFG... MAX98520_R2044_PCM_RX_SRC2: + case MAX98520_R2090_AMP_VOL_CTRL... MAX98520_R2092_AMP_DSP_CFG: + case MAX98520_R2094_SSM_CFG... MAX98520_R2095_AMP_CFG: + case MAX98520_R20B0_ADC_SR... MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB: + case MAX98520_R20D0_DHT_CFG1... MAX98520_R20D6_DHT_HYSTERESIS_CFG: + case MAX98520_R210E_AUTO_RESTART_BEHAVIOR... MAX98520_R210F_GLOBAL_EN: + case MAX98520_R2161_BOOST_TM1... MAX98520_R2163_BOOST_TM3: + return true; + default: + return false; + } +}; + +static bool max98520_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98520_R210F_GLOBAL_EN: + case MAX98520_R21FF_REVISION_ID: + case MAX98520_R2000_SW_RESET: + case MAX98520_R2001_STATUS_1 ... MAX98520_R2002_STATUS_2: + case MAX98520_R20B4_ADC_READBACK_CTRL + ... MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB: + return true; + default: + return false; + } +} + +static const struct snd_kcontrol_new max98520_snd_controls[] = { +/* Volume */ +SOC_SINGLE_TLV("Digital Volume", MAX98520_R2090_AMP_VOL_CTRL, + 0, 0x7F, 1, max98520_digital_tlv), +SOC_SINGLE_TLV("Speaker Volume", MAX98520_R2091_AMP_PATH_GAIN, + 0, 0x5, 0, max98520_spk_tlv), +/* Volume Ramp Up/Down Enable*/ +SOC_SINGLE("Ramp Up Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_VOL_RMPUP_SHIFT, 1, 0), +SOC_SINGLE("Ramp Down Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_VOL_RMPDN_SHIFT, 1, 0), +/* Clock Monitor Enable */ +SOC_SINGLE("CLK Monitor Switch", MAX98520_R2037_ERR_MON_CTRL, + MAX98520_CTRL_CMON_EN_SHIFT, 1, 0), +/* Clock Monitor Config */ +SOC_SINGLE("CLKMON Autorestart Switch", MAX98520_R2030_CLK_MON_CTRL, + MAX98520_CMON_AUTORESTART_SHIFT, 1, 0), +/* Dither Enable */ +SOC_SINGLE("Dither Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_DITH_EN_SHIFT, 1, 0), +/* DC Blocker Enable */ +SOC_SINGLE("DC Blocker Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_DCBLK_EN_SHIFT, 1, 0), +/* Speaker Safe Mode Enable */ +SOC_SINGLE("Speaker Safemode Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_SAFE_EN_SHIFT, 1, 0), +/* AMP SSM Enable */ +SOC_SINGLE("CP Bypass Switch", MAX98520_R2094_SSM_CFG, + MAX98520_SSM_RCVR_MODE_SHIFT, 1, 0), +/* Dynamic Headroom Tracking */ +SOC_SINGLE("DHT Switch", MAX98520_R20D8_DHT_EN, 0, 1, 0), +SOC_SINGLE("DHT Limiter Mode", MAX98520_R20D2_LIMITER_CFG2, + MAX98520_DHT_LIMITER_MODE_SHIFT, 1, 0), +SOC_SINGLE("DHT Hysteresis Switch", MAX98520_R20D6_DHT_HYSTERESIS_CFG, + MAX98520_DHT_HYSTERESIS_SWITCH_SHIFT, 1, 0), +SOC_SINGLE_TLV("DHT Rot Pnt", MAX98520_R20D0_DHT_CFG1, + MAX98520_DHT_VROT_PNT_SHIFT, 10, 1, max98520_dht_rotation_point_tlv), +SOC_SINGLE_TLV("DHT Supply Headroom", MAX98520_R20D1_LIMITER_CFG1, + MAX98520_DHT_SUPPLY_HR_SHIFT, 16, 0, max98520_dht_supply_hr_tlv), +SOC_SINGLE_TLV("DHT Limiter Threshold", MAX98520_R20D2_LIMITER_CFG2, + MAX98520_DHT_LIMITER_THRESHOLD_SHIFT, 0xF, 1, max98520_dht_lim_thresh_tlv), +SOC_SINGLE_TLV("DHT Max Attenuation", MAX98520_R20D3_DHT_CFG2, + MAX98520_DHT_MAX_ATTEN_SHIFT, 20, 1, max98520_dht_max_atten_tlv), +SOC_SINGLE_TLV("DHT Hysteresis", MAX98520_R20D6_DHT_HYSTERESIS_CFG, + MAX98520_DHT_HYSTERESIS_SHIFT, 0x7, 0, max98520_dht_hysteresis_tlv), +SOC_ENUM("DHT Attack Rate", max98520_dht_attack_rate_enum), +SOC_ENUM("DHT Release Rate", max98520_dht_release_rate_enum), +/* ADC configuration */ +SOC_SINGLE("ADC PVDD CH Switch", MAX98520_R20CF_MEAS_ADC_CFG, 0, 1, 0), +SOC_SINGLE("ADC PVDD FLT Switch", MAX98520_R20B2_ADC_PVDD0_CFG, MAX98520_FLT_EN_SHIFT, 1, 0), +SOC_SINGLE("ADC TEMP FLT Switch", MAX98520_R20B3_ADC_THERMAL_CFG, MAX98520_FLT_EN_SHIFT, 1, 0), +SOC_SINGLE("ADC PVDD MSB", MAX98520_R20B6_ADC_PVDD_READBACK_MSB, 0, 0xFF, 0), +SOC_SINGLE("ADC PVDD LSB", MAX98520_R20B7_ADC_PVDD_READBACK_LSB, 0, 0x01, 0), +SOC_SINGLE("ADC TEMP MSB", MAX98520_R20B8_ADC_TEMP_READBACK_MSB, 0, 0xFF, 0), +SOC_SINGLE("ADC TEMP LSB", MAX98520_R20B9_ADC_TEMP_READBACK_LSB, 0, 0x01, 0), +}; + +static const struct snd_soc_dapm_route max98520_audio_map[] = { + /* Plabyack */ + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98520_dai[] = { + { + .name = "max98520-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98520_RATES, + .formats = MAX98520_FORMATS, + }, + .ops = &max98520_dai_ops, + } + +}; + +static void max98520_reset(struct max98520_priv *max98520, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_write(max98520->regmap, MAX98520_R2000_SW_RESET, 1); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98520->regmap, MAX98520_R21FF_REVISION_ID, &reg); + if (!ret) + return; + + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + +static int max98520_probe(struct snd_soc_component *component) +{ + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + + /* Software Reset */ + max98520_reset(max98520, component->dev); + + /* Enable DC blocker */ + regmap_update_bits(max98520->regmap, MAX98520_R2092_AMP_DSP_CFG, 1, 1); + /* Enable Clock Monitor Auto-restart */ + regmap_write(max98520->regmap, MAX98520_R2030_CLK_MON_CTRL, 0x1); + + /* set Rx Enable */ + regmap_update_bits(max98520->regmap, + MAX98520_R204F_PCM_RX_EN, + MAX98520_PCM_RX_EN_MASK, + 1); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int max98520_suspend(struct device *dev) +{ + struct max98520_priv *max98520 = dev_get_drvdata(dev); + + regcache_cache_only(max98520->regmap, true); + regcache_mark_dirty(max98520->regmap); + return 0; +} + +static int max98520_resume(struct device *dev) +{ + struct max98520_priv *max98520 = dev_get_drvdata(dev); + + regcache_cache_only(max98520->regmap, false); + max98520_reset(max98520, dev); + regcache_sync(max98520->regmap); + return 0; +} +#endif + +static const struct dev_pm_ops max98520_pm = { + SET_SYSTEM_SLEEP_PM_OPS(max98520_suspend, max98520_resume) +}; + +static const struct snd_soc_component_driver soc_codec_dev_max98520 = { + .probe = max98520_probe, + .controls = max98520_snd_controls, + .num_controls = ARRAY_SIZE(max98520_snd_controls), + .dapm_widgets = max98520_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98520_dapm_widgets), + .dapm_routes = max98520_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98520_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config max98520_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98520_R21FF_REVISION_ID, + .reg_defaults = max98520_reg, + .num_reg_defaults = ARRAY_SIZE(max98520_reg), + .readable_reg = max98520_readable_register, + .volatile_reg = max98520_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static void max98520_power_on(struct max98520_priv *max98520, bool poweron) +{ + if (max98520->reset_gpio) + gpiod_set_value_cansleep(max98520->reset_gpio, !poweron); +} + +static int max98520_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + int ret = 0; + int reg = 0; + struct max98520_priv *max98520 = NULL; + struct i2c_adapter *adapter = to_i2c_adapter(i2c->dev.parent); + + ret = i2c_check_functionality(adapter, I2C_FUNC_SMBUS_BYTE | I2C_FUNC_SMBUS_BYTE_DATA); + if (!ret) { + dev_err(&i2c->dev, "I2C check functionality failed\n"); + return -ENXIO; + } + + max98520 = devm_kzalloc(&i2c->dev, sizeof(*max98520), GFP_KERNEL); + + if (!max98520) { + ret = -ENOMEM; + return ret; + } + i2c_set_clientdata(i2c, max98520); + + /* regmap initialization */ + max98520->regmap = + devm_regmap_init_i2c(i2c, &max98520_regmap); + if (IS_ERR(max98520->regmap)) { + ret = PTR_ERR(max98520->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + /* Power on device */ + max98520->reset_gpio = devm_gpiod_get_optional(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (max98520->reset_gpio) { + if (IS_ERR(max98520->reset_gpio)) { + ret = PTR_ERR(max98520->reset_gpio); + dev_err(&i2c->dev, "Unable to request GPIO pin: %d.\n", ret); + return ret; + } + + max98520_power_on(max98520, 1); + } + + /* Check Revision ID */ + ret = regmap_read(max98520->regmap, MAX98520_R21FF_REVISION_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, + "Failed to read: 0x%02X\n", MAX98520_R21FF_REVISION_ID); + return ret; + } + dev_info(&i2c->dev, "MAX98520 revisionID: 0x%02X\n", reg); + + /* codec registration */ + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_max98520, + max98520_dai, ARRAY_SIZE(max98520_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static const struct i2c_device_id max98520_i2c_id[] = { + { "max98520", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98520_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98520_of_match[] = { + { .compatible = "maxim,max98520", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98520_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98520_acpi_match[] = { + { "MX98520", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98520_acpi_match); +#endif + +static struct i2c_driver max98520_i2c_driver = { + .driver = { + .name = "max98520", + .of_match_table = of_match_ptr(max98520_of_match), + .acpi_match_table = ACPI_PTR(max98520_acpi_match), + .pm = &max98520_pm, + }, + .probe = max98520_i2c_probe, + .id_table = max98520_i2c_id, +}; + +module_i2c_driver(max98520_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98520 driver"); +MODULE_AUTHOR("George Song <george.song(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/max98520.h b/sound/soc/codecs/max98520.h new file mode 100644 index 000000000000..89a95c25afcf --- /dev/null +++ b/sound/soc/codecs/max98520.h @@ -0,0 +1,159 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2021, Maxim Integrated. + */ + +#ifndef _MAX98520_H +#define _MAX98520_H + +#define MAX98520_R2000_SW_RESET 0x2000 +#define MAX98520_R2001_STATUS_1 0x2001 +#define MAX98520_R2002_STATUS_2 0x2002 +#define MAX98520_R2020_THERM_WARN_THRESH 0x2020 +#define MAX98520_R2021_THERM_SHDN_THRESH 0x2021 +#define MAX98520_R2022_THERM_HYSTERESIS 0x2022 +#define MAX98520_R2023_THERM_FOLDBACK_SET 0x2023 +#define MAX98520_R2027_THERM_FOLDBACK_EN 0x2027 +#define MAX98520_R2030_CLK_MON_CTRL 0x2030 +#define MAX98520_R2037_ERR_MON_CTRL 0x2037 +#define MAX98520_R2040_PCM_MODE_CFG 0x2040 +#define MAX98520_R2041_PCM_CLK_SETUP 0x2041 +#define MAX98520_R2042_PCM_SR_SETUP 0x2042 +#define MAX98520_R2043_PCM_RX_SRC1 0x2043 +#define MAX98520_R2044_PCM_RX_SRC2 0x2044 +#define MAX98520_R204F_PCM_RX_EN 0x204F +#define MAX98520_R2090_AMP_VOL_CTRL 0x2090 +#define MAX98520_R2091_AMP_PATH_GAIN 0x2091 +#define MAX98520_R2092_AMP_DSP_CFG 0x2092 +#define MAX98520_R2094_SSM_CFG 0x2094 +#define MAX98520_R2095_AMP_CFG 0x2095 +#define MAX98520_R209F_AMP_EN 0x209F +#define MAX98520_R20B0_ADC_SR 0x20B0 +#define MAX98520_R20B1_ADC_RESOLUTION 0x20B1 +#define MAX98520_R20B2_ADC_PVDD0_CFG 0x20B2 +#define MAX98520_R20B3_ADC_THERMAL_CFG 0x20B3 +#define MAX98520_R20B4_ADC_READBACK_CTRL 0x20B4 +#define MAX98520_R20B5_ADC_READBACK_UPDATE 0x20B5 +#define MAX98520_R20B6_ADC_PVDD_READBACK_MSB 0x20B6 +#define MAX98520_R20B7_ADC_PVDD_READBACK_LSB 0x20B7 +#define MAX98520_R20B8_ADC_TEMP_READBACK_MSB 0x20B8 +#define MAX98520_R20B9_ADC_TEMP_READBACK_LSB 0x20B9 +#define MAX98520_R20BA_ADC_LOW_PVDD_READBACK_MSB 0x20BA +#define MAX98520_R20BB_ADC_LOW_READBACK_LSB 0x20BB +#define MAX98520_R20BC_ADC_HIGH_TEMP_READBACK_MSB 0x20BC +#define MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB 0x20BD +#define MAX98520_R20CF_MEAS_ADC_CFG 0x20CF +#define MAX98520_R20D0_DHT_CFG1 0x20D0 +#define MAX98520_R20D1_LIMITER_CFG1 0x20D1 +#define MAX98520_R20D2_LIMITER_CFG2 0x20D2 +#define MAX98520_R20D3_DHT_CFG2 0x20D3 +#define MAX98520_R20D4_DHT_CFG3 0x20D4 +#define MAX98520_R20D5_DHT_CFG4 0x20D5 +#define MAX98520_R20D6_DHT_HYSTERESIS_CFG 0x20D6 +#define MAX98520_R20D8_DHT_EN 0x20D8 +#define MAX98520_R210E_AUTO_RESTART_BEHAVIOR 0x210E +#define MAX98520_R210F_GLOBAL_EN 0x210F +#define MAX98520_R2161_BOOST_TM1 0x2161 +#define MAX98520_R2162_BOOST_TM2 0x2162 +#define MAX98520_R2163_BOOST_TM3 0x2163 +#define MAX98520_R21FF_REVISION_ID 0x21FF + +/* MAX98520_R2030_CLK_MON_CTRL */ +#define MAX98520_CMON_AUTORESTART_SHIFT (0) + +/* MAX98520_R2037_ERR_MON_CTRL */ +#define MAX98520_CTRL_CMON_EN_SHIFT (0) + +/* MAX98520_R2040_PCM_MODE_CFG */ +#define MAX98520_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) +#define MAX98520_PCM_MODE_CFG_FORMAT_SHIFT (3) +#define MAX98520_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 2) +#define MAX98520_PCM_FORMAT_I2S (0x0 << 3) +#define MAX98520_PCM_FORMAT_LJ (0x1 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE0 (0x3 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE1 (0x4 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE2 (0x5 << 3) +#define MAX98520_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98520_R2041_PCM_CLK_SETUP */ +#define MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 4) +#define MAX98520_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98520_R2042_PCM_SR_SETUP */ +#define MAX98520_PCM_SR_SHIFT (0) +#define MAX98520_IVADC_SR_SHIFT (4) +#define MAX98520_PCM_SR_MASK (0xF << MAX98520_PCM_SR_SHIFT) +#define MAX98520_IVADC_SR_MASK (0xF << MAX98520_IVADC_SR_SHIFT) +#define MAX98520_PCM_SR_8000 (0x0) +#define MAX98520_PCM_SR_11025 (0x1) +#define MAX98520_PCM_SR_12000 (0x2) +#define MAX98520_PCM_SR_16000 (0x3) +#define MAX98520_PCM_SR_22050 (0x4) +#define MAX98520_PCM_SR_24000 (0x5) +#define MAX98520_PCM_SR_32000 (0x6) +#define MAX98520_PCM_SR_44100 (0x7) +#define MAX98520_PCM_SR_48000 (0x8) +#define MAX98520_PCM_SR_88200 (0x9) +#define MAX98520_PCM_SR_96000 (0xA) +#define MAX98520_PCM_SR_176400 (0xB) +#define MAX98520_PCM_SR_192000 (0xC) + +/* MAX98520_R2044_PCM_RX_SRC2 */ +#define MAX98520_PCM_DMIX_CH1_SHIFT (0xF << 0) +#define MAX98520_PCM_DMIX_CH0_SRC_MASK (0xF << 0) +#define MAX98520_PCM_DMIX_CH1_SRC_MASK (0xF << MAX98520_PCM_DMIX_CH1_SHIFT) + +/* MAX98520_R204F_PCM_RX_EN */ +#define MAX98520_PCM_RX_EN_MASK (0x1 << 0) +#define MAX98520_PCM_RX_BYP_EN_MASK (0x1 << 1) + +/* MAX98520_R2092_AMP_DSP_CFG */ +#define MAX98520_DSP_SPK_DCBLK_EN_SHIFT (0) +#define MAX98520_DSP_SPK_DITH_EN_SHIFT (1) +#define MAX98520_DSP_SPK_INVERT_SHIFT (2) +#define MAX98520_DSP_SPK_VOL_RMPUP_SHIFT (3) +#define MAX98520_DSP_SPK_VOL_RMPDN_SHIFT (4) +#define MAX98520_DSP_SPK_SAFE_EN_SHIFT (5) + +#define MAX98520_SPK_SAFE_EN_MASK (0x1 << MAX98520_DSP_SPK_SAFE_EN_SHIFT) + +/* MAX98520_R2094_SSM_CFG */ +#define MAX98520_SSM_EN_SHIFT (0) +#define MAX98520_SSM_MOD_SHIFT (1) +#define MAX98520_SSM_RCVR_MODE_SHIFT (3) + +/* MAX98520_R2095_AMP_CFG */ +#define MAX98520_CFG_DYN_MODE_SHIFT (4) +#define MAX98520_CFG_SPK_MODE_SHIFT (3) + +/* MAX98520_R20D0_DHT_CFG1 */ +#define MAX98520_DHT_VROT_PNT_SHIFT (0) + +/* MAX98520_R20D1_LIMITER_CFG1 */ +#define MAX98520_DHT_SUPPLY_HR_SHIFT (0) + +/* MAX98520_R20D2_DHT_CFG2 */ +#define MAX98520_DHT_LIMITER_MODE_SHIFT (0) +#define MAX98520_DHT_LIMITER_THRESHOLD_SHIFT (1) + +/* MAX98520_R20D3_DHT_CFG2 */ +#define MAX98520_DHT_MAX_ATTEN_SHIFT (0) + +/* MAX98520_R20D6_DHT_HYSTERESIS_CFG */ +#define MAX98520_DHT_HYSTERESIS_SWITCH_SHIFT (0) +#define MAX98520_DHT_HYSTERESIS_SHIFT (1) + +/* MAX98520_R20B2_ADC_PVDD0_CFG, MAX98520_R20B3_ADC_THERMAL_CFG */ +#define MAX98520_FLT_EN_SHIFT (4) + +struct max98520_priv { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + unsigned int ch_size; + bool tdm_mode; +}; +#endif + -- 2.25.1

4 6

[PATCH] ASoC: rockchip: use generic DMA engine configuration
by John Keeping 20 Oct '21

20 Oct '21

This effectively reverts commit 75b31192fe6a ("ASoC: rockchip: add config for rockchip dmaengine pcm register"). There doesn't seem to be any rationale given for why these specific values are helpful. The generic DMA engine provides sensible defaults here and works well with Rockchip I2S. In fact the period size here is really quite restrictive when dealing with 8 channels of 32-bit data as the effective period size is just 256 frames. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/Makefile | 3 +-- sound/soc/rockchip/rockchip_i2s.c | 3 +-- sound/soc/rockchip/rockchip_pcm.c | 44 ------------------------------- sound/soc/rockchip/rockchip_pcm.h | 11 -------- 4 files changed, 2 insertions(+), 59 deletions(-) delete mode 100644 sound/soc/rockchip/rockchip_pcm.c delete mode 100644 sound/soc/rockchip/rockchip_pcm.h diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index b10f5e7b136d..6a3e61178152 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -2,11 +2,10 @@ # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o snd-soc-rockchip-i2s-tdm-objs := rockchip_i2s_tdm.o -snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S_TDM) += snd-soc-rockchip-i2s-tdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 7e89f5b0c237..a6d7656c206e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -20,7 +20,6 @@ #include <sound/dmaengine_pcm.h> #include "rockchip_i2s.h" -#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -756,7 +755,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = rockchip_pcm_platform_register(&pdev->dev); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); goto err_suspend; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c deleted file mode 100644 index 02254e42135e..000000000000 --- a/sound/soc/rockchip/rockchip_pcm.c +++ /dev/null @@ -1,44 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#include <linux/device.h> -#include <linux/init.h> -#include <linux/module.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/dmaengine_pcm.h> - -#include "rockchip_pcm.h" - -static const struct snd_pcm_hardware snd_rockchip_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 52, - .buffer_bytes_max = 64 * 1024, - .fifo_size = 32, -}; - -static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { - .pcm_hardware = &snd_rockchip_hardware, - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .prealloc_buffer_size = 32 * 1024, -}; - -int rockchip_pcm_platform_register(struct device *dev) -{ - return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); -} -EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); - -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h deleted file mode 100644 index 7f00e2ce3603..000000000000 --- a/sound/soc/rockchip/rockchip_pcm.h +++ /dev/null @@ -1,11 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#ifndef _ROCKCHIP_PCM_H -#define _ROCKCHIP_PCM_H - -int rockchip_pcm_platform_register(struct device *dev); - -#endif -- 2.33.1

3 5

[v3 1/2] ASoC: dt-bindings: max98520: add initial bindings
by George Song 20 Oct '21

20 Oct '21

add initial bindings for max98520 audio amplifier Signed-off-by: George Song <george.song(a)maximintegrated.com> --- .../bindings/sound/maxim,max98520.yaml | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..b6509cb2c8e0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <george.song(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; + -- 2.25.1

3 6

[PATCH 0/3] ASoC/pdx86/input: Introduce and use soc_intel_is_*() helpers
by Hans de Goede 19 Oct '21

19 Oct '21

Hi All, We have been open-coding x86_match_cpu() checks for enabling some SoC specific behavior in various places. The sound/soc/intel drivers used to also open-code this but this was cleaned up a while ago introducing a number of soc_intel_is_*() helpers. This series moves the definition of these helpers to a more public place and uses it in a couple of more places outside the sound tree. Mark, I know we are a bit late in the cycle, but if you can pick up patch 1/3 (assuming on one objects) for 5.16, then the rest can be applied after 5.16-rc1 is out. Regards, Hans Hans de Goede (3): ASoC: Intel: Move soc_intel_is_foo() helpers to a generic header platform/x86: intel_int0002_vgpio: Use the new soc_intel_is_byt/cht helpers Input: axp20x-pek - Use new soc_intel_is_cht() helper drivers/input/misc/axp20x-pek.c | 26 ++------- drivers/platform/x86/intel/int0002_vgpio.c | 14 +---- include/linux/platform_data/x86/soc.h | 65 ++++++++++++++++++++++ sound/soc/intel/common/soc-intel-quirks.h | 51 +---------------- 4 files changed, 75 insertions(+), 81 deletions(-) create mode 100644 include/linux/platform_data/x86/soc.h -- 2.31.1

4 12

[alsa-devel] [PATCH v7 0/8] Fix year 2038 issue for sound subsystem
by Arnd Bergmann 19 Oct '21

19 Oct '21

This is a series I worked on with Baolin in 2017 and 2018, but we never quite managed to finish up the last pieces. During the ALSA developer meetup at ELC-E 2018 in Edinburgh, a decision was made to go with this approach for keeping best compatibility with existing source code, and then I failed to follow up by resending the patches. Now I have patches for all remaining time_t uses in the kernel, so it's absolutely time to revisit them. I have done more review of the patches myself and found a couple of minor issues that I have fixed up, otherwise the series is still the same as before. Conceptually, the idea of these patches is: - 64-bit applications should see no changes at all, neither compile-time nor run-time. - 32-bit code compiled with a 64-bit time_t currently does not work with ALSA, and requires kernel changes and/or sound/asound.h changes - Most 32-bit code using these interfaces will work correctly on a modified kernel, with or without the uapi header changes. - 32-bit code using SNDRV_TIMER_IOCTL_TREAD requires the updated header file for 64-bit time_t support - 32-bit i386 user space with 64-bit time_t is broken for SNDRV_PCM_IOCTL_STATUS, SNDRV_RAWMIDI_IOCTL_STATUS and SNDRV_PCM_IOCTL_SYNC_PTR because of i386 alignment. This is also addressed by the updated uapi header. - PCM mmap is currently supported on native x86 kernels (both 32-bit and 64-bit) but not for compat mode. This series breaks the 32-bit native mmap support for 32-bit time_t, but instead allows it for 64-bit time_t on both native and compat kernels. This seems to be the best trade-off, as mmap support is optional already, and most 32-bit code runs in compat mode anyway. - I've tried to avoid breaking compilation of 32-bit code as much as possible. Anything that does break however is likely code that is already broken on 64-bit time_t and needs source changes to fix them. I hope I addressed all review comments by now, so please pull this for linux-5.6. A git branch with the same contents is available for testing at [1]. Arnd [1] https://git.kernel.org/pub/scm/linux/kernel/git/arnd/playground.git y2038-alsa-v7 [2] https://lore.kernel.org/lkml/CAK8P3a2Os66+iwQYf97qh05W2JP8rmWao8zmKoHiXqVHv… Changes since v6: (Arnd): - Add a patch to update the API versions - Hide a timespec reference in #ifndef __KERNEL__ to remove the last reference to time_t - Use a more readable way to do padding and describe it in the changelog - Rebase to linux-5.5-rc1, changing include/sound/soc-component.h and sound/drivers/aloop.c as needed. Changes since v5 (Arnd): - Rebased to linux-5.4-rc4 - Updated to completely remove timespec and time_t references from alsa - found and fixed a few bugs Changes since v4 (Baolin): - Add patch 5 to change trigger_tstamp member of struct snd_pcm_runtime. - Add patch 8 to change internal timespec. - Add more explanation in commit message. - Use ktime_get_real_ts64() in patch 6. - Split common code out into a separate function in patch 6. - Fix tu->tread bug in patch 6 and remove #if __BITS_PER_LONG == 64 macro. Changes since v3: - Move struct snd_pcm_status32 to pcm.h file. - Modify comments and commit message. - Add new patch2 ~ patch6. Changes since v2: - Renamed all structures to make clear. - Remove CONFIG_X86_X32 macro and introduced new compat_snd_pcm_status64_x86_32. Changes since v1: - Add one macro for struct snd_pcm_status_32 which only active in 32bits kernel. - Convert pcm_compat.c to use struct snd_pcm_status_64. - Convert pcm_native.c to use struct snd_pcm_status_64. --- Arnd Bergmann (3): ALSA: move snd_pcm_ioctl_sync_ptr_compat into pcm_native.c ALSA: add new 32-bit layout for snd_pcm_mmap_status/control ALSA: bump uapi version numbers Baolin Wang (6): ALSA: Replace timespec with timespec64 ALSA: Avoid using timespec for struct snd_timer_status ALSA: Avoid using timespec for struct snd_ctl_elem_value ALSA: Avoid using timespec for struct snd_pcm_status ALSA: Avoid using timespec for struct snd_rawmidi_status ALSA: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 74 ++++++-- include/sound/soc-component.h | 4 +- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 145 +++++++++++++-- sound/core/pcm.c | 12 +- sound/core/pcm_compat.c | 282 ++++++++---------------------- sound/core/pcm_lib.c | 38 ++-- sound/core/pcm_native.c | 226 +++++++++++++++++++++--- sound/core/rawmidi.c | 132 +++++++++++--- sound/core/rawmidi_compat.c | 87 +++------ sound/core/timer.c | 229 ++++++++++++++++++------ sound/core/timer_compat.c | 62 +------ sound/drivers/aloop.c | 2 +- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 15 files changed, 817 insertions(+), 494 deletions(-) -- 2.20.0

5 46

Re: [PATCH] sound/soc: adds TAS5754M digital input amplifier component driver
by Takashi Iwai 19 Oct '21

19 Oct '21

1 0

[PATCH v2] sound/usb : Fix mic sound on Jieli webcam
by Marco Giunta 19 Oct '21

19 Oct '21

Hi, this is my second attempt (https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178361.…) to fix mic sound on a Jieli webcam. I found that the mic works only when ep packet size is set to wMaxPacketSize, so I've removed the datainterval hack. I also fixed the problem with the volume control (mixer). Now the mic sound works (no more Minion voice) and there are no more errors in syslog about volume range. I arbitrarily choose a resolution value (16): I read in a comment that there should be no more than 255 levels, so 4096 (max volume) / 16 = 0-255 ;-) Could you review this patch? Thanks, Marco Jieli Technology USB Webcam microphone needs some quirks to work. Signed-off-by: Marco Giunta <giun7a(a)gmail.com> --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 6 ++++++ 2 files changed, 13 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9b713b4a5..20ef12dd8 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1198,6 +1198,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 1; } break; + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 326d1b0ea..2263e43fd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1534,6 +1534,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */ case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */ + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ return true; } @@ -1874,6 +1875,11 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ + /* mic works only when ep packet size is set to wMaxPacketSize */ + fp->attributes |= UAC_EP_CS_ATTR_FILL_MAX; + break; + } } -- 2.31.1

2 4

[PATCH 1/2] ALSA: memalloc: Drop superfluous snd_dma_buffer_sync() declaration
by Takashi Iwai 19 Oct '21

19 Oct '21

snd_dma_buffer_sync() is declared twice, and the one outside the ifdef CONFIG_HAS_DMA could lead to a build error when CONFIG_HAS_DMA=n. As it's an overlooked leftover after rebase, drop this line. Fixes: a25684a95646 ("ALSA: memalloc: Support for non-contiguous page allocation") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Link: https://lore.kernel.org/r/20211019165402.4fa82c38@canb.auug.org.au Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index df615052dad4..653dfffb3ac8 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -97,9 +97,6 @@ static inline void snd_dma_buffer_sync(struct snd_dma_buffer *dmab, enum snd_dma_sync_mode mode) {} #endif -void snd_dma_buffer_sync(struct snd_dma_buffer *dmab, - enum snd_dma_sync_mode mode); - dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, size_t offset); struct page *snd_sgbuf_get_page(struct snd_dma_buffer *dmab, size_t offset); unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab, -- 2.26.2

1 1

[PATCH] ASoC: amd: remove duplicate include in acp5x-mach.c
by cgel.zte＠gmail.com 19 Oct '21

19 Oct '21

From: Lv Ruyi <lv.ruyi(a)zte.com.cn> 'linux/io.h' and 'linux/module.h' included in 'acp5x-mach.c' is duplicated.Only one duplicate header file is reserved. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Lv Ruyi <lv.ruyi(a)zte.com.cn> --- sound/soc/amd/vangogh/acp5x-mach.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index 14cf325e4b23..2d9229ba9c59 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -8,16 +8,14 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <linux/module.h> -#include <linux/io.h> #include <sound/pcm.h> #include <sound/pcm_params.h> - #include <sound/jack.h> + #include <linux/clk.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/io.h> -- 2.25.1

1 0

[PATCH v2 0/4] ALSA: hda: New NHLT functions and cleanup
by Cezary Rojewski 19 Oct '21

19 Oct '21

Changes add two crucial functions: endpoint presence-check and retrieval of endpoint's BLOB (hardware configuration) to NHLT API. Few cleanups accompany the above: Work is done to align NHLT-struct naming with other, commonly used ACPI-structs. While cleaning up, don't forget about "is DMIC in NHLT?" check. No need to check for channel count or anything DMIC-configuration related, just straight up verify link_type presence. Changes in v2: - patch "ALSA hda: Drop device-argument in NHLT functions" has been dropped - updated newly added declarations in intel-nhlt.h so warning: "no-previous-prototype-for-function" and error: "use-of-undeclared-identifier" are no longer observed when CONFIG_SND_INTEL_NHLT is not enabled - added Mark's tag to the last patch of the series Amadeusz Sławiński (4): ALSA: hda: Follow ACPI convention in NHLT struct naming ALSA: hda: Fill gaps in NHLT endpoint-interface ALSA: hda: Simplify DMIC-in-NHLT check ASoC: Intel: Skylake: Use NHLT API to search for blob include/sound/intel-nhlt.h | 53 ++++++++---- sound/hda/intel-dsp-config.c | 4 +- sound/hda/intel-nhlt.c | 110 ++++++++++++++++++++++++- sound/soc/intel/skylake/skl-nhlt.c | 108 +----------------------- sound/soc/intel/skylake/skl-pcm.c | 3 + sound/soc/intel/skylake/skl-topology.c | 29 ++++--- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.h | 6 +- sound/soc/sof/intel/hda.c | 2 +- 9 files changed, 171 insertions(+), 145 deletions(-) -- 2.25.1

3 6

[PATCH v3 0/4] ASoC: SOF: Intel: power optimizations with HDaudio SPIB register
by Pierre-Louis Bossart 18 Oct '21

18 Oct '21

This patchset was initially provided in a larger series that was split in two [1]. This part only provides support for the SPIB register support, added on Intel platforms since Skylake (2015). The use of the SPIB register helps reduce power consumption - though to a smaller degree than DMI_L1. This hardware capability is however incompatible with userspace-initiated rewinds typically used by PulseAudio. In the past (2015..2017) Intel suggested an API extension to let applications disable rewinds. At the time the feedback was that such a capability was too Intel-specific and SPIB remained unused except for loading DSP code. We now see devices with smaller batteries being released, and it's time to revisit Linux support for SPIB to extend battery life. In this update the rewinds are disabled via an opt-in kernel parameter. In the previous reviews, there was consensus that a Kconfig option was too complicated for distributions to set, and we are missing a TBD API to expose such capabilities to user-space. The debate on whether or not to use rewinds, and the impact of disabling rewinds, will likely be closed when Intel releases the 'deep-buffer' support, currently under development [2][3]. With this solution, rewinds will not be needed, ever. When an application deals with content that is not latency-sensitive (e.g. music playback), it will be able to reduce power consumption by selecting a different PCM device with increased buffering capabilities. Low-latency streams will be handled by the 'regular' path. In other words, the impossible compromise between power and latency will be handled with different PCM devices/profiles for the same endpoint, and we can push the design of capability negotiation to a later time when all the building blocks (firmware topology, kernel, userspace) are ready - we still have firmware xruns, DPCM race conditions to solve, and a need to describe these alternate PCM devices with UCM using 'modifiers'. [1] https://lore.kernel.org/alsa-devel/20210610205326.1176400-1-pierre-louis.bo… [2] https://github.com/thesofproject/linux/pull/3146 [3] https://github.com/thesofproject/sof/pull/4611 Changes since v2: Added check on appl_ptr (Takashi) Moved rewind detection to deal with SYNC_PTR (Takashi) Pierre-Louis Bossart (2): ALSA: pcm: unconditionally check if appl_ptr is in 0..boundary range ALSA: pcm: introduce INFO_NO_REWINDS flag Ranjani Sridharan (2): ASOC: SOF: pcm: add .ack callback support ASoC: SOF: Intel: add .ack support for HDaudio platforms include/uapi/sound/asound.h | 2 +- sound/core/pcm_lib.c | 17 +++++++++++++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-pcm.c | 41 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda-stream.c | 2 ++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 1 + sound/soc/sof/ops.h | 10 ++++++++ sound/soc/sof/pcm.c | 9 +++++++ sound/soc/sof/sof-priv.h | 3 +++ 11 files changed, 85 insertions(+), 3 deletions(-) -- 2.25.1

2 12

[PATCH] ASoC: Intel: sof_sdw: Initialize the sof_sdw_quirk with RT711_JD_NULL
by Chris Chiu 18 Oct '21

18 Oct '21

The jd_src of RT711 which is initialized in rt711/rt711_sdca_init will be overridden by rt711/rt711_sdca_add_codec_device_props when the sof_sdw_quirk is not RT711_JD_NULL. It will force the JD mode to RT711_JD1 and cause confusion while debugging the JD mode of the boards without quirk. Initialize sof_sdw_quirk with RT711_JD_NULL to honor the jd_src value in rt711/rt711_sdca init. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6b06248a9327..d05c0565e09c 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -15,7 +15,7 @@ #include "sof_sdw_common.h" #include "../../codecs/rt711.h" -unsigned long sof_sdw_quirk = RT711_JD1; +unsigned long sof_sdw_quirk = RT711_JD_NULL; static int quirk_override = -1; module_param_named(quirk, quirk_override, int, 0444); MODULE_PARM_DESC(quirk, "Board-specific quirk override"); -- 2.20.1

2 3

[PATCH] ALSA: uapi: Fix a C++ style comment in asound.h
by Takashi Iwai 18 Oct '21

18 Oct '21

UAPI header should have no C++ style comment but only in the traditional C style comment, but there is still one place we used it mistakenly. This patch corrects it. Fixes: 542283566679 ("ALSA: ctl: remove unused macro for timestamping of elem_value") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 5859ca0a1439..5fbb79e30819 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -1002,7 +1002,7 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_WRITE (1<<1) #define SNDRV_CTL_ELEM_ACCESS_READWRITE (SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE) #define SNDRV_CTL_ELEM_ACCESS_VOLATILE (1<<2) /* control value may be changed without a notification */ -// (1 << 3) is unused. +/* (1 << 3) is unused. */ #define SNDRV_CTL_ELEM_ACCESS_TLV_READ (1<<4) /* TLV read is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE (1<<5) /* TLV write is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE (SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) -- 2.26.2

2 1

[PATCH 0/5] ALSA: hda: New NHLT functions and cleanup
by Cezary Rojewski 18 Oct '21

18 Oct '21

Changes add two crucial functions: endpoint presence-check and retrieval of endpoint's BLOB (hardware configuration) to NHLT API. Few cleanups accompany the above: First, drop device pointer usage in NHLT interface as those function don't really make use of it. Work is done to align NHLT-struct naming with other, commonly used ACPI-structs. While cleaning up, don't forget about "is DMIC in NHLT?" check. No need to check for channel count or anything DMIC-configuration related, just straight up verify link_type presence. Amadeusz Sławiński (5): ALSA: hda: Drop device-argument in NHLT functions ALSA: hda: Follow ACPI convention in NHLT struct naming ALSA: hda: Fill gaps in NHLT endpoint-interface ALSA: hda: Simplify DMIC-in-NHLT check ASoC: Intel: Skylake: Use NHLT API to search for blob include/sound/intel-nhlt.h | 37 +++++-- sound/hda/intel-dsp-config.c | 6 +- sound/hda/intel-nhlt.c | 127 ++++++++++++++++++++++--- sound/soc/intel/skylake/skl-nhlt.c | 108 +-------------------- sound/soc/intel/skylake/skl-pcm.c | 3 + sound/soc/intel/skylake/skl-topology.c | 27 +++--- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.c | 5 +- sound/soc/intel/skylake/skl.h | 6 +- sound/soc/sof/intel/hda.c | 6 +- 10 files changed, 173 insertions(+), 153 deletions(-) -- 2.25.1

5 15

[PATCH] ALSA: firewire: Fix C++ style comments in uapi header
by Takashi Iwai 18 Oct '21

18 Oct '21

UAPI headers are built with -std=c90 and C++ style comments are explicitly prohibited. The recent commit overlooked the rule and caused the error at header installation. This patch corrects those. Fixes: bea36afa102e ("ALSA: firewire-motu: add message parser to gather meter information in register DSP model") Fixes: 90b28f3bb85c ("ALSA: firewire-motu: add message parser for meter information in command DSP model") Fixes: 634ec0b2906e ("ALSA: firewire-motu: notify event for parameter change in register DSP model") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Link: https://lore.kernel.org/r/20211018113812.0a16efb0@canb.auug.org.au Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/firewire.h | 70 ++++++++++++++++++----------------- 1 file changed, 37 insertions(+), 33 deletions(-) diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index 76190a0cb069..e52a97b3ceaa 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -68,8 +68,8 @@ struct snd_firewire_event_tascam_control { struct snd_firewire_event_motu_register_dsp_change { unsigned int type; - __u32 count; // The number of changes. - __u32 changes[]; // Encoded event for change of register DSP. + __u32 count; /* The number of changes. */ + __u32 changes[]; /* Encoded event for change of register DSP. */ }; union snd_firewire_event { @@ -119,25 +119,27 @@ struct snd_firewire_tascam_state { __be32 data[SNDRV_FIREWIRE_TASCAM_STATE_COUNT]; }; -// In below MOTU models, software is allowed to control their DSP by accessing to registers. -// - 828mk2 -// - 896hd -// - Traveler -// - 8 pre -// - Ultralite -// - 4 pre -// - Audio Express -// -// On the other hand, the status of DSP is split into specific messages included in the sequence of -// isochronous packet. ALSA firewire-motu driver gathers the messages and allow userspace applications -// to read it via ioctl. In 828mk2, 896hd, and Traveler, hardware meter for all of physical inputs -// are put into the message, while one pair of physical outputs is selected. The selection is done by -// LSB one byte in asynchronous write quadlet transaction to 0x'ffff'f000'0b2c. -// -// I note that V3HD/V4HD uses asynchronous transaction for the purpose. The destination address is -// registered to 0x'ffff'f000'0b38 and '0b3c by asynchronous write quadlet request. The size of -// message differs between 23 and 51 quadlets. For the case, the number of mixer bus can be extended -// up to 12. +/* + * In below MOTU models, software is allowed to control their DSP by accessing to registers. + * - 828mk2 + * - 896hd + * - Traveler + * - 8 pre + * - Ultralite + * - 4 pre + * - Audio Express + * + * On the other hand, the status of DSP is split into specific messages included in the sequence of + * isochronous packet. ALSA firewire-motu driver gathers the messages and allow userspace applications + * to read it via ioctl. In 828mk2, 896hd, and Traveler, hardware meter for all of physical inputs + * are put into the message, while one pair of physical outputs is selected. The selection is done by + * LSB one byte in asynchronous write quadlet transaction to 0x'ffff'f000'0b2c. + * + * I note that V3HD/V4HD uses asynchronous transaction for the purpose. The destination address is + * registered to 0x'ffff'f000'0b38 and '0b3c by asynchronous write quadlet request. The size of + * message differs between 23 and 51 quadlets. For the case, the number of mixer bus can be extended + * up to 12. + */ #define SNDRV_FIREWIRE_MOTU_REGISTER_DSP_METER_COUNT 40 @@ -219,18 +221,20 @@ struct snd_firewire_motu_register_dsp_parameter { __u8 reserved[64]; }; -// In below MOTU models, software is allowed to control their DSP by command in frame of -// asynchronous transaction to 0x'ffff'0001'0000: -// -// - 828 mk3 (FireWire only and Hybrid) -// - 896 mk3 (FireWire only and Hybrid) -// - Ultralite mk3 (FireWire only and Hybrid) -// - Traveler mk3 -// - Track 16 -// -// On the other hand, the states of hardware meter is split into specific messages included in the -// sequence of isochronous packet. ALSA firewire-motu driver gathers the message and allow userspace -// application to read it via ioctl. +/* + * In below MOTU models, software is allowed to control their DSP by command in frame of + * asynchronous transaction to 0x'ffff'0001'0000: + * + * - 828 mk3 (FireWire only and Hybrid) + * - 896 mk3 (FireWire only and Hybrid) + * - Ultralite mk3 (FireWire only and Hybrid) + * - Traveler mk3 + * - Track 16 + * + * On the other hand, the states of hardware meter is split into specific messages included in the + * sequence of isochronous packet. ALSA firewire-motu driver gathers the message and allow userspace + * application to read it via ioctl. + */ #define SNDRV_FIREWIRE_MOTU_COMMAND_DSP_METER_COUNT 400 -- 2.26.2

2 1

[PATCH v3] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 18 Oct '21

18 Oct '21

The document was merged as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [1] revealed that the commit 1afc60e00de3 breaks dt_binding_check due to dt-bindings/clock/mt8192-clk.h doesn't exist. As a temporary fix, commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") reverted commit 1afc60e00de3. dt-bindings/clock/mt8192-clk.h is in mainline per commit f35f1a23e0e1 ("clk: mediatek: Add dt-bindings of MT8192 clocks"). Re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v2 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189058.ht…): - Simplify the commit message. Changes from v1 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189048.ht…): - Add more context to the commit message. .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..0170c93802d2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.309.g3052b89438-goog

4 5

[PATCH RFC] virtio: wrap config->reset calls
by Michael S. Tsirkin 18 Oct '21

18 Oct '21

This will enable cleanups down the road. The idea is to disable cbs, then add "flush_queued_cbs" callback as a parameter, this way drivers can flush any work queued after callbacks have been disabled. Signed-off-by: Michael S. Tsirkin <mst(a)redhat.com> --- arch/um/drivers/virt-pci.c | 2 +- drivers/block/virtio_blk.c | 4 ++-- drivers/bluetooth/virtio_bt.c | 2 +- drivers/char/hw_random/virtio-rng.c | 2 +- drivers/char/virtio_console.c | 4 ++-- drivers/crypto/virtio/virtio_crypto_core.c | 8 ++++---- drivers/firmware/arm_scmi/virtio.c | 2 +- drivers/gpio/gpio-virtio.c | 2 +- drivers/gpu/drm/virtio/virtgpu_kms.c | 2 +- drivers/i2c/busses/i2c-virtio.c | 2 +- drivers/iommu/virtio-iommu.c | 2 +- drivers/net/caif/caif_virtio.c | 2 +- drivers/net/virtio_net.c | 4 ++-- drivers/net/wireless/mac80211_hwsim.c | 2 +- drivers/nvdimm/virtio_pmem.c | 2 +- drivers/rpmsg/virtio_rpmsg_bus.c | 2 +- drivers/scsi/virtio_scsi.c | 2 +- drivers/virtio/virtio.c | 5 +++++ drivers/virtio/virtio_balloon.c | 2 +- drivers/virtio/virtio_input.c | 2 +- drivers/virtio/virtio_mem.c | 2 +- fs/fuse/virtio_fs.c | 4 ++-- include/linux/virtio.h | 1 + net/9p/trans_virtio.c | 2 +- net/vmw_vsock/virtio_transport.c | 4 ++-- sound/virtio/virtio_card.c | 4 ++-- 26 files changed, 39 insertions(+), 33 deletions(-) diff --git a/arch/um/drivers/virt-pci.c b/arch/um/drivers/virt-pci.c index c08066633023..22c4d87c9c15 100644 --- a/arch/um/drivers/virt-pci.c +++ b/arch/um/drivers/virt-pci.c @@ -616,7 +616,7 @@ static void um_pci_virtio_remove(struct virtio_device *vdev) int i; /* Stop all virtqueues */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); device_set_wakeup_enable(&vdev->dev, false); diff --git a/drivers/block/virtio_blk.c b/drivers/block/virtio_blk.c index 303caf2d17d0..83d0af3fbf30 100644 --- a/drivers/block/virtio_blk.c +++ b/drivers/block/virtio_blk.c @@ -910,7 +910,7 @@ static void virtblk_remove(struct virtio_device *vdev) mutex_lock(&vblk->vdev_mutex); /* Stop all the virtqueues. */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); /* Virtqueues are stopped, nothing can use vblk->vdev anymore. */ vblk->vdev = NULL; @@ -929,7 +929,7 @@ static int virtblk_freeze(struct virtio_device *vdev) struct virtio_blk *vblk = vdev->priv; /* Ensure we don't receive any more interrupts */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); /* Make sure no work handler is accessing the device. */ flush_work(&vblk->config_work); diff --git a/drivers/bluetooth/virtio_bt.c b/drivers/bluetooth/virtio_bt.c index 57908ce4fae8..24a9258962fa 100644 --- a/drivers/bluetooth/virtio_bt.c +++ b/drivers/bluetooth/virtio_bt.c @@ -364,7 +364,7 @@ static void virtbt_remove(struct virtio_device *vdev) struct hci_dev *hdev = vbt->hdev; hci_unregister_dev(hdev); - vdev->config->reset(vdev); + virtio_reset_device(vdev); hci_free_dev(hdev); vbt->hdev = NULL; diff --git a/drivers/char/hw_random/virtio-rng.c b/drivers/char/hw_random/virtio-rng.c index a90001e02bf7..95980489514b 100644 --- a/drivers/char/hw_random/virtio-rng.c +++ b/drivers/char/hw_random/virtio-rng.c @@ -134,7 +134,7 @@ static void remove_common(struct virtio_device *vdev) vi->hwrng_removed = true; vi->data_avail = 0; complete(&vi->have_data); - vdev->config->reset(vdev); + virtio_reset_device(vdev); vi->busy = false; if (vi->hwrng_register_done) hwrng_unregister(&vi->hwrng); diff --git a/drivers/char/virtio_console.c b/drivers/char/virtio_console.c index 7eaf303a7a86..08bbd693436f 100644 --- a/drivers/char/virtio_console.c +++ b/drivers/char/virtio_console.c @@ -1957,7 +1957,7 @@ static void virtcons_remove(struct virtio_device *vdev) spin_unlock_irq(&pdrvdata_lock); /* Disable interrupts for vqs */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); /* Finish up work that's lined up */ if (use_multiport(portdev)) cancel_work_sync(&portdev->control_work); @@ -2139,7 +2139,7 @@ static int virtcons_freeze(struct virtio_device *vdev) portdev = vdev->priv; - vdev->config->reset(vdev); + virtio_reset_device(vdev); if (use_multiport(portdev)) virtqueue_disable_cb(portdev->c_ivq); diff --git a/drivers/crypto/virtio/virtio_crypto_core.c b/drivers/crypto/virtio/virtio_crypto_core.c index e2375d992308..8e977b7627cb 100644 --- a/drivers/crypto/virtio/virtio_crypto_core.c +++ b/drivers/crypto/virtio/virtio_crypto_core.c @@ -404,7 +404,7 @@ static int virtcrypto_probe(struct virtio_device *vdev) free_engines: virtcrypto_clear_crypto_engines(vcrypto); free_vqs: - vcrypto->vdev->config->reset(vdev); + virtio_reset_device(vdev); virtcrypto_del_vqs(vcrypto); free_dev: virtcrypto_devmgr_rm_dev(vcrypto); @@ -436,7 +436,7 @@ static void virtcrypto_remove(struct virtio_device *vdev) if (virtcrypto_dev_started(vcrypto)) virtcrypto_dev_stop(vcrypto); - vdev->config->reset(vdev); + virtio_reset_device(vdev); virtcrypto_free_unused_reqs(vcrypto); virtcrypto_clear_crypto_engines(vcrypto); virtcrypto_del_vqs(vcrypto); @@ -456,7 +456,7 @@ static int virtcrypto_freeze(struct virtio_device *vdev) { struct virtio_crypto *vcrypto = vdev->priv; - vdev->config->reset(vdev); + virtio_reset_device(vdev); virtcrypto_free_unused_reqs(vcrypto); if (virtcrypto_dev_started(vcrypto)) virtcrypto_dev_stop(vcrypto); @@ -492,7 +492,7 @@ static int virtcrypto_restore(struct virtio_device *vdev) free_engines: virtcrypto_clear_crypto_engines(vcrypto); free_vqs: - vcrypto->vdev->config->reset(vdev); + virtio_reset_device(vdev); virtcrypto_del_vqs(vcrypto); return err; } diff --git a/drivers/firmware/arm_scmi/virtio.c b/drivers/firmware/arm_scmi/virtio.c index 11e8efb71375..6b8d93fe8848 100644 --- a/drivers/firmware/arm_scmi/virtio.c +++ b/drivers/firmware/arm_scmi/virtio.c @@ -452,7 +452,7 @@ static void scmi_vio_remove(struct virtio_device *vdev) * outstanding message on any vqueue to be ignored by complete_cb: now * we can just stop processing buffers and destroy the vqueues. */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); /* Ensure scmi_vdev is visible as NULL */ smp_store_mb(scmi_vdev, NULL); diff --git a/drivers/gpio/gpio-virtio.c b/drivers/gpio/gpio-virtio.c index d24f1c9264bc..5029f01966f4 100644 --- a/drivers/gpio/gpio-virtio.c +++ b/drivers/gpio/gpio-virtio.c @@ -203,7 +203,7 @@ static void virtio_gpio_request_vq(struct virtqueue *vq) static void virtio_gpio_free_vqs(struct virtio_device *vdev) { - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); } diff --git a/drivers/gpu/drm/virtio/virtgpu_kms.c b/drivers/gpu/drm/virtio/virtgpu_kms.c index f3379059f324..6aa605b8d3a1 100644 --- a/drivers/gpu/drm/virtio/virtgpu_kms.c +++ b/drivers/gpu/drm/virtio/virtgpu_kms.c @@ -257,7 +257,7 @@ void virtio_gpu_deinit(struct drm_device *dev) flush_work(&vgdev->ctrlq.dequeue_work); flush_work(&vgdev->cursorq.dequeue_work); flush_work(&vgdev->config_changed_work); - vgdev->vdev->config->reset(vgdev->vdev); + virtio_reset_device(vgdev->vdev); vgdev->vdev->config->del_vqs(vgdev->vdev); } diff --git a/drivers/i2c/busses/i2c-virtio.c b/drivers/i2c/busses/i2c-virtio.c index f10a603b13fb..eb3261ac64dc 100644 --- a/drivers/i2c/busses/i2c-virtio.c +++ b/drivers/i2c/busses/i2c-virtio.c @@ -177,7 +177,7 @@ static int virtio_i2c_xfer(struct i2c_adapter *adap, struct i2c_msg *msgs, static void virtio_i2c_del_vqs(struct virtio_device *vdev) { - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); } diff --git a/drivers/iommu/virtio-iommu.c b/drivers/iommu/virtio-iommu.c index 80930ce04a16..1d4e1e7cf175 100644 --- a/drivers/iommu/virtio-iommu.c +++ b/drivers/iommu/virtio-iommu.c @@ -1115,7 +1115,7 @@ static void viommu_remove(struct virtio_device *vdev) iommu_device_unregister(&viommu->iommu); /* Stop all virtqueues */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); dev_info(&vdev->dev, "device removed\n"); diff --git a/drivers/net/caif/caif_virtio.c b/drivers/net/caif/caif_virtio.c index 91230894692d..444ef6a342f6 100644 --- a/drivers/net/caif/caif_virtio.c +++ b/drivers/net/caif/caif_virtio.c @@ -754,7 +754,7 @@ static void cfv_remove(struct virtio_device *vdev) debugfs_remove_recursive(cfv->debugfs); vringh_kiov_cleanup(&cfv->ctx.riov); - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->vringh_config->del_vrhs(cfv->vdev); cfv->vr_rx = NULL; vdev->config->del_vqs(cfv->vdev); diff --git a/drivers/net/virtio_net.c b/drivers/net/virtio_net.c index 79bd2585ec6b..8c10fcad73a4 100644 --- a/drivers/net/virtio_net.c +++ b/drivers/net/virtio_net.c @@ -3274,7 +3274,7 @@ static int virtnet_probe(struct virtio_device *vdev) return 0; free_unregister_netdev: - vi->vdev->config->reset(vdev); + virtio_reset_device(vdev); unregister_netdev(dev); free_failover: @@ -3290,7 +3290,7 @@ static int virtnet_probe(struct virtio_device *vdev) static void remove_vq_common(struct virtnet_info *vi) { - vi->vdev->config->reset(vi->vdev); + virtio_reset_device(vi->vdev); /* Free unused buffers in both send and recv, if any. */ free_unused_bufs(vi); diff --git a/drivers/net/wireless/mac80211_hwsim.c b/drivers/net/wireless/mac80211_hwsim.c index 0adae76eb8df..9ee430c1d4a2 100644 --- a/drivers/net/wireless/mac80211_hwsim.c +++ b/drivers/net/wireless/mac80211_hwsim.c @@ -4369,7 +4369,7 @@ static void remove_vqs(struct virtio_device *vdev) { int i; - vdev->config->reset(vdev); + virtio_reset_device(vdev); for (i = 0; i < ARRAY_SIZE(hwsim_vqs); i++) { struct virtqueue *vq = hwsim_vqs[i]; diff --git a/drivers/nvdimm/virtio_pmem.c b/drivers/nvdimm/virtio_pmem.c index 726c7354d465..995b6cdc67ed 100644 --- a/drivers/nvdimm/virtio_pmem.c +++ b/drivers/nvdimm/virtio_pmem.c @@ -105,7 +105,7 @@ static void virtio_pmem_remove(struct virtio_device *vdev) nvdimm_bus_unregister(nvdimm_bus); vdev->config->del_vqs(vdev); - vdev->config->reset(vdev); + virtio_reset_device(vdev); } static struct virtio_driver virtio_pmem_driver = { diff --git a/drivers/rpmsg/virtio_rpmsg_bus.c b/drivers/rpmsg/virtio_rpmsg_bus.c index 8e49a3bacfc7..6a11952822df 100644 --- a/drivers/rpmsg/virtio_rpmsg_bus.c +++ b/drivers/rpmsg/virtio_rpmsg_bus.c @@ -1015,7 +1015,7 @@ static void rpmsg_remove(struct virtio_device *vdev) size_t total_buf_space = vrp->num_bufs * vrp->buf_size; int ret; - vdev->config->reset(vdev); + virtio_reset_device(vdev); ret = device_for_each_child(&vdev->dev, NULL, rpmsg_remove_device); if (ret) diff --git a/drivers/scsi/virtio_scsi.c b/drivers/scsi/virtio_scsi.c index 07d0250f17c3..f2502a8a5213 100644 --- a/drivers/scsi/virtio_scsi.c +++ b/drivers/scsi/virtio_scsi.c @@ -778,7 +778,7 @@ static void virtscsi_init_vq(struct virtio_scsi_vq *virtscsi_vq, static void virtscsi_remove_vqs(struct virtio_device *vdev) { /* Stop all the virtqueues. */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); } diff --git a/drivers/virtio/virtio.c b/drivers/virtio/virtio.c index 0a5b54034d4b..a67dd0eca578 100644 --- a/drivers/virtio/virtio.c +++ b/drivers/virtio/virtio.c @@ -204,6 +204,11 @@ int virtio_finalize_features(struct virtio_device *dev) } EXPORT_SYMBOL_GPL(virtio_finalize_features); +static void virtio_reset_device(struct virtio_device *dev) +{ + dev->config->reset(dev); +} + static int virtio_dev_probe(struct device *_d) { int err, i; diff --git a/drivers/virtio/virtio_balloon.c b/drivers/virtio/virtio_balloon.c index c22ff0117b46..f4c34a2a6b8e 100644 --- a/drivers/virtio/virtio_balloon.c +++ b/drivers/virtio/virtio_balloon.c @@ -1056,7 +1056,7 @@ static void remove_common(struct virtio_balloon *vb) return_free_pages_to_mm(vb, ULONG_MAX); /* Now we reset the device so we can clean up the queues. */ - vb->vdev->config->reset(vb->vdev); + virtio_reset_device(vb->vdev); vb->vdev->config->del_vqs(vb->vdev); } diff --git a/drivers/virtio/virtio_input.c b/drivers/virtio/virtio_input.c index ce51ae165943..3aa46703872d 100644 --- a/drivers/virtio/virtio_input.c +++ b/drivers/virtio/virtio_input.c @@ -347,7 +347,7 @@ static void virtinput_remove(struct virtio_device *vdev) spin_unlock_irqrestore(&vi->lock, flags); input_unregister_device(vi->idev); - vdev->config->reset(vdev); + virtio_reset_device(vdev); while ((buf = virtqueue_detach_unused_buf(vi->sts)) != NULL) kfree(buf); vdev->config->del_vqs(vdev); diff --git a/drivers/virtio/virtio_mem.c b/drivers/virtio/virtio_mem.c index bef8ad6bf466..3bab0a625a4b 100644 --- a/drivers/virtio/virtio_mem.c +++ b/drivers/virtio/virtio_mem.c @@ -2722,7 +2722,7 @@ static void virtio_mem_remove(struct virtio_device *vdev) } /* reset the device and cleanup the queues */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); kfree(vm); diff --git a/fs/fuse/virtio_fs.c b/fs/fuse/virtio_fs.c index 0ad89c6629d7..27c3b74070a2 100644 --- a/fs/fuse/virtio_fs.c +++ b/fs/fuse/virtio_fs.c @@ -895,7 +895,7 @@ static int virtio_fs_probe(struct virtio_device *vdev) return 0; out_vqs: - vdev->config->reset(vdev); + virtio_reset_device(vdev); virtio_fs_cleanup_vqs(vdev, fs); kfree(fs->vqs); @@ -927,7 +927,7 @@ static void virtio_fs_remove(struct virtio_device *vdev) list_del_init(&fs->list); virtio_fs_stop_all_queues(fs); virtio_fs_drain_all_queues_locked(fs); - vdev->config->reset(vdev); + virtio_reset_device(vdev); virtio_fs_cleanup_vqs(vdev, fs); vdev->priv = NULL; diff --git a/include/linux/virtio.h b/include/linux/virtio.h index 41edbc01ffa4..72292a62cd90 100644 --- a/include/linux/virtio.h +++ b/include/linux/virtio.h @@ -138,6 +138,7 @@ int virtio_finalize_features(struct virtio_device *dev); int virtio_device_freeze(struct virtio_device *dev); int virtio_device_restore(struct virtio_device *dev); #endif +void virtio_reset_device(struct virtio_device *dev); size_t virtio_max_dma_size(struct virtio_device *vdev); diff --git a/net/9p/trans_virtio.c b/net/9p/trans_virtio.c index 490a4c900339..19c69821dd04 100644 --- a/net/9p/trans_virtio.c +++ b/net/9p/trans_virtio.c @@ -721,7 +721,7 @@ static void p9_virtio_remove(struct virtio_device *vdev) mutex_unlock(&virtio_9p_lock); - vdev->config->reset(vdev); + virtio_reset_device(vdev); vdev->config->del_vqs(vdev); sysfs_remove_file(&(vdev->dev.kobj), &dev_attr_mount_tag.attr); diff --git a/net/vmw_vsock/virtio_transport.c b/net/vmw_vsock/virtio_transport.c index 4f7c99dfd16c..fb3302fff627 100644 --- a/net/vmw_vsock/virtio_transport.c +++ b/net/vmw_vsock/virtio_transport.c @@ -665,7 +665,7 @@ static void virtio_vsock_remove(struct virtio_device *vdev) vsock_for_each_connected_socket(virtio_vsock_reset_sock); /* Stop all work handlers to make sure no one is accessing the device, - * so we can safely call vdev->config->reset(). + * so we can safely call virtio_reset_device(). */ mutex_lock(&vsock->rx_lock); vsock->rx_run = false; @@ -682,7 +682,7 @@ static void virtio_vsock_remove(struct virtio_device *vdev) /* Flush all device writes and interrupts, device will not use any * more buffers. */ - vdev->config->reset(vdev); + virtio_reset_device(vdev); mutex_lock(&vsock->rx_lock); while ((pkt = virtqueue_detach_unused_buf(vsock->vqs[VSOCK_VQ_RX]))) diff --git a/sound/virtio/virtio_card.c b/sound/virtio/virtio_card.c index 150ab3e37013..e2847c040f75 100644 --- a/sound/virtio/virtio_card.c +++ b/sound/virtio/virtio_card.c @@ -350,7 +350,7 @@ static void virtsnd_remove(struct virtio_device *vdev) snd_card_free(snd->card); vdev->config->del_vqs(vdev); - vdev->config->reset(vdev); + virtio_reset_device(vdev); for (i = 0; snd->substreams && i < snd->nsubstreams; ++i) { struct virtio_pcm_substream *vss = &snd->substreams[i]; @@ -379,7 +379,7 @@ static int virtsnd_freeze(struct virtio_device *vdev) virtsnd_ctl_msg_cancel_all(snd); vdev->config->del_vqs(vdev); - vdev->config->reset(vdev); + virtio_reset_device(vdev); for (i = 0; i < snd->nsubstreams; ++i) cancel_work_sync(&snd->substreams[i].elapsed_period); -- MST

12 13

[v2 1/2] ASoC: dt-bindings: max98520: changed signed-off-by email address
by George Song 18 Oct '21

18 Oct '21

changed signed-off-by email address to maximintegrated group. Signed-off-by: George Song <george.song(a)maximintegrated.com> --- Documentation/devicetree/bindings/sound/maxim,max98520.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml index d1f2b5ee775a..ad38f56e1c49 100644 --- a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Maxim Integrated MAX98520 Speaker Amplifier Driver maintainers: - - George Song <George.song(a)analog.com> + - George Song <George.song(a)maximintegrated.com> properties: compatible: -- 2.25.1

2 5

[PATCH 0/4] Getting rid of the reset controller in i2s-tdm
by Nicolas Frattaroli 17 Oct '21

17 Oct '21

Hello, after some discussion with Heiko on IRC, he has admitted to me that the rockchip,cru property, and its corresponding half a reset controller in the driver, is weighing heavily on his mind. The background is that if the lrck only uses one clock for both rx and tx direction, then according to the downstream driver, the rx and tx resets should be asserted at roughly the same time to keep things in sync. Since there is no existing kernel way of doing this, the driver would manually write to the CRU's registers to achieve this, violating abstractions. We've agreed that an atomic bulk reset API would be the best way to achieve what it does in a clean fashion. The details of such an API have yet to be worked out by me, but as it turns out, this is not a pressing need. During my investigation, I noticed that I can simply drop the synchronised reset for now and assert the two resets manually one after the other, and deassert them in the same manner. For the case I care about, which is audio playback, this seems to work just fine. Should someone actually find a case where this causes a problem, it should be fixed with an atomic bulk reset API. Patch 1 removes the direct CRU writing stuff from the i2s-tdm driver. Patch 2 drops the rockchip,cru property from the bindings; they have not yet been in a kernel release, so as far as I know, we can still change them with no regard for backwards compatibility. Patch 3 adds the rk356x i2s1 node without the rockchip,cru property. Patch 4 adds the analog audio output on Quartz64, included here for Heiko's convenience. Regards, Nicolas Frattaroli Nicolas Frattaroli (4): ASoC: rockchip: i2s-tdm: Strip out direct CRU use ASoC: dt-bindings: rockchip: i2s-tdm: Drop rockchip,cru property arm64: dts: rockchip: Add i2s1 on rk356x arm64: dts: rockchip: Add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 16 --- .../boot/dts/rockchip/rk3566-quartz64-a.dts | 31 ++++- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 25 ++++ sound/soc/rockchip/rockchip_i2s_tdm.c | 126 +++--------------- 4 files changed, 76 insertions(+), 122 deletions(-) -- 2.33.1

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[PATCH v2 0/3] ALSA: Support for non-contiguous and non-coherent page allocations
by Takashi Iwai 17 Oct '21

17 Oct '21

Hi, this is a revised patchset that is created from the latest code. The previous one caused a slight conflict with the recent changes, also some patch descriptions have been updated. *** Here is a patch set to add the support for non-contiguous and non-coherent page allocations to the ALSA core memalloc helpers. Currently no driver is using, but this also gave a nice cleanup for the existing SG-buffer helpers on x86. Takashi === Takashi Iwai (3): ALSA: memalloc: Support for non-contiguous page allocation ALSA: memalloc: Support for non-coherent page allocation ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type include/sound/memalloc.h | 61 +++++++++-- sound/core/Makefile | 1 - sound/core/memalloc.c | 207 +++++++++++++++++++++++++++++++++--- sound/core/memalloc_local.h | 1 + sound/core/pcm_lib.c | 3 + sound/core/pcm_local.h | 7 ++ sound/core/pcm_memory.c | 13 ++- sound/core/pcm_native.c | 16 +++ sound/core/sgbuf.c | 201 ---------------------------------- 9 files changed, 282 insertions(+), 228 deletions(-) delete mode 100644 sound/core/sgbuf.c -- 2.26.2

1 3

[PATCH 0/3] ALSA: Support for non-contiguous and non-coherent page allocations
by Takashi Iwai 17 Oct '21

17 Oct '21

Hi, here is a patch set to add the support for non-contiguous and non-coherent page allocations to the ALSA core memalloc helpers. Currently no driver is using, but this also gave a nice cleanup for the existing SG-buffer helpers on x86. Takashi === Takashi Iwai (3): ALSA: memalloc: Support for non-contiguous page allocation ALSA: memalloc: Support for non-coherent page allocation ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type include/sound/memalloc.h | 61 +++++++++-- sound/core/Makefile | 1 - sound/core/memalloc.c | 207 +++++++++++++++++++++++++++++++++--- sound/core/memalloc_local.h | 1 + sound/core/pcm_lib.c | 3 + sound/core/pcm_local.h | 7 ++ sound/core/pcm_memory.c | 13 ++- sound/core/pcm_native.c | 16 +++ sound/core/sgbuf.c | 201 ---------------------------------- 9 files changed, 282 insertions(+), 228 deletions(-) delete mode 100644 sound/core/sgbuf.c -- 2.26.2

1 4

[RFC PATCH] sound: ISA: not for M68K
by Randy Dunlap 17 Oct '21

17 Oct '21

On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) --- linux-next-20211015.orig/sound/isa/Kconfig +++ linux-next-20211015/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. --- linux-next-20211015.orig/sound/pci/Kconfig +++ linux-next-20211015/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. --- linux-next-20211015.orig/sound/core/Makefile +++ linux-next-20211015/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o

2 1

[PATCH] ASoC: dt-bindings: rockchip: i2s-tdm: fix rockchip, cru requirement
by Nicolas Frattaroli 16 Oct '21

16 Oct '21

The behaviour described in the bindings is exactly the opposite of what it should be, so this patch fixes the description and adjusts the if conditions for adding rockchip,cru to required. Fixes: 510f1c133aed ("ASoC: dt-bindings: rockchip: add i2s-tdm bindings") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- .../devicetree/bindings/sound/rockchip,i2s-tdm.yaml | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml index ce3e18b50230..93f5221e9381 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -86,7 +86,7 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of the cru. - Required if neither trcm-sync-tx-only nor trcm-sync-rx-only are specified. + Required if either trcm-sync-tx-only or trcm-sync-rx-only are specified. rockchip,grf: $ref: /schemas/types.yaml#/definitions/phandle @@ -147,8 +147,13 @@ required: allOf: - if: properties: - rockchip,trcm-sync-tx-only: false - rockchip,trcm-sync-rx-only: false + rockchip,trcm-sync-tx-only: true + then: + required: + - rockchip,cru + - if: + properties: + rockchip,trcm-sync-rx-only: true then: required: - rockchip,cru -- 2.33.0

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[PATCH v5 00/16] ASoC: Add Audio Graph Card2 support
by Kuninori Morimoto 15 Oct '21

15 Oct '21

Hi Mark We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi-CPU/Codec support, Codec2Codec support, etc. This patch adds more flexible new Audio-Graph-Card2 driver for it. Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have full compatibility. The reason why I need Audio-Graph-Card2 instead of updating Audio-Graph-Card is that it is very difficult to keep compatibility. Audio-Graph-Card2 supports Normal/DPCM/Codec2Codec Connection wich Single/Multi DAIs. And it is possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and its custom driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it can indicate called function name. We can use it to trace callback order, understanding ALSA SoC behavior, etc, etc... Sample DT settings of Audio Graph Card2 is using Test-Component as CPU/Codec DAI. You can easily try to use/test it if you added below line to your DT file. Your .config needs to have below CONFIGs to use/test it. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE CONFIG_SND_TEST_COMPONENT Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87tuitusy4.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6jn56x0.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - don't use "port" base for_each loop v2 -> v3 - Rename audio-graph-card2 to rich-graph-card - Rename DSP to DPCM not to confuse - Normal/DPCM/Codec2Codec can use Single/Multi DAIs. - use dpcm/multi/codec2codec node instead of using extra compatible - Sample DTSI patch is separated to Single/Multi. v3 -> v4 - Rename rich-graph-card to audio-graph-card2 - fixup custom sample driver's connection bug - test-component compatible uses "verbose" instead of "vv" v4 -> v5 - tidyup git-log comment at - tidyup Custom Sample comment Kuninori Morimoto (16): ASoC: test-component: add Test Component YAML bindings ASoC: test-component: add Test Component for Sound debug/test ASoC: simple-card-utils: add asoc_graph_is_ports0() ASoC: simple-card-utils: add codec2codec support ASoC: add Audio Graph Card2 driver ASoC: audio-graph-card2: add Multi CPU/Codec support ASoC: audio-graph-card2: add DPCM support ASoC: audio-graph-card2: add Codec2Codec support ASoC: add Audio Graph Card2 Yaml Document ASoC: add Audio Graph Card2 Custom Sample ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Nulti) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Multi) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Multi) .../bindings/sound/audio-graph-card2.yaml | 57 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 21 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../generic/audio-graph-card2-custom-sample.c | 183 +++ .../audio-graph-card2-custom-sample.dtsi | 227 +++ sound/soc/generic/audio-graph-card2.c | 1281 +++++++++++++++++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 11 files changed, 2536 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.c create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

2 17

[PATCH V2 1/3] ASoc: amd: create platform device for VG machine driver
by Vijendar Mukunda 15 Oct '21

15 Oct '21

Create platform device for Vangogh machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v1: - changed machine driver platform device name as "acp5x_mach" sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/vangogh/pci-acp5x.c | 3 +++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index a808635f9740..fe5e1fa98974 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -23,7 +23,7 @@ #define ACP_ERR_INTR_MASK 0x20000000 #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF -#define ACP5x_DEVS 3 +#define ACP5x_DEVS 4 #define ACP5x_REG_START 0x1240000 #define ACP5x_REG_END 0x1250200 #define ACP5x_I2STDM_REG_START 0x1242400 diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index a57b762d9f2e..2b6b9edc36e2 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -213,6 +213,9 @@ static int snd_acp5x_probe(struct pci_dev *pci, pdevinfo[2].num_res = 1; pdevinfo[2].res = &adata->res[2]; + pdevinfo[3].name = "acp5x_mach"; + pdevinfo[3].id = 0; + pdevinfo[3].parent = &pci->dev; for (i = 0; i < ACP5x_DEVS; i++) { adata->pdev[i] = platform_device_register_full(&pdevinfo[i]); -- 2.25.1

2 3

[PATCH 00/16] ASoC: cs42l42: Collection of bugfixes
by Richard Fitzgerald 15 Oct '21

15 Oct '21

This patch set contains various bugfixes for the cs42l42 codec driver. Patches marked "Fixes" will apply cleanly to the patch that first introduced the bug. Patches NOT marked "Fixes" will not apply cleanly to the point the bug was first introduced and/or the bug is not having sufficient impact to risk churning older code versions. Richard Fitzgerald (16): ASoC: cs42l42: Don't reconfigure the PLL while it is running ASoC: cs42l42: Always configure both ASP TX channels ASoC: cs42l42: Correct some register default values ASoC: cs42l42: Don't set defaults for volatile registers ASoC: cs42l42: Defer probe if request_threaded_irq() returns EPROBE_DEFER ASoC: cs42l42: Reset GPIO is mandatory ASoC: cs42l42: Correct power-up sequence to match datasheet ASoC: cs42l42: Reset and power-down on driver remove() ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler ASoC: cs42l42: Don't claim to support 192k ASoC: cs42l42: Use PLL for SCLK > 12.288MHz ASoC: cs42l42: Allow time for HP/ADC to power-up after enable ASoC: cs42l42: Set correct SRC MCLK ASoC: cs42l42: Mark OSC_SWITCH_STATUS register volatile ASoC: cs42l42: Fix WARN in remove() if running without an interrupt ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts sound/soc/codecs/cs42l42.c | 282 +++++++++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 9 +- 2 files changed, 160 insertions(+), 131 deletions(-) -- 2.11.0

2 20

[PATCH v4 00/16] ASoC: Add Audio Graph Card2 support
by Kuninori Morimoto 15 Oct '21

15 Oct '21

Hi Mark We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi-CPU/Codec support, Codec2Codec support, etc. This patch adds more flexible new Audio-Graph-Card2 driver for it. Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have full compatibility. The reason why I need Audio-Graph-Card2 instead of updating Audio-Graph-Card is that it is very difficult to keep compatibility. Audio-Graph-Card2 supports Normal/DPCM/Codec2Codec Connection wich Single/Multi DAIs. And it is possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and its custom driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it can indicate called function name. We can use it to trace callback order, understanding ALSA SoC behavior, etc, etc... Sample DT settings of Audio Graph Card2 is using Test-Component as CPU/Codec DAI. You can easily try to use/test it if you added below line to your DT file. Your .config needs to have below CONFIGs to use/test it. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE CONFIG_SND_TEST_COMPONENT Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87tuitusy4.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - don't use "port" base for_each loop v2 -> v3 - Rename audio-graph-card2 to rich-graph-card - Rename DSP to DPCM not to confuse - Normal/DPCM/Codec2Codec can use Single/Multi DAIs. - use dpcm/multi/codec2codec node instead of using extra compatible - Sample DTSI patch is separated to Single/Multi. v3 -> v4 - Rename rich-graph-card to audio-graph-card2 - fixup custom sample driver's connection bug - test-component compatible uses "verbose" instead of "vv" Kuninori Morimoto (16): ASoC: test-component: add Test Component YAML bindings ASoC: test-component: add Test Component for Sound debug/test ASoC: simple-card-utils: add asoc_graph_is_ports0() ASoC: simple-card-utils: add codec2codec support ASoC: add Audio Graph Card2 driver ASoC: audio-graph-card2: add Multi CPU/Codec support ASoC: audio-graph-card2: add DPCM support ASoC: audio-graph-card2: add Codec2Codec support ASoC: add Audio Graph Card2 Yaml Document ASoC: add Audio Graph Card2 Custom Sample ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Nulti) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Multi) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Multi) .../bindings/sound/audio-graph-card2.yaml | 57 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 21 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../generic/audio-graph-card2-custom-sample.c | 174 +++ .../audio-graph-card2-custom-sample.dtsi | 227 +++ sound/soc/generic/audio-graph-card2.c | 1281 +++++++++++++++++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 11 files changed, 2527 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.c create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

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[PATCH] ASoC: soc-component: improve error reporting for register access
by Srinivas Kandagatla 15 Oct '21

15 Oct '21

Currently errors on register read/write/update are reported with an error code and the corresponding function but does not provide any details on the which register number did it actually fail. register number can give better clue and it should be easy to locate the code and fix. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Personally I found this patch very useful while developing and debugging. Ex: below error "ASoC: error at soc_component_read_no_lock on soc@0:codec: -16" did not provide much info except that it failed to read, but after this patch error message looks like: "ASoC: error at soc_component_read_no_lock on soc@0:codec for register: [0x00003125] -16" which was easy to locate and fix. sound/soc/soc-component.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index a08a897c5230..c76ff9c59dfb 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -13,9 +13,10 @@ #include <sound/soc.h> #include <linux/bitops.h> -#define soc_component_ret(dai, ret) _soc_component_ret(dai, __func__, ret) +#define soc_component_ret(dai, ret) _soc_component_ret(dai, __func__, ret, -1) +#define soc_component_ret_reg_rw(dai, ret, reg) _soc_component_ret(dai, __func__, ret, reg) static inline int _soc_component_ret(struct snd_soc_component *component, - const char *func, int ret) + const char *func, int ret, int reg) { /* Positive/Zero values are not errors */ if (ret >= 0) @@ -27,9 +28,14 @@ static inline int _soc_component_ret(struct snd_soc_component *component, case -ENOTSUPP: break; default: - dev_err(component->dev, - "ASoC: error at %s on %s: %d\n", - func, component->name, ret); + if (reg == -1) + dev_err(component->dev, + "ASoC: error at %s on %s: %d\n", + func, component->name, ret); + else + dev_err(component->dev, + "ASoC: error at %s on %s for register: [0x%08x] %d\n", + func, component->name, reg, ret); } return ret; @@ -687,7 +693,7 @@ static unsigned int soc_component_read_no_lock( ret = -EIO; if (ret < 0) - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); return val; } @@ -723,7 +729,7 @@ static int soc_component_write_no_lock( else if (component->driver->write) ret = component->driver->write(component, reg, val); - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); } /** @@ -765,7 +771,7 @@ static int snd_soc_component_update_bits_legacy( mutex_unlock(&component->io_mutex); - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); } /** @@ -793,7 +799,7 @@ int snd_soc_component_update_bits(struct snd_soc_component *component, mask, val, &change); if (ret < 0) - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); return change; } EXPORT_SYMBOL_GPL(snd_soc_component_update_bits); @@ -829,7 +835,7 @@ int snd_soc_component_update_bits_async(struct snd_soc_component *component, mask, val, &change); if (ret < 0) - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); return change; } EXPORT_SYMBOL_GPL(snd_soc_component_update_bits_async); -- 2.21.0

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[RFC PATCH v3 00/13] ASoC : soc-pcm: fix trigger race conditions with shared BE
by Pierre-Louis Bossart 15 Oct '21

15 Oct '21

We've been adding a 'deep buffer' PCM device to several SOF topologies in order to reduce power consumption. The typical use-case would be music playback over a headset: this additional PCM device provides more buffering and longer latencies, leaving the rest of the system sleep for longer periods. Notifications and 'regular' low-latency audio playback would still use the 'normal' PCM device and be mixed with the 'deep buffer' before rendering on the headphone endpoint. The tentative direction would be to expose this alternate device to PulseAudio/PipeWire/CRAS via the UCM SectionModifier definitions. That seemed a straightforward topology change until our automated validation stress tests started reporting issues on SoundWire platforms, when e.g. two START triggers might be send and conversely the STOP trigger is never sent. The SoundWire stream state management flagged inconsistent states when the two 'normal' and 'deep buffer' devices are used concurrently with rapid play/stop/pause monkey testing. Looking at the soc-pcm.c code, it seems that the BE state management needs a lot of love. a) there is no consistent protection for the BE state. In some parts of the code, the state updates are protected by a spinlock but in the trigger they are not. When we open/play/close the two PCM devices in stress tests, we end-up testing a state that is being modified. That can't be good. b) there is a conceptual deadlock: on stop we check the FE states to see if a shared BE can be stopped, but since we trigger the BE first the FE states have not been modified yet, so the TRIGGER_STOP is never sent. This patchset suggests the removal of the dedicated 'dpcm_lock', and the use of the FE PCM lock before walking through the BE list, a mutual exclusion between triggers using the BE PCM lock, and the use of a refcount to decide when to trigger the BE. With these patches I am able to run our entire validation suite without any issues with this new 'deep buffer' topology, and no regressions on existing solutions [1] One might ask 'how come we didn't see this earlier'? The answer is probably that the .trigger callbacks in most implementations seems to perform DAPM operations, and sending the triggers multiple times is not an issue. In the case of SoundWire, we do use the .trigger callback to reconfigure the bus using the 'bank switch' mechanism. It could be acceptable to tolerate a trigger multiple times, but the deadlock on stop cannot be fixed at the SoundWire layer alone. I chose to send this patchset as an RFCv3 to gather more feedback and make use others know about DPCM issues. We're going to spend more time on this but if others can provide feedback/test results it would be greatly appreciated. The change in the locking model could be problematic on other platforms so we do want more time to comment/test before even considering a merge. Opens: 1) is this the right solution? The DPCM code is far from simple, has notions such as SND_SOC_DPCM_UPDATE_NO and 'trigger_pending' that I have no background on. It's not clear if these cases are still needed with the locking changes. 2) There are other reports of kernel oopses [2] that seem related to the lack of protection. I'd be good to confirm if this patchset solve these problems as well. [1] https://github.com/thesofproject/linux/pull/3146 [2] https://lore.kernel.org/alsa-devel/002f01d7b4f5$c030f4a0$4092dde0$@samsung.… changes since RFCv2: Removal of dpcm_lock to use FE PCM locks (credits to Takashi Iwai for the suggestion). The FE PCM lock is now used before each use of for_each_dpcm_be() - with the exception of the trigger where the lock is already taken. This change is also applied in drivers which make use of this loop (compress, SH, FSL). Addition of BE PCM lock to deal with mutual exclusion between triggers for the same BE. Alignment of the BE atomicity on the FE on connections, this is required to avoid sleeping in atomic context. Additional cleanups (indentation, static functions) changes since RFC v1: Removed unused function Removed exported symbols only used in soc-pcm.c, used static instead Use a mutex instead of a spinlock Protect all for_each_dpcm_be() loops Fix bugs introduced in the refcount Pierre-Louis Bossart (13): ASoC: soc-pcm: remove snd_soc_dpcm_fe_can_update() ASoC: soc-pcm: don't export local functions, use static ASoC: soc-pcm: use proper indentation on 'continue' ASoC: soc-pcm: introduce snd_soc_dpcm_fe_lock_irq/unlock_irq() ASoC: soc-pcm: align BE 'atomicity' with that of the FE ASoC: soc-pcm: remove dpcm spin_lock, use PCM stream lock ASoC: soc-pcm: protect for_each_dpcm_be() loops ASoC: soc-compress: protect for_each_dpcm_be() loops ASoC: sh: rcar: protect for_each_dpcm_be() loops ASoC: fsl: asrc_dma: protect for_each_dpcm_be() loops ASoC: soc-pcm: serialize BE triggers ASoC: soc-pcm: test refcount before triggering ASoC: soc-pcm: fix BE handling of PAUSE_RELEASE include/sound/soc-dpcm.h | 20 +-- include/sound/soc.h | 2 - sound/soc/fsl/fsl_asrc_dma.c | 2 + sound/soc/sh/rcar/core.c | 2 + sound/soc/soc-compress.c | 4 + sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 253 ++++++++++++++++++++++++----------- 7 files changed, 187 insertions(+), 97 deletions(-) -- 2.25.1

3 25

[GIT PULL] ASoC fixes for v5.15-rc5
by Mark Brown 15 Oct '21

15 Oct '21

The following changes since commit cfacfefd382af3b42905108b54f02820dca225c4: ASoC: SOF: trace: Omit error print when waking up trace sleepers (2021-09-17 13:16:36 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc5 for you to fetch changes up to 6b9b546dc00797c74bef491668ce5431ff54e1e2: ASoC: wm8960: Fix clock configuration on slave mode (2021-10-13 16:25:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A colletion of smallish mostly driver specific fixes, the biggest thing here is fixing some of the core code to generate change notifications properly when writing to controls which will fix issues with UIs not showing the correct values. There's one build fix here with a slightly misleading changelog saying it's adding IRQ config support, it's adding a missing select of the regmap-irq code rather than adding a feature. ---------------------------------------------------------------- Andy Shevchenko (4): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation Hans de Goede (1): ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working Mark Brown (2): ASoC: pcm179x: Add missing entries SPI to device ID table ASoC: cs4341: Add SPI device ID table Peter Rosin (1): ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers Shengjiu Wang (2): ASoC: fsl_xcvr: Fix channel swap issue with ARC ASoC: wm8960: Fix clock configuration on slave mode Srinivasa Rao Mandadapu (2): ASoC: wcd938x: Fix jack detection issue ASoC: codec: wcd938x: Add irq config support Stefan Binding (1): ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets Takashi Iwai (1): ASoC: DAPM: Fix missing kctl change notifications Yang Yingliang (1): ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked() sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l42.c | 16 +++------------ sound/soc/codecs/cs4341.c | 7 +++++++ sound/soc/codecs/nau8824.c | 4 ++-- sound/soc/codecs/pcm179x-spi.c | 1 + sound/soc/codecs/pcm512x.c | 2 ++ sound/soc/codecs/wcd938x.c | 6 +++--- sound/soc/codecs/wm8960.c | 13 +++++++++--- sound/soc/fsl/fsl_xcvr.c | 17 +++++++++++----- sound/soc/intel/boards/bytcht_es8316.c | 37 +++++++++++----------------------- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 13 +++++++----- 12 files changed, 62 insertions(+), 56 deletions(-)

2 1

[PATCH v2] Fixes HP Spectre x360 15-eb1xxx speakers
by Davide Baldo 15 Oct '21

15 Oct '21

In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and rear speakers are silent, this patch fixes that by overriding the pin layout and by initializing the amplifier which needs a GPIO pin to be set to 1 then 0, similar to the existing HP Spectre x360 14 model. In order to have volume control, both front and rear speakers were forced to use the DAC1. This patch also correctly map the mute LED but since there is no microphone on/off switch exposed by the alsa subsystem it never turns on by itself. There are still known audio issues in this laptop: headset microphone doesn't work, the button to mute/unmute microphone is not yet mapped, the LED of the mute/unmute speakers doesn't seems to be exposed via GPIO and never turns on. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213953 Signed-off-by: Davide Baldo <davide(a)baldo.me> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b9230a274b0..dda979eaeb44 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6414,6 +6414,44 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +/* GPIO1 = amplifier on/off + * GPIO3 = mic mute LED + */ +static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* front/high speakers */ + { 0x17, 0x90170130 }, /* back/bass speakers */ + { } + }; + + //enable micmute led + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->micmute_led_polarity = 1; + /* needed for amp of back speakers */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + snd_hda_apply_pincfgs(codec, pincfgs); + /* share DAC to have unified volume control */ + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp of back speakers */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6548,6 +6586,7 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -8240,6 +8279,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360, }, + [ALC285_FIXUP_HP_SPECTRE_X360_EB1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_eb1 + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -8567,6 +8610,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), @@ -8987,6 +9032,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, -- 2.32.0

2 1

[PATCH] sound/usb: Provide quirk for Sennheiser GSP670 Headset
by Brendan Grieve 15 Oct '21

15 Oct '21

As per discussion at: https://github.com/szszoke/sennheiser-gsp670-pulseaudio-profile/issues/13 The GSP670 has 2 playback and 1 recording device that by default are detected in an incompatible order for alsa. This may have been done to make it compatible for the console by the manufacturer and only affects the latest firmware which uses its own ID. This quirk will resolve this by reordering the channels. Signed-off-by: Brendan Grieve <brendan(a)grieve.com.au> --- sound/usb/quirks-table.h | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index de18fff69280..2af8c68fac27 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -4012,6 +4012,38 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Sennheiser GSP670 + * Change order of interfaces loaded + */ + USB_DEVICE(0x1395, 0x0300), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + // Communication + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Recording + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Main + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.25.1

2 1

[PATCH] ASoC: rsnd: Add of_node_put() before return
by Wan Jiabing 15 Oct '21

15 Oct '21

Fix following coccicheck warning: ./sound/soc/sh/rcar/core.c:1223:1-23: WARNING: Function for_each_child_of_node should have of_node_put() before return Early exits from for_each_child_of_node should decrement the node reference counter. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 978bd0406729..6a8fe0da7670 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1225,6 +1225,7 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name if (i < 0) { dev_err(dev, "strange node numbering (%s)", of_node_full_name(node)); + of_node_put(np); return 0; } i++; -- 2.20.1

1 0

[PATCH] ALSA: pcm: Unify snd_pcm_delay() and snd_pcm_hwsync()
by Takashi Iwai 14 Oct '21

14 Oct '21

Both snd_pcm_delay() and snd_pcm_hwsync() do the almost same thing. The only difference is that the former calculate the delay, so unify them as a code cleanup, and treat NULL delay argument only for hwsync operation. Also, the patch does a slight code refactoring in snd_pcm_delay(). The initialization of the delay value is done in the caller side now. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 46c643db18eb..627b201b6084 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2932,32 +2932,24 @@ static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream, return ret; } -static int snd_pcm_hwsync(struct snd_pcm_substream *substream) -{ - int err; - - snd_pcm_stream_lock_irq(substream); - err = do_pcm_hwsync(substream); - snd_pcm_stream_unlock_irq(substream); - return err; -} - static int snd_pcm_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *delay) { int err; - snd_pcm_sframes_t n = 0; snd_pcm_stream_lock_irq(substream); err = do_pcm_hwsync(substream); - if (!err) - n = snd_pcm_calc_delay(substream); + if (delay && !err) + *delay = snd_pcm_calc_delay(substream); snd_pcm_stream_unlock_irq(substream); - if (!err) - *delay = n; return err; } +static inline int snd_pcm_hwsync(struct snd_pcm_substream *substream) +{ + return snd_pcm_delay(substream, NULL); +} + static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, struct snd_pcm_sync_ptr __user *_sync_ptr) { @@ -3275,7 +3267,7 @@ static int snd_pcm_common_ioctl(struct file *file, return snd_pcm_hwsync(substream); case SNDRV_PCM_IOCTL_DELAY: { - snd_pcm_sframes_t delay; + snd_pcm_sframes_t delay = 0; snd_pcm_sframes_t __user *res = arg; int err; -- 2.31.1

2 3

[PATCH 0/3] ALSA: usb-audio: Small improvements in mixer handling
by Takashi Iwai 14 Oct '21

14 Oct '21

Hi, this is a patchset with small fixes for improving the USB-audio mixer behavior. Instead of treating the error from a control message continuously, initialize all at the probe and keep the rest working. The first two patches are rather fixes for error messages in get_ctl_value_v2() function, the last one implements the new behavior. Takashi === Takashi Iwai (3): ALSA: usb-audio: Downgrade error message in get_ctl_value_v2() ALSA: usb-audio: Drop superfluous error message after disconnection ALSA: usb-audio: Initialize every feature unit once at probe time sound/usb/mixer.c | 42 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 36 insertions(+), 6 deletions(-) -- 2.26.2

2 6

[PATCH 2/2] ASoC: max98520: Add max98520 audio amplifier driver
by George Song 14 Oct '21

14 Oct '21

Signed-off-by: George Song <george.song(a)analog.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98520.c | 777 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98520.h | 159 ++++++++ 4 files changed, 943 insertions(+) create mode 100644 sound/soc/codecs/max98520.c create mode 100644 sound/soc/codecs/max98520.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..86b121d2f381 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -115,6 +115,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MAX98357A imply SND_SOC_MAX98371 imply SND_SOC_MAX98504 + imply SND_SOC_MAX98520 imply SND_SOC_MAX9867 imply SND_SOC_MAX98925 imply SND_SOC_MAX98926 @@ -922,6 +923,10 @@ config SND_SOC_MAX98927 tristate "Maxim Integrated MAX98927 Speaker Amplifier" depends on I2C +config SND_SOC_MAX98520 + tristate "Maxim Integrated MAX98520 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX98373 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..daf63e31fdd0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -123,6 +123,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max98927-objs := max98927.o +snd-soc-max98520-objs := max98520.o snd-soc-max98373-objs := max98373.o snd-soc-max98373-i2c-objs := max98373-i2c.o snd-soc-max98373-sdw-objs := max98373-sdw.o @@ -450,6 +451,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o +obj-$(CONFIG_SND_SOC_MAX98520) += snd-soc-max98520.o obj-$(CONFIG_SND_SOC_MAX98373) += snd-soc-max98373.o obj-$(CONFIG_SND_SOC_MAX98373_I2C) += snd-soc-max98373-i2c.o obj-$(CONFIG_SND_SOC_MAX98373_SDW) += snd-soc-max98373-sdw.o diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c new file mode 100644 index 000000000000..1a5e0ceae362 --- /dev/null +++ b/sound/soc/codecs/max98520.c @@ -0,0 +1,777 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2021, Maxim Integrated + +#include <linux/acpi.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98520.h" + +static struct reg_default max98520_reg[] = { + {MAX98520_R2000_SW_RESET, 0x00}, + {MAX98520_R2001_STATUS_1, 0x00}, + {MAX98520_R2002_STATUS_2, 0x00}, + {MAX98520_R2020_THERM_WARN_THRESH, 0x46}, + {MAX98520_R2021_THERM_SHDN_THRESH, 0x64}, + {MAX98520_R2022_THERM_HYSTERESIS, 0x02}, + {MAX98520_R2023_THERM_FOLDBACK_SET, 0x31}, + {MAX98520_R2027_THERM_FOLDBACK_EN, 0x01}, + {MAX98520_R2030_CLK_MON_CTRL, 0x00}, + {MAX98520_R2037_ERR_MON_CTRL, 0x01}, + {MAX98520_R2040_PCM_MODE_CFG, 0xC0}, + {MAX98520_R2041_PCM_CLK_SETUP, 0x04}, + {MAX98520_R2042_PCM_SR_SETUP, 0x08}, + {MAX98520_R2043_PCM_RX_SRC1, 0x00}, + {MAX98520_R2044_PCM_RX_SRC2, 0x00}, + {MAX98520_R204F_PCM_RX_EN, 0x00}, + {MAX98520_R2090_AMP_VOL_CTRL, 0x00}, + {MAX98520_R2091_AMP_PATH_GAIN, 0x03}, + {MAX98520_R2092_AMP_DSP_CFG, 0x02}, + {MAX98520_R2094_SSM_CFG, 0x01}, + {MAX98520_R2095_AMP_CFG, 0xF0}, + {MAX98520_R209F_AMP_EN, 0x00}, + {MAX98520_R20B0_ADC_SR, 0x00}, + {MAX98520_R20B1_ADC_RESOLUTION, 0x00}, + {MAX98520_R20B2_ADC_PVDD0_CFG, 0x02}, + {MAX98520_R20B3_ADC_THERMAL_CFG, 0x02}, + {MAX98520_R20B4_ADC_READBACK_CTRL, 0x00}, + {MAX98520_R20B5_ADC_READBACK_UPDATE, 0x00}, + {MAX98520_R20B6_ADC_PVDD_READBACK_MSB, 0x00}, + {MAX98520_R20B7_ADC_PVDD_READBACK_LSB, 0x00}, + {MAX98520_R20B8_ADC_TEMP_READBACK_MSB, 0x00}, + {MAX98520_R20B9_ADC_TEMP_READBACK_LSB, 0x00}, + {MAX98520_R20BA_ADC_LOW_PVDD_READBACK_MSB, 0xFF}, + {MAX98520_R20BB_ADC_LOW_READBACK_LSB, 0x01}, + {MAX98520_R20BC_ADC_HIGH_TEMP_READBACK_MSB, 0x00}, + {MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB, 0x00}, + {MAX98520_R20CF_MEAS_ADC_CFG, 0x00}, + {MAX98520_R20D0_DHT_CFG1, 0x00}, + {MAX98520_R20D1_LIMITER_CFG1, 0x08}, + {MAX98520_R20D2_LIMITER_CFG2, 0x00}, + {MAX98520_R20D3_DHT_CFG2, 0x14}, + {MAX98520_R20D4_DHT_CFG3, 0x02}, + {MAX98520_R20D5_DHT_CFG4, 0x04}, + {MAX98520_R20D6_DHT_HYSTERESIS_CFG, 0x07}, + {MAX98520_R20D8_DHT_EN, 0x00}, + {MAX98520_R210E_AUTO_RESTART_BEHAVIOR, 0x00}, + {MAX98520_R210F_GLOBAL_EN, 0x00}, + {MAX98520_R21FF_REVISION_ID, 0x00}, +}; + +static int max98520_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + unsigned int format = 0; + unsigned int invert = 0; + + dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE; + break; + default: + dev_err(component->dev, "DAI invert mode unsupported\n"); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE, + invert); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format = MAX98520_PCM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + format = MAX98520_PCM_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98520_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98520_PCM_FORMAT_TDM_MODE0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_FORMAT_MASK, + format << MAX98520_PCM_MODE_CFG_FORMAT_SHIFT); + + return 0; +} + +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, 320, +}; + +static int max98520_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} + +static int max98520_set_clock(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98520->ch_size; + int value; + + if (!max98520->tdm_mode) { + /* BCLK configuration */ + value = max98520_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_CLK_SETUP_BSEL_MASK, + value); + } + dev_dbg(component->dev, "%s tdm_mode:%d out\n", __func__, max98520->tdm_mode); + return 0; +} + +static int max98520_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + unsigned int sampling_rate = 0; + unsigned int chan_sz = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + params_format(params)); + goto err; + } + + max98520->ch_size = snd_pcm_format_width(params_format(params)); + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(component->dev, "format supported %d", + params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98520_PCM_SR_8000; + break; + case 11025: + sampling_rate = MAX98520_PCM_SR_11025; + break; + case 12000: + sampling_rate = MAX98520_PCM_SR_12000; + break; + case 16000: + sampling_rate = MAX98520_PCM_SR_16000; + break; + case 22050: + sampling_rate = MAX98520_PCM_SR_22050; + break; + case 24000: + sampling_rate = MAX98520_PCM_SR_24000; + break; + case 32000: + sampling_rate = MAX98520_PCM_SR_32000; + break; + case 44100: + sampling_rate = MAX98520_PCM_SR_44100; + break; + case 48000: + sampling_rate = MAX98520_PCM_SR_48000; + break; + case 88200: + sampling_rate = MAX98520_PCM_SR_88200; + break; + case 96000: + sampling_rate = MAX98520_PCM_SR_96000; + break; + case 176400: + sampling_rate = MAX98520_PCM_SR_176400; + break; + case 192000: + sampling_rate = MAX98520_PCM_SR_192000; + break; + default: + dev_err(component->dev, "rate %d not supported\n", + params_rate(params)); + goto err; + } + + dev_dbg(component->dev, " %s ch_size: %d, sampling rate : %d out\n", __func__, + snd_pcm_format_width(params_format(params)), params_rate(params)); + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98520->regmap, + MAX98520_R2042_PCM_SR_SETUP, + MAX98520_PCM_SR_MASK, + sampling_rate); + + return max98520_set_clock(component, params); +err: + dev_dbg(component->dev, "%s out error", __func__); + return -EINVAL; +} + +static int max98520_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + int bsel = 0; + unsigned int chan_sz = 0; + + if (!tx_mask && !rx_mask && !slots && !slot_width) + max98520->tdm_mode = false; + else + max98520->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98520_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(component->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2041_PCM_CLK_SETUP, + MAX98520_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98520_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98520->regmap, + MAX98520_R2040_PCM_MODE_CFG, + MAX98520_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_update_bits(max98520->regmap, + MAX98520_R2044_PCM_RX_SRC2, + MAX98520_PCM_DMIX_CH0_SRC_MASK, + rx_mask); + regmap_update_bits(max98520->regmap, + MAX98520_R2044_PCM_RX_SRC2, + MAX98520_PCM_DMIX_CH1_SRC_MASK, + rx_mask << MAX98520_PCM_DMIX_CH1_SHIFT); + + return 0; +} + +#define MAX98520_RATES SNDRV_PCM_RATE_8000_192000 + +#define MAX98520_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops max98520_dai_ops = { + .set_fmt = max98520_dai_set_fmt, + .hw_params = max98520_dai_hw_params, + .set_tdm_slot = max98520_dai_tdm_slot, +}; + +static int max98520_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_dbg(component->dev, " AMP ON\n"); + + regmap_write(max98520->regmap, MAX98520_R210F_GLOBAL_EN, 1); + usleep_range(30000, 31000); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, " AMP OFF\n"); + + regmap_write(max98520->regmap, MAX98520_R210F_GLOBAL_EN, 0); + usleep_range(30000, 31000); + max98520->tdm_mode = false; + break; + default: + return 0; + } + return 0; +} + +static const char * const max98520_switch_text[] = { + "Left", "Right", "LeftRight"}; + +static const struct soc_enum dai_sel_enum = + SOC_ENUM_SINGLE(MAX98520_R2043_PCM_RX_SRC1, + 0, 3, max98520_switch_text); + +static const struct snd_kcontrol_new max98520_dai_controls = + SOC_DAPM_ENUM("DAI Sel", dai_sel_enum); + +static const struct snd_soc_dapm_widget max98520_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", + MAX98520_R209F_AMP_EN, 0, 0, max98520_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98520_dai_controls), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static DECLARE_TLV_DB_SCALE(max98520_digital_tlv, -6300, 50, 1); +static const DECLARE_TLV_DB_RANGE(max98520_spk_tlv, + 0, 5, TLV_DB_SCALE_ITEM(600, 300, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_lim_thresh_tlv, + 0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_hysteresis_tlv, + 0, 3, TLV_DB_SCALE_ITEM(100, 100, 0), + 4, 7, TLV_DB_SCALE_ITEM(600, 200, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_rotation_point_tlv, + 0, 1, TLV_DB_SCALE_ITEM(-1500, 300, 0), + 2, 4, TLV_DB_SCALE_ITEM(-1000, 200, 0), + 5, 10, TLV_DB_SCALE_ITEM(-500, 100, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_supply_hr_tlv, + 0, 16, TLV_DB_SCALE_ITEM(-2000, 250, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98520_dht_max_atten_tlv, + 1, 20, TLV_DB_SCALE_ITEM(-2000, 100, 0), +); + +static const char * const max98520_dht_attack_rate_text[] = { + "20us", "40us", "80us", "160us", "320us", "640us", + "1.28ms", "2.56ms", "5.12ms", "10.24ms", "20.48ms", "40.96ms", + "81.92ms", "163.84ms" +}; + +static SOC_ENUM_SINGLE_DECL(max98520_dht_attack_rate_enum, + MAX98520_R20D4_DHT_CFG3, 0, + max98520_dht_attack_rate_text); + +static const char * const max98520_dht_release_rate_text[] = { + "2ms", "4ms", "8ms", "16ms", "32ms", "64ms", "128ms", "256ms", "512ms", + "1.024s", "2.048s", "4.096s", "8.192s", "16.384s" +}; + +static SOC_ENUM_SINGLE_DECL(max98520_dht_release_rate_enum, + MAX98520_R20D5_DHT_CFG4, 0, + max98520_dht_release_rate_text); + +static bool max98520_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98520_R2000_SW_RESET: + case MAX98520_R2027_THERM_FOLDBACK_EN: + case MAX98520_R2030_CLK_MON_CTRL: + case MAX98520_R2037_ERR_MON_CTRL: + case MAX98520_R204F_PCM_RX_EN: + case MAX98520_R209F_AMP_EN: + case MAX98520_R20CF_MEAS_ADC_CFG: + case MAX98520_R20D8_DHT_EN: + case MAX98520_R21FF_REVISION_ID: + case MAX98520_R2001_STATUS_1... MAX98520_R2002_STATUS_2: + case MAX98520_R2020_THERM_WARN_THRESH... MAX98520_R2023_THERM_FOLDBACK_SET: + case MAX98520_R2040_PCM_MODE_CFG... MAX98520_R2044_PCM_RX_SRC2: + case MAX98520_R2090_AMP_VOL_CTRL... MAX98520_R2092_AMP_DSP_CFG: + case MAX98520_R2094_SSM_CFG... MAX98520_R2095_AMP_CFG: + case MAX98520_R20B0_ADC_SR... MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB: + case MAX98520_R20D0_DHT_CFG1... MAX98520_R20D6_DHT_HYSTERESIS_CFG: + case MAX98520_R210E_AUTO_RESTART_BEHAVIOR... MAX98520_R210F_GLOBAL_EN: + case MAX98520_R2161_BOOST_TM1... MAX98520_R2163_BOOST_TM3: + return true; + default: + return false; + } +}; + +static bool max98520_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98520_R210F_GLOBAL_EN: + case MAX98520_R21FF_REVISION_ID: + case MAX98520_R2000_SW_RESET: + case MAX98520_R2001_STATUS_1 ... MAX98520_R2002_STATUS_2: + case MAX98520_R20B4_ADC_READBACK_CTRL + ... MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB: + return true; + default: + return false; + } +} + +static const struct snd_kcontrol_new max98520_snd_controls[] = { +/* Volume */ +SOC_SINGLE_TLV("Digital Volume", MAX98520_R2090_AMP_VOL_CTRL, + 0, 0x7F, 1, max98520_digital_tlv), +SOC_SINGLE_TLV("Speaker Volume", MAX98520_R2091_AMP_PATH_GAIN, + 0, 0x5, 0, max98520_spk_tlv), +/* Volume Ramp Up/Down Enable*/ +SOC_SINGLE("Ramp Up Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_VOL_RMPUP_SHIFT, 1, 0), +SOC_SINGLE("Ramp Down Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_VOL_RMPDN_SHIFT, 1, 0), +/* Clock Monitor Enable */ +SOC_SINGLE("CLK Monitor Switch", MAX98520_R2037_ERR_MON_CTRL, + MAX98520_CTRL_CMON_EN_SHIFT, 1, 0), +/* Clock Monitor Config */ +SOC_SINGLE("CLKMON Autorestart Switch", MAX98520_R2030_CLK_MON_CTRL, + MAX98520_CMON_AUTORESTART_SHIFT, 1, 0), +/* Dither Enable */ +SOC_SINGLE("Dither Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_DITH_EN_SHIFT, 1, 0), +/* DC Blocker Enable */ +SOC_SINGLE("DC Blocker Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_DCBLK_EN_SHIFT, 1, 0), +/* Speaker Safe Mode Enable */ +SOC_SINGLE("Speaker Safemode Switch", MAX98520_R2092_AMP_DSP_CFG, + MAX98520_DSP_SPK_SAFE_EN_SHIFT, 1, 0), +/* AMP SSM Enable */ +SOC_SINGLE("CP Bypass Switch", MAX98520_R2094_SSM_CFG, + MAX98520_SSM_RCVR_MODE_SHIFT, 1, 0), +/* AMP Dynamic Mode Configuration */ +SOC_SINGLE("Dynamic Mode Switch", MAX98520_R2095_AMP_CFG, + MAX98520_CFG_DYN_MODE_SHIFT, 1, 1), +/* AMP Speaker Mode Switch */ +SOC_SINGLE("Speaker Mode Switch", MAX98520_R2095_AMP_CFG, + MAX98520_CFG_SPK_MODE_SHIFT, 1, 0), +/* Dynamic Headroom Tracking */ +SOC_SINGLE("DHT Switch", MAX98520_R20D8_DHT_EN, 0, 1, 0), +SOC_SINGLE("DHT Limiter Mode", MAX98520_R20D2_LIMITER_CFG2, + MAX98520_DHT_LIMITER_MODE_SHIFT, 1, 0), +SOC_SINGLE("DHT Hysteresis Switch", MAX98520_R20D6_DHT_HYSTERESIS_CFG, + MAX98520_DHT_HYSTERESIS_SWITCH_SHIFT, 1, 0), +SOC_SINGLE_TLV("DHT Rot Pnt", MAX98520_R20D0_DHT_CFG1, + MAX98520_DHT_VROT_PNT_SHIFT, 10, 1, max98520_dht_rotation_point_tlv), +SOC_SINGLE_TLV("DHT Supply Headroom", MAX98520_R20D1_LIMITER_CFG1, + MAX98520_DHT_SUPPLY_HR_SHIFT, 16, 0, max98520_dht_supply_hr_tlv), +SOC_SINGLE_TLV("DHT Limiter Threshold", MAX98520_R20D2_LIMITER_CFG2, + MAX98520_DHT_LIMITER_THRESHOLD_SHIFT, 0xF, 1, max98520_dht_lim_thresh_tlv), +SOC_SINGLE_TLV("DHT Max Attenuation", MAX98520_R20D3_DHT_CFG2, + MAX98520_DHT_MAX_ATTEN_SHIFT, 20, 1, max98520_dht_max_atten_tlv), +SOC_SINGLE_TLV("DHT Hysteresis", MAX98520_R20D6_DHT_HYSTERESIS_CFG, + MAX98520_DHT_HYSTERESIS_SHIFT, 0x7, 0, max98520_dht_hysteresis_tlv), +SOC_ENUM("DHT Attack Rate", max98520_dht_attack_rate_enum), +SOC_ENUM("DHT Release Rate", max98520_dht_release_rate_enum), +/* ADC configuration */ +SOC_SINGLE("ADC PVDD CH Switch", MAX98520_R20CF_MEAS_ADC_CFG, 0, 1, 0), +SOC_SINGLE("ADC PVDD FLT Switch", MAX98520_R20B2_ADC_PVDD0_CFG, + MAX98520_FLT_EN_SHIFT, 1, 0), +SOC_SINGLE("ADC TEMP FLT Switch", MAX98520_R20B3_ADC_THERMAL_CFG, + MAX98520_FLT_EN_SHIFT, 1, 0), +SOC_SINGLE("ADC PVDD MSB", MAX98520_R20B6_ADC_PVDD_READBACK_MSB, 0, 0xFF, 0), +SOC_SINGLE("ADC PVDD LSB", MAX98520_R20B7_ADC_PVDD_READBACK_LSB, 0, 0x01, 0), +SOC_SINGLE("ADC TEMP MSB", MAX98520_R20B8_ADC_TEMP_READBACK_MSB, 0, 0xFF, 0), +SOC_SINGLE("ADC TEMP LSB", MAX98520_R20B9_ADC_TEMP_READBACK_LSB, 0, 0x01, 0), +}; + +static const struct snd_soc_dapm_route max98520_audio_map[] = { + /* Plabyack */ + {"DAI Sel Mux", "Left", "Amp Enable"}, + {"DAI Sel Mux", "Right", "Amp Enable"}, + {"DAI Sel Mux", "LeftRight", "Amp Enable"}, + {"BE_OUT", NULL, "DAI Sel Mux"}, +}; + +static struct snd_soc_dai_driver max98520_dai[] = { + { + .name = "max98520-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98520_RATES, + .formats = MAX98520_FORMATS, + }, + .ops = &max98520_dai_ops, + } + +}; + +static void max98520_reset(struct max98520_priv *max98520, struct device *dev) +{ + int ret, reg, count; + + /* Software Reset */ + ret = regmap_write(max98520->regmap, MAX98520_R2000_SW_RESET, 1); + if (ret) + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + + count = 0; + while (count < 3) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98520->regmap, MAX98520_R21FF_REVISION_ID, &reg); + if (!ret) { + dev_info(dev, "Reset completed (retry:%d)\n", count); + return; + } + count++; + } + dev_err(dev, "Reset failed. (ret:%d)\n", ret); +} + +static int max98520_probe(struct snd_soc_component *component) +{ + struct max98520_priv *max98520 = + snd_soc_component_get_drvdata(component); + + /* Software Reset */ + max98520_reset(max98520, component->dev); + usleep_range(30000, 31000); + + /* L/R mix configuration */ + regmap_write(max98520->regmap, MAX98520_R2043_PCM_RX_SRC1, 0x2); + + regmap_write(max98520->regmap, MAX98520_R2044_PCM_RX_SRC2, 0x10); + /* Enable DC blocker */ + regmap_update_bits(max98520->regmap, MAX98520_R2092_AMP_DSP_CFG, 1, 1); + /* Disable Speaker Safe Mode */ + regmap_update_bits(max98520->regmap, + MAX98520_R2092_AMP_DSP_CFG, MAX98520_SPK_SAFE_EN_MASK, 0); + /* Enable Clock Monitor Auto-restart */ + regmap_write(max98520->regmap, MAX98520_R2030_CLK_MON_CTRL, 0x1); + + /* Hard coded values for the experiments */ + regmap_write(max98520->regmap, MAX98520_R21FF_REVISION_ID, 0x54); + regmap_write(max98520->regmap, MAX98520_R21FF_REVISION_ID, 0x4d); + regmap_write(max98520->regmap, MAX98520_R2161_BOOST_TM1, 0x2); + regmap_write(max98520->regmap, MAX98520_R2095_AMP_CFG, 0xc8); + + /* set Rx Enable */ + regmap_update_bits(max98520->regmap, + MAX98520_R204F_PCM_RX_EN, + MAX98520_PCM_RX_EN_MASK, 1); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int max98520_suspend(struct device *dev) +{ + struct max98520_priv *max98520 = dev_get_drvdata(dev); + + regcache_cache_only(max98520->regmap, true); + regcache_mark_dirty(max98520->regmap); + return 0; +} +static int max98520_resume(struct device *dev) +{ + struct max98520_priv *max98520 = dev_get_drvdata(dev); + + regcache_cache_only(max98520->regmap, false); + max98520_reset(max98520, dev); + regcache_sync(max98520->regmap); + return 0; +} +#endif + +static const struct dev_pm_ops max98520_pm = { + SET_SYSTEM_SLEEP_PM_OPS(max98520_suspend, max98520_resume) +}; + +static const struct snd_soc_component_driver soc_codec_dev_max98520 = { + .probe = max98520_probe, + .controls = max98520_snd_controls, + .num_controls = ARRAY_SIZE(max98520_snd_controls), + .dapm_widgets = max98520_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98520_dapm_widgets), + .dapm_routes = max98520_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98520_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config max98520_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98520_R21FF_REVISION_ID, + .reg_defaults = max98520_reg, + .num_reg_defaults = ARRAY_SIZE(max98520_reg), + .readable_reg = max98520_readable_register, + .volatile_reg = max98520_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98520_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + + int ret = 0; + int reg = 0; + struct max98520_priv *max98520 = NULL; + struct i2c_adapter *adapter = to_i2c_adapter(i2c->dev.parent); + + ret = i2c_check_functionality(adapter, + I2C_FUNC_SMBUS_BYTE + | I2C_FUNC_SMBUS_BYTE_DATA); + if (!ret) { + dev_err(&i2c->dev, "I2C check functionality failed\n"); + return -ENXIO; + } + + max98520 = devm_kzalloc(&i2c->dev, sizeof(*max98520), GFP_KERNEL); + + if (!max98520) { + ret = -ENOMEM; + return ret; + } + i2c_set_clientdata(i2c, max98520); + + /* regmap initialization */ + max98520->regmap + = devm_regmap_init_i2c(i2c, &max98520_regmap); + if (IS_ERR(max98520->regmap)) { + ret = PTR_ERR(max98520->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + /* Power on device */ + if (gpio_is_valid(max98520->reset_gpio)) { + ret = devm_gpio_request(&i2c->dev, max98520->reset_gpio, + "MAX98520_RESET"); + if (ret) { + dev_err(&i2c->dev, "%s: Failed to request gpio %d\n", + __func__, max98520->reset_gpio); + return -EINVAL; + } + gpio_direction_output(max98520->reset_gpio, 0); + msleep(50); + gpio_direction_output(max98520->reset_gpio, 1); + msleep(20); + } + + /* Check Revision ID */ + ret = regmap_read(max98520->regmap, + MAX98520_R21FF_REVISION_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, + "Failed to read: 0x%02X\n", MAX98520_R21FF_REVISION_ID); + return ret; + } + dev_info(&i2c->dev, "MAX98520 revisionID: 0x%02X\n", reg); + + /* codec registration */ + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_max98520, + max98520_dai, ARRAY_SIZE(max98520_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static const struct i2c_device_id max98520_i2c_id[] = { + { "max98520", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98520_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id max98520_of_match[] = { + { .compatible = "maxim,max98520", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98520_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98520_acpi_match[] = { + { "MX98520", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98520_acpi_match); +#endif + +static struct i2c_driver max98520_i2c_driver = { + .driver = { + .name = "max98520", + .of_match_table = of_match_ptr(max98520_of_match), + .acpi_match_table = ACPI_PTR(max98520_acpi_match), + .pm = &max98520_pm, + }, + .probe = max98520_i2c_probe, + .id_table = max98520_i2c_id, +}; + +module_i2c_driver(max98520_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98520 driver"); +MODULE_AUTHOR("Ryan Lee <ryans.lee(a)analog.com>"); +MODULE_AUTHOR("George Song <george.song(a)analog.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/max98520.h b/sound/soc/codecs/max98520.h new file mode 100644 index 000000000000..afb5f3917136 --- /dev/null +++ b/sound/soc/codecs/max98520.h @@ -0,0 +1,159 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2021, Maxim Integrated. + */ + +#ifndef _MAX98520_H +#define _MAX98520_H + +#define MAX98520_R2000_SW_RESET 0x2000 +#define MAX98520_R2001_STATUS_1 0x2001 +#define MAX98520_R2002_STATUS_2 0x2002 +#define MAX98520_R2020_THERM_WARN_THRESH 0x2020 +#define MAX98520_R2021_THERM_SHDN_THRESH 0x2021 +#define MAX98520_R2022_THERM_HYSTERESIS 0x2022 +#define MAX98520_R2023_THERM_FOLDBACK_SET 0x2023 +#define MAX98520_R2027_THERM_FOLDBACK_EN 0x2027 +#define MAX98520_R2030_CLK_MON_CTRL 0x2030 +#define MAX98520_R2037_ERR_MON_CTRL 0x2037 +#define MAX98520_R2040_PCM_MODE_CFG 0x2040 +#define MAX98520_R2041_PCM_CLK_SETUP 0x2041 +#define MAX98520_R2042_PCM_SR_SETUP 0x2042 +#define MAX98520_R2043_PCM_RX_SRC1 0x2043 +#define MAX98520_R2044_PCM_RX_SRC2 0x2044 +#define MAX98520_R204F_PCM_RX_EN 0x204F +#define MAX98520_R2090_AMP_VOL_CTRL 0x2090 +#define MAX98520_R2091_AMP_PATH_GAIN 0x2091 +#define MAX98520_R2092_AMP_DSP_CFG 0x2092 +#define MAX98520_R2094_SSM_CFG 0x2094 +#define MAX98520_R2095_AMP_CFG 0x2095 +#define MAX98520_R209F_AMP_EN 0x209F +#define MAX98520_R20B0_ADC_SR 0x20B0 +#define MAX98520_R20B1_ADC_RESOLUTION 0x20B1 +#define MAX98520_R20B2_ADC_PVDD0_CFG 0x20B2 +#define MAX98520_R20B3_ADC_THERMAL_CFG 0x20B3 +#define MAX98520_R20B4_ADC_READBACK_CTRL 0x20B4 +#define MAX98520_R20B5_ADC_READBACK_UPDATE 0x20B5 +#define MAX98520_R20B6_ADC_PVDD_READBACK_MSB 0x20B6 +#define MAX98520_R20B7_ADC_PVDD_READBACK_LSB 0x20B7 +#define MAX98520_R20B8_ADC_TEMP_READBACK_MSB 0x20B8 +#define MAX98520_R20B9_ADC_TEMP_READBACK_LSB 0x20B9 +#define MAX98520_R20BA_ADC_LOW_PVDD_READBACK_MSB 0x20BA +#define MAX98520_R20BB_ADC_LOW_READBACK_LSB 0x20BB +#define MAX98520_R20BC_ADC_HIGH_TEMP_READBACK_MSB 0x20BC +#define MAX98520_R20BD_ADC_HIGH_TEMP_READBACK_LSB 0x20BD +#define MAX98520_R20CF_MEAS_ADC_CFG 0x20CF +#define MAX98520_R20D0_DHT_CFG1 0x20D0 +#define MAX98520_R20D1_LIMITER_CFG1 0x20D1 +#define MAX98520_R20D2_LIMITER_CFG2 0x20D2 +#define MAX98520_R20D3_DHT_CFG2 0x20D3 +#define MAX98520_R20D4_DHT_CFG3 0x20D4 +#define MAX98520_R20D5_DHT_CFG4 0x20D5 +#define MAX98520_R20D6_DHT_HYSTERESIS_CFG 0x20D6 +#define MAX98520_R20D8_DHT_EN 0x20D8 +#define MAX98520_R210E_AUTO_RESTART_BEHAVIOR 0x210E +#define MAX98520_R210F_GLOBAL_EN 0x210F +#define MAX98520_R2161_BOOST_TM1 0x2161 +#define MAX98520_R2162_BOOST_TM2 0x2162 +#define MAX98520_R2163_BOOST_TM3 0x2163 +#define MAX98520_R21FF_REVISION_ID 0x21FF + +/* MAX98520_R2030_CLK_MON_CTRL */ +#define MAX98520_CMON_AUTORESTART_SHIFT (0) + +/* MAX98520_R2037_ERR_MON_CTRL */ +#define MAX98520_CTRL_CMON_EN_SHIFT (0) + +/* MAX98520_R2040_PCM_MODE_CFG */ +#define MAX98520_PCM_MODE_CFG_FORMAT_MASK (0x7 << 3) +#define MAX98520_PCM_MODE_CFG_FORMAT_SHIFT (3) +#define MAX98520_PCM_TX_CH_INTERLEAVE_MASK (0x1 << 2) +#define MAX98520_PCM_FORMAT_I2S (0x0 << 3) +#define MAX98520_PCM_FORMAT_LJ (0x1 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE0 (0x3 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE1 (0x4 << 3) +#define MAX98520_PCM_FORMAT_TDM_MODE2 (0x5 << 3) +#define MAX98520_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) +#define MAX98520_PCM_MODE_CFG_CHANSZ_32 (0x3 << 6) + +/* MAX98520_R2041_PCM_CLK_SETUP */ +#define MAX98520_PCM_MODE_CFG_PCM_BCLKEDGE (0x1 << 4) +#define MAX98520_PCM_CLK_SETUP_BSEL_MASK (0xF << 0) + +/* MAX98520_R2042_PCM_SR_SETUP */ +#define MAX98520_PCM_SR_SHIFT (0) +#define MAX98520_IVADC_SR_SHIFT (4) +#define MAX98520_PCM_SR_MASK (0xF << MAX98520_PCM_SR_SHIFT) +#define MAX98520_IVADC_SR_MASK (0xF << MAX98520_IVADC_SR_SHIFT) +#define MAX98520_PCM_SR_8000 (0x0) +#define MAX98520_PCM_SR_11025 (0x1) +#define MAX98520_PCM_SR_12000 (0x2) +#define MAX98520_PCM_SR_16000 (0x3) +#define MAX98520_PCM_SR_22050 (0x4) +#define MAX98520_PCM_SR_24000 (0x5) +#define MAX98520_PCM_SR_32000 (0x6) +#define MAX98520_PCM_SR_44100 (0x7) +#define MAX98520_PCM_SR_48000 (0x8) +#define MAX98520_PCM_SR_88200 (0x9) +#define MAX98520_PCM_SR_96000 (0xA) +#define MAX98520_PCM_SR_176400 (0xB) +#define MAX98520_PCM_SR_192000 (0xC) + +/* MAX98520_R2044_PCM_RX_SRC2 */ +#define MAX98520_PCM_DMIX_CH1_SHIFT (0xF << 0) +#define MAX98520_PCM_DMIX_CH0_SRC_MASK (0xF << 0) +#define MAX98520_PCM_DMIX_CH1_SRC_MASK (0xF << MAX98520_PCM_DMIX_CH1_SHIFT) + +/* MAX98520_R204F_PCM_RX_EN */ +#define MAX98520_PCM_RX_EN_MASK (0x1 << 0) +#define MAX98520_PCM_RX_BYP_EN_MASK (0x1 << 1) + +/* MAX98520_R2092_AMP_DSP_CFG */ +#define MAX98520_DSP_SPK_DCBLK_EN_SHIFT (0) +#define MAX98520_DSP_SPK_DITH_EN_SHIFT (1) +#define MAX98520_DSP_SPK_INVERT_SHIFT (2) +#define MAX98520_DSP_SPK_VOL_RMPUP_SHIFT (3) +#define MAX98520_DSP_SPK_VOL_RMPDN_SHIFT (4) +#define MAX98520_DSP_SPK_SAFE_EN_SHIFT (5) + +#define MAX98520_SPK_SAFE_EN_MASK (0x1 << MAX98520_DSP_SPK_SAFE_EN_SHIFT) + +/* MAX98520_R2094_SSM_CFG */ +#define MAX98520_SSM_EN_SHIFT (0) +#define MAX98520_SSM_MOD_SHIFT (1) +#define MAX98520_SSM_RCVR_MODE_SHIFT (3) + +/* MAX98520_R2095_AMP_CFG */ +#define MAX98520_CFG_DYN_MODE_SHIFT (4) +#define MAX98520_CFG_SPK_MODE_SHIFT (3) + +/* MAX98520_R20D0_DHT_CFG1 */ +#define MAX98520_DHT_VROT_PNT_SHIFT (0) + +/* MAX98520_R20D1_LIMITER_CFG1 */ +#define MAX98520_DHT_SUPPLY_HR_SHIFT (0) + +/* MAX98520_R20D2_DHT_CFG2 */ +#define MAX98520_DHT_LIMITER_MODE_SHIFT (0) +#define MAX98520_DHT_LIMITER_THRESHOLD_SHIFT (1) + +/* MAX98520_R20D3_DHT_CFG2 */ +#define MAX98520_DHT_MAX_ATTEN_SHIFT (0) + +/* MAX98520_R20D6_DHT_HYSTERESIS_CFG */ +#define MAX98520_DHT_HYSTERESIS_SWITCH_SHIFT (0) +#define MAX98520_DHT_HYSTERESIS_SHIFT (1) + +/* MAX98520_R20B2_ADC_PVDD0_CFG, MAX98520_R20B3_ADC_THERMAL_CFG */ +#define MAX98520_FLT_EN_SHIFT (4) + +struct max98520_priv { + struct regmap *regmap; + int reset_gpio; + unsigned int ch_size; + bool tdm_mode; +}; +#endif + -- 2.25.1

2 1

[PATCH] ASoC: rt5682: fix a little pop while playback
by derek.fang＠realtek.com 14 Oct '21

14 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> A little pop can be heard obviously from HP while playing a silent. This patch fixes it by using two functions: 1. Enable HP 1bit output mode. 2. Change the charge pump switch size during playback on and off. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 56 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt5682.h | 20 ++++++++++++++ 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 914fe7debc05..78b4cb5fb6c8 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -46,6 +46,8 @@ static const struct reg_sequence patch_list[] = { {RT5682_SAR_IL_CMD_1, 0x22b7}, {RT5682_SAR_IL_CMD_3, 0x0365}, {RT5682_SAR_IL_CMD_6, 0x0110}, + {RT5682_CHARGE_PUMP_1, 0x0210}, + {RT5682_HP_LOGIC_CTRL_2, 0x0007}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -1515,21 +1517,29 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_write(component, - RT5682_HP_LOGIC_CTRL_2, 0x0012); - snd_soc_component_write(component, - RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_AMP_MUTE, 0); + snd_soc_component_update_bits(component, RT5682_HP_LOGIC_CTRL_2, + RT5682_HP_LC2_SIG_SOUR2_MASK, RT5682_HP_LC2_SIG_SOUR2_REG); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN); + usleep_range(5000, 10000); + snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, + RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L); break; case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, 0); + snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, + RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_M); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x0); - snd_soc_component_write(component, - RT5682_HP_CTRL_2, 0x0000); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; @@ -1637,6 +1647,23 @@ static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); +static const unsigned int rt5682_hpo_sig_out_values[] = { + 2, + 7, +}; + +static const char * const rt5682_hpo_sig_out_mode[] = { + "Legacy", + "OneBit", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_hpo_sig_out_enum, + RT5682_HP_LOGIC_CTRL_2, 0, RT5682_HP_LC2_SIG_SOUR1_MASK, + rt5682_hpo_sig_out_mode, rt5682_hpo_sig_out_values); + +static const struct snd_kcontrol_new rt5682_hpo_sig_demux = + SOC_DAPM_ENUM("HPO Signal Demux", rt5682_hpo_sig_out_enum); + static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, 0, NULL, 0), @@ -1820,6 +1847,10 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, &hpor_switch), + SND_SOC_DAPM_OUT_DRV("HPO Legacy", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HPO OneBit", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_DEMUX("HPO Signal Demux", SND_SOC_NOPM, 0, 0, &rt5682_hpo_sig_demux), + /* CLK DET */ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1987,10 +2018,19 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "Vref1"}, - {"HPOL Playback", "Switch", "HP Amp"}, - {"HPOR Playback", "Switch", "HP Amp"}, + + {"HPO Signal Demux", NULL, "HP Amp"}, + + {"HPO Legacy", "Legacy", "HPO Signal Demux"}, + {"HPO OneBit", "OneBit", "HPO Signal Demux"}, + + {"HPOL Playback", "Switch", "HPO Legacy"}, + {"HPOR Playback", "Switch", "HPO Legacy"}, + {"HPOL", NULL, "HPOL Playback"}, {"HPOR", NULL, "HPOR Playback"}, + {"HPOL", NULL, "HPO OneBit"}, + {"HPOR", NULL, "HPO OneBit"}, }; static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 262512babc50..d93829c35585 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -375,6 +375,14 @@ #define RT5682_R_VOL_MASK (0x3f) #define RT5682_R_VOL_SFT 0 +/* Headphone Amp Control 2 (0x0003) */ +#define RT5682_HP_C2_DAC_AMP_MUTE_SFT 15 +#define RT5682_HP_C2_DAC_AMP_MUTE (0x1 << 15) +#define RT5682_HP_C2_DAC_L_EN_SFT 14 +#define RT5682_HP_C2_DAC_L_EN (0x1 << 14) +#define RT5682_HP_C2_DAC_R_EN_SFT 13 +#define RT5682_HP_C2_DAC_R_EN (0x1 << 13) + /*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ #define RT5682_G_HP (0xf << 8) #define RT5682_G_HP_SFT 8 @@ -1265,6 +1273,10 @@ #define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) /* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_SW_SIZE_MASK (0x7 << 8) +#define RT5682_CP_SW_SIZE_L (0x4 << 8) +#define RT5682_CP_SW_SIZE_M (0x2 << 8) +#define RT5682_CP_SW_SIZE_S (0x1 << 8) #define RT5682_CP_CLK_HP_MASK (0x3 << 4) #define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) #define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) @@ -1315,6 +1327,14 @@ #define RT5682_DEB_STO_DAC_MASK (0x7 << 4) #define RT5682_DEB_80_MS (0x0 << 4) +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5682_HP_LC2_SIG_SOUR2_MASK (0x1 << 4) +#define RT5682_HP_LC2_SIG_SOUR2_REG (0x1 << 4) +#define RT5682_HP_LC2_SIG_SOUR2_DC_CAL (0x0 << 4) +#define RT5682_HP_LC2_SIG_SOUR1_MASK (0x7) +#define RT5682_HP_LC2_SIG_SOUR1_1BIT (0x7) +#define RT5682_HP_LC2_SIG_SOUR1_LEGA (0x2) + /* SAR ADC Inline Command Control 1 (0x0210) */ #define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) #define RT5682_SAR_BUTT_DET_EN (0x1 << 15) -- 2.17.1

2 1

[PATCH] ALSA: usb-audio: add Schiit Hel device to quirk table
by Greg Kroah-Hartman 14 Oct '21

14 Oct '21

The Shciit Hel device responds to the ctl message for the mic capture switch with a timeout of -EPIPE: usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 This seems safe to ignore as the device works properly with the control message quirk, so add it to the quirk table so all is good. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-usb(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6ee6d24c847f..ec11be366dc0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1884,6 +1884,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ + QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ -- 2.33.0

2 1

[RFC] sound: cs35l41: Add support for Legion 7 16ACHg6 laptop
by Lucas Tanure 14 Oct '21

14 Oct '21

Hi, I would like to get some guidance about this solution to support the 16ACHg6 laptop. Hardware: - The 16ACHg6 laptop has two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - To load the firmware for CS35L41, this solution will require the wm_adsp library, which requires regmap, header definitions and register tables. - To minimize the duplication of the code, the HDA functions will be placed inside the ASoC CS35L41 driver. - Finally, HDA patch_realtek will access exposed functions from ASoC CS35L41 driver to initialize the amplifiers, start and stop streams and load firmware. Notes: - This is a work in progress, so the code is not functional, its only intent is to demonstrate the overall solution - If accepted, this will be split into a couple of patches for a new patch chain Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- drivers/acpi/scan.c | 1 + drivers/platform/x86/i2c-multi-instantiate.c | 7 ++ include/sound/cs35l41.h | 4 ++ sound/pci/hda/Kconfig | 1 + sound/pci/hda/patch_realtek.c | 21 +++++- sound/soc/codecs/cs35l41.c | 75 ++++++++++++++++---- 6 files changed, 95 insertions(+), 14 deletions(-) diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 5b54c80b9d32..c1c27a408420 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1703,6 +1703,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"BSG2150", }, {"INT33FE", }, {"INT3515", }, + {"CLSA0100", }, {} }; diff --git a/drivers/platform/x86/i2c-multi-instantiate.c b/drivers/platform/x86/i2c-multi-instantiate.c index a50153ecd560..b61f7e30d42a 100644 --- a/drivers/platform/x86/i2c-multi-instantiate.c +++ b/drivers/platform/x86/i2c-multi-instantiate.c @@ -139,6 +139,12 @@ static const struct i2c_inst_data bsg2150_data[] = { {} }; +static const struct i2c_inst_data clsa0100_data[] = { + { "cs35l41", IRQ_RESOURCE_GPIO, 0 }, + { "cs35l41", IRQ_RESOURCE_GPIO, 0 }, + {} +}; + /* * Device with _HID INT3515 (TI PD controllers) has some unresolved interrupt * issues. The most common problem seen is interrupt flood. @@ -170,6 +176,7 @@ static const struct i2c_inst_data bsg2150_data[] = { static const struct acpi_device_id i2c_multi_inst_acpi_ids[] = { { "BSG1160", (unsigned long)bsg1160_data }, { "BSG2150", (unsigned long)bsg2150_data }, + { "CLSA0100", (unsigned long)clsa0100_data }, { } }; MODULE_DEVICE_TABLE(acpi, i2c_multi_inst_acpi_ids); diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h index 1f1e3c6c9be1..4d665b7dbfdf 100644 --- a/include/sound/cs35l41.h +++ b/include/sound/cs35l41.h @@ -23,6 +23,8 @@ struct cs35l41_irq_cfg { }; struct cs35l41_platform_data { + bool no_bst; + bool hda; int bst_ind; int bst_ipk; int bst_cap; @@ -31,4 +33,6 @@ struct cs35l41_platform_data { struct cs35l41_irq_cfg irq_config2; }; +void cs35l41_hda_init(void); + #endif /* __CS35L41_H */ diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ab9d2746e804..37202466f033 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -95,6 +95,7 @@ config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC select SND_HDA_GENERIC_LEDS + select SND_SOC_CS35L41_I2C help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..2a0ac9a1b613 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -21,6 +21,7 @@ #include <sound/core.h> #include <sound/jack.h> #include <sound/hda_codec.h> +#include <sound/cs35l41.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -6443,6 +6444,18 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, } } +static void alc287_fixup_lenovo_y760(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PROBE) { + codec_info(cdc, "HDA_FIXUP_ACT_PROBE\n"); + cs35l41_hda_init(); + } else if (action == HDA_FIXUP_ACT_INIT) { + codec_info(cdc, "HDA_FIXUP_ACT_INIT\n"); + } else if (action == HDA_FIXUP_ACT_FREE) { + codec_info(cdc, "HDA_FIXUP_ACT_FREE\n"); + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6663,7 +6676,8 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, - ALC287_FIXUP_13S_GEN2_SPEAKERS + ALC287_FIXUP_13S_GEN2_SPEAKERS, + ALC287_FIXUP_LENOVO_Y760 }; static const struct hda_fixup alc269_fixups[] = { @@ -8361,6 +8375,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC287_FIXUP_LENOVO_Y760] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_lenovo_y760, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8755,6 +8773,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3847, "Legion Y760", ALC287_FIXUP_LENOVO_Y760), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index b16eb6610c0e..f643ed1b48c0 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -21,9 +21,17 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include <linux/acpi.h> #include "cs35l41.h" +static struct list_head *cs35l41_hda_lst; + +struct cs35l41_hda_node { + struct list_head node; + struct cs35l41_private *cs35l41; +}; + static const char * const cs35l41_supplies[CS35L41_NUM_SUPPLIES] = { "VA", "VP", @@ -1039,9 +1047,7 @@ static int cs35l41_set_pdata(struct cs35l41_private *cs35l41) { int ret; - /* Set Platform Data */ - /* Required */ - if (cs35l41->pdata.bst_ipk && + if (!cs35l41->pdata.no_bst && cs35l41->pdata.bst_ipk && cs35l41->pdata.bst_ind && cs35l41->pdata.bst_cap) { ret = cs35l41_boost_config(cs35l41, cs35l41->pdata.bst_ind, cs35l41->pdata.bst_cap, @@ -1051,8 +1057,7 @@ static int cs35l41_set_pdata(struct cs35l41_private *cs35l41) return ret; } } else { - dev_err(cs35l41->dev, "Incomplete Boost component DT config\n"); - return -EINVAL; + dev_info(cs35l41->dev, "Boost disabled\n"); } /* Optional */ @@ -1148,9 +1153,31 @@ static int cs35l41_handle_pdata(struct device *dev, { struct cs35l41_irq_cfg *irq_gpio1_config = &pdata->irq_config1; struct cs35l41_irq_cfg *irq_gpio2_config = &pdata->irq_config2; + struct acpi_device *adev; + struct device *phys_dev; unsigned int val; int ret; + if (memcmp(dev_name(cs35l41->dev), "i2c-CLSA0100", 12) == 0) { + pdata->no_bst = true; + pdata->hda = true; + adev = acpi_dev_get_first_match_dev("CLSA0100", "1", -1); + if (!adev) { + dev_err(dev, "Failed to find an ACPI device\n"); + return -ENODEV; + } + + phys_dev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + if (!phys_dev) { + dev_err(dev, "Failed to find a physical device\n"); + return -ENODEV; + } + cs35l41->reset_gpio = gpiod_get_index(phys_dev, NULL, 0, GPIOD_ASIS); + return 0; + } + ret = device_property_read_u32(dev, "cirrus,boost-peak-milliamp", &val); if (ret >= 0) pdata->bst_ipk = val; @@ -1237,10 +1264,22 @@ static const struct reg_sequence cs35l41_revb2_errata_patch[] = { { 0x00000040, 0x00003333 }, }; +void cs35l41_hda_init(void) +{ + struct list_head *p; + int i = 0; + + list_for_each(p, cs35l41_hda_lst) { + pr_info("%s %d\n", __func__, i++); + } +} +EXPORT_SYMBOL_GPL(cs35l41_hda_init); + int cs35l41_probe(struct cs35l41_private *cs35l41, struct cs35l41_platform_data *pdata) { u32 regid, reg_revid, i, mtl_revid, int_status, chipid_match; + struct cs35l41_hda_node *cs35l41_hda; int irq_pol = 0; int ret; @@ -1269,8 +1308,8 @@ int cs35l41_probe(struct cs35l41_private *cs35l41, } /* returning NULL can be an option if in stereo mode */ - cs35l41->reset_gpio = devm_gpiod_get_optional(cs35l41->dev, "reset", - GPIOD_OUT_LOW); + if (!cs35l41->reset_gpio) + cs35l41->reset_gpio = devm_gpiod_get_optional(cs35l41->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l41->reset_gpio)) { ret = PTR_ERR(cs35l41->reset_gpio); cs35l41->reset_gpio = NULL; @@ -1413,12 +1452,22 @@ int cs35l41_probe(struct cs35l41_private *cs35l41, goto err; } - ret = devm_snd_soc_register_component(cs35l41->dev, - &soc_component_dev_cs35l41, - cs35l41_dai, ARRAY_SIZE(cs35l41_dai)); - if (ret < 0) { - dev_err(cs35l41->dev, "Register codec failed: %d\n", ret); - goto err; + if (!cs35l41->pdata.hda) { + ret = devm_snd_soc_register_component(cs35l41->dev, + &soc_component_dev_cs35l41, + cs35l41_dai, ARRAY_SIZE(cs35l41_dai)); + if (ret < 0) { + dev_err(cs35l41->dev, "Register codec failed: %d\n", ret); + goto err; + } + } else { + if (!cs35l41_hda_lst) { + cs35l41_hda_lst = devm_kzalloc(cs35l41->dev, sizeof(*cs35l41_hda_lst), + GFP_KERNEL); + INIT_LIST_HEAD(cs35l41_hda_lst); + } + cs35l41_hda = devm_kzalloc(cs35l41->dev, sizeof(*cs35l41_hda), GFP_KERNEL); + list_add(&cs35l41_hda->node, cs35l41_hda_lst); } dev_info(cs35l41->dev, "Cirrus Logic CS35L41 (%x), Revision: %02X\n", -- 2.33.0

3 3

[PATCH] ALSA: usb-audio: allow -EPIPE errors for some v2 messages
by Greg Kroah-Hartman 14 Oct '21

14 Oct '21

The Schiit Hel device does not like to respond to all get_ctl_value_v2() requests for some reason. This used to work in older kernels, but now with more strict checking, this failure causes the device to fail to work. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- This fixes the Shiit Hel device that I have. It used to work on older kernels (a year ago?), but stopped working for some reason and I didn't take the time to track it down. This change fixes the issue for me, but feels wrong for some reason. At least now I can use the device as a headphone driver, much better than the built-in one for my current machine... If needed, I can take the time to do bisection to track down the real issue here, it might be due to stricter endpoint checking in the USB core, but that feels wrong somehow. Here's the debugfs USB output for this device, showing the endpoints: T: Bus=07 Lev=02 Prnt=02 Port=01 Cnt=01 Dev#= 5 Spd=480 MxCh= 0 D: Ver= 2.00 Cls=ef(misc ) Sub=02 Prot=01 MxPS=64 #Cfgs= 1 P: Vendor=30be ProdID=0101 Rev=01.02 S: Manufacturer=Schiit Audio S: Product=Schiit Hel C: #Ifs= 4 Cfg#= 1 Atr=c0 MxPwr=0mA I: If#= 0 Alt= 0 #EPs= 1 Cls=01(audio) Sub=01 Prot=20 Driver=snd-usb-audio E: Ad=8f(I) Atr=03(Int.) MxPS= 6 Ivl=1ms I: If#= 1 Alt= 1 #EPs= 2 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=05(O) Atr=05(Isoc) MxPS= 104 Ivl=125us E: Ad=85(I) Atr=11(Isoc) MxPS= 4 Ivl=1ms I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=88(I) Atr=05(Isoc) MxPS= 156 Ivl=125us I: If#= 3 Alt= 0 #EPs= 1 Cls=03(HID ) Sub=00 Prot=00 Driver=usbhid E: Ad=84(I) Atr=03(Int.) MxPS= 16 Ivl=1ms Any other suggestions to fix this are welcome. sound/usb/mixer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a2ce535df14b..37d3d697776b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -371,11 +371,11 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, validx, idx, buf, size); snd_usb_unlock_shutdown(chip); - if (ret < 0) { + if ((ret < 0) && (ret != -EPIPE)) { error: usb_audio_err(chip, - "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, idx, cval->val_type); + "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, ret = %d\n", + request, validx, idx, cval->val_type, ret); return ret; } -- 2.33.0

2 11

[PATCH] ASoC: soc-compress: prevent the potentially use of null pointer
by Jiasheng Jiang 14 Oct '21

14 Oct '21

The codec_dai and the cpu_dai are from rtd. When the rtd->dais[0] or rtd->dais[rtd->num_cpus] is None, it could lead to the use of null pointer in the snd_soc_dai_stream_valid(). So it might be better to add the check before the use of snd_soc_dai_stream_valid(). Fixes: 467fece ("ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/soc-compress.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b4f5935..67c3df1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,12 +535,14 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - playback = 1; - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - capture = 1; + if (codec_dai && cpu_dai) { + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + playback = 1; + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + capture = 1; + } /* * Compress devices are unidirectional so only one of the directions -- 2.7.4

2 1

[PATCH] ASoC: dt-bindings: rockchip: i2s-tdm: Fix rockchip, i2s-[rt]x-route
by Geert Uytterhoeven 13 Oct '21

13 Oct '21

make dt_binding_check: Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml: properties:rockchip,i2s-rx-route: {'required': ['maxItems']} is not allowed for {'$ref': '/schemas/types.yaml#/definitions/uint32-array', 'description': 'Defines the mapping of I2S RX sdis to I2S data bus lines. By default, they are mapped one-to-one. rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0.', 'maxItems': 4, 'items': [{'enum': [0, 1, 2, 3]}]} hint: "maxItems" is not needed with an "items" list from schema $id: http://devicetree.org/meta-schemas/items.yaml# Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml: properties:rockchip,i2s-tx-route: {'required': ['maxItems']} is not allowed for {'$ref': '/schemas/types.yaml#/definitions/uint32-array', 'description': 'Defines the mapping of I2S TX sdos to I2S data bus lines. By default, they are mapped one-to-one. rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0.', 'maxItems': 4, 'items': [{'enum': [0, 1, 2, 3]}]} hint: "maxItems" is not needed with an "items" list from schema $id: http://devicetree.org/meta-schemas/items.yaml# Drop the "-" from the enums to fix this. Fixes: 510f1c133aedcf69 ("ASoC: dt-bindings: rockchip: add i2s-tdm bindings") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- v4/v5 differed slightly from what Rob suggested in his review of v3 https://lore.kernel.org/all/YS6kHkeOKCJMOOIf@robh.at.kernel.org/ --- Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml index dce8b4136ec6db4e..ce3e18b50230dc9b 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml @@ -112,7 +112,7 @@ properties: rockchip,i2s-rx-route = <3> would mean sdi3 is receiving from data0. maxItems: 4 items: - - enum: [0, 1, 2, 3] + enum: [0, 1, 2, 3] rockchip,i2s-tx-route: $ref: /schemas/types.yaml#/definitions/uint32-array @@ -122,7 +122,7 @@ properties: rockchip,i2s-tx-route = <3> would mean sdo3 is sending to data0. maxItems: 4 items: - - enum: [0, 1, 2, 3] + enum: [0, 1, 2, 3] rockchip,io-multiplex: description: -- 2.25.1

2 1

[PATCH v2] ASoC: rt1011: Fix 'I2S Reference' enum control
by Peter Ujfalusi 13 Oct '21

13 Oct '21

There are several things the patch adding the support for 'I2S Reference' got wrong: - "None" selection is in fact equals to last selected reference - The custom put overrides RX/TX len, TDM slot sizes, etc - the enum is useless in most part for the reference tracking - there is no need for EXT control as there is a single bit in RT1011_TDM1_SET_1 register (bit 7) which selects the reference - it was using ucontrol->value.integer.value[0] in the put/get callbacks which causesed access to 'I2S Reference' enum with alsamixer to fail Complements: c3de683c4d1d ("ASoC: rt1011: Fix 'I2S Reference' enum control caused error") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v1: - Rebased on top of next - changed the tag from Fixes to Complements to make sure that the crash fix and the fix for the implementation stays together. This patch is an iteration on https://lore.kernel.org/alsa-devel/20211011144518.2518-1-peter.ujfalusi@lin… to fix 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011"). In essence it is reverting the original patch to use SOC_ENUM_SINGLE_DECL/SOC_ENUM to handle the bit to select the I2S reference. Regards, Peter sound/soc/codecs/rt1011.c | 53 ++++----------------------------------- sound/soc/codecs/rt1011.h | 7 ------ 2 files changed, 5 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index bdfcbb81fa19..297af7ff824c 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1312,54 +1312,12 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, } static const char * const rt1011_i2s_ref_texts[] = { - "None", "Left Channel", "Right Channel" + "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_EXT_DECL(rt1011_i2s_ref_enum, rt1011_i2s_ref_texts); - -static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.enumerated.item[0]; - - switch (i2s_ref_ch) { - case RT1011_I2S_REF_LEFT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - case RT1011_I2S_REF_RIGHT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - default: - dev_info(component->dev, "I2S Reference: Do nothing\n"); - } - - rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; - - return 0; -} - -static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - - ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; - - return 0; -} +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, + RT1011_TDM1_SET_1, 7, + rt1011_i2s_ref_texts); static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ @@ -1400,8 +1358,7 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), /* I2S Reference */ - SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, - rt1011_i2s_ref_get, rt1011_i2s_ref_put), + SOC_ENUM("I2S Reference", rt1011_i2s_ref_enum), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index afb2fad94216..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,12 +654,6 @@ enum { RT1011_AIFS }; -enum { - RT1011_I2S_REF_NONE, - RT1011_I2S_REF_LEFT_CH, - RT1011_I2S_REF_RIGHT_CH, -}; - /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -698,7 +692,6 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; - unsigned int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- 2.33.0

2 1

[PATCH] ASoC: rt1011: Fix 'I2S Reference' enum control
by Peter Ujfalusi 13 Oct '21

13 Oct '21

There are several things the patch adding the support for 'I2S Reference' got wrong: - "None" selection is in fact equals to last selected reference - The custom put overrides RX/TX len, TDM slot sizes, etc - the enum is useless in most part for the reference tracking - there is no need for EXT control as there is a single bit in RT1011_TDM1_SET_1 register (bit 7) which selects the reference - it was using ucontrol->value.integer.value[0] in the put/get callbacks which causes access to 'I2S Reference' enum with alsamixer to fail: $ alsamixer cannot load mixer controls: Invalid argument cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow It should have used ucontrol->value.enumerated.item[0], but since there is no need for the custom code, it does not really matter. Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, This patch is an iteration on https://lore.kernel.org/alsa-devel/20211011144518.2518-1-peter.ujfalusi@lin… to fix 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011"). In essence it is reverting the original patch to use SOC_ENUM_SINGLE_DECL/SOC_ENUM to handle the bit to select the I2S reference. Regards, Peter sound/soc/codecs/rt1011.c | 56 +++++---------------------------------- sound/soc/codecs/rt1011.h | 7 ----- 2 files changed, 6 insertions(+), 57 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 508597866dff..297af7ff824c 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,56 +1311,13 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref[] = { - "None", "Left Channel", "Right Channel" +static const char * const rt1011_i2s_ref_texts[] = { + "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, - rt1011_i2s_ref); - -static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.integer.value[0]; - - switch (i2s_ref_ch) { - case RT1011_I2S_REF_LEFT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - case RT1011_I2S_REF_RIGHT_CH: - regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); - regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); - regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); - break; - default: - dev_info(component->dev, "I2S Reference: Do nothing\n"); - } - - rt1011->i2s_ref = ucontrol->value.integer.value[0]; - - return 0; -} - -static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct rt1011_priv *rt1011 = - snd_soc_component_get_drvdata(component); - - ucontrol->value.integer.value[0] = rt1011->i2s_ref; - - return 0; -} +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, + RT1011_TDM1_SET_1, 7, + rt1011_i2s_ref_texts); static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ @@ -1401,8 +1358,7 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), /* I2S Reference */ - SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, - rt1011_i2s_ref_get, rt1011_i2s_ref_put), + SOC_ENUM("I2S Reference", rt1011_i2s_ref_enum), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index afb2fad94216..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,12 +654,6 @@ enum { RT1011_AIFS }; -enum { - RT1011_I2S_REF_NONE, - RT1011_I2S_REF_LEFT_CH, - RT1011_I2S_REF_RIGHT_CH, -}; - /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -698,7 +692,6 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; - unsigned int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- 2.33.0

2 1

[PATCH v4 0/2] ASoC: rt9120: Add Richtek RT9120 supprot
by cy_huang 13 Oct '21

13 Oct '21

From: ChiYuan Huang <cy_huang(a)richtek.com> This patch series Add the Richtek RT9120 support. In v4: - Add 'classd_tlv' for 'SPK Gain Volume' control item. - Unify the tlv declaration to the postfix '_tlv'. - Fix 'digital_tlv' mute as 1 to declare the minimum is muted. In v3: - Add dvdd regulator binding to check the dvdd voltage domain. - Refine sdo_select_text. - Use switch case in 'internal_power_event' function. - Remove the volume and mute initially write in component probe. - Remove the mute API. It's no need by HW design. In v2: - Add missing #sound-dai-cells property. ChiYuan Huang (2): ASoC: dt-bindings: rt9120: Add initial bindings ASoC: rt9120: Add rt9210 audio amplifier support .../devicetree/bindings/sound/richtek,rt9120.yaml | 59 +++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt9120.c | 495 +++++++++++++++++++++ 4 files changed, 566 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml create mode 100644 sound/soc/codecs/rt9120.c -- 2.7.4

2 3

[PATCH v2] ASoC: wm8960: Fix clock configuration on slave mode
by Shengjiu Wang 13 Oct '21

13 Oct '21

There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - change error message to warning - add more comments sound/soc/codecs/wm8960.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392..499604f1e178 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,9 +742,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; } -- 2.17.1

3 2

[PATCH] ASoC: amd: vangogh: constify static struct snd_soc_dai_ops
by Rikard Falkeborn 13 Oct '21

13 Oct '21

The only usage of acp5x_i2s_dai_ops is to assign its address to the ops field in the snd_soc_dai_driver struct, which is a pointer to const. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/amd/vangogh/acp5x-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 2705e57eb713..002db3971ca9 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -348,7 +348,7 @@ static int acp5x_i2s_trigger(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_dai_ops acp5x_i2s_dai_ops = { +static const struct snd_soc_dai_ops acp5x_i2s_dai_ops = { .hw_params = acp5x_i2s_hwparams, .trigger = acp5x_i2s_trigger, .set_fmt = acp5x_i2s_set_fmt, -- 2.33.0

2 1

[PATCH] ASoC: mediatek: Constify static snd_soc_ops
by Rikard Falkeborn 13 Oct '21

13 Oct '21

These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 2 +- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 44a8d5cfb0aa..d9fd6eb786aa 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -146,7 +146,7 @@ static int mt2701_cs42448_be_ops_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt2701_cs42448_be_ops = { +static const struct snd_soc_ops mt2701_cs42448_be_ops = { .hw_params = mt2701_cs42448_be_ops_hw_params }; diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 414e422c0eba..f56de1b918bf 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -40,7 +40,7 @@ static int mt2701_wm8960_be_ops_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops mt2701_wm8960_be_ops = { +static const struct snd_soc_ops mt2701_wm8960_be_ops = { .hw_params = mt2701_wm8960_be_ops_hw_params }; diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 94dcbd36c869..aeb1af86047e 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -335,7 +335,7 @@ static void mt8183_mt6358_tdm_shutdown(struct snd_pcm_substream *substream) __func__, ret); } -static struct snd_soc_ops mt8183_mt6358_tdm_ops = { +static const struct snd_soc_ops mt8183_mt6358_tdm_ops = { .startup = mt8183_mt6358_tdm_startup, .shutdown = mt8183_mt6358_tdm_shutdown, }; -- 2.33.0

2 1

[PATCH v3] component: do not leave master devres group open after bind
by Kai Vehmanen 13 Oct '21

13 Oct '21

In current code, the devres group for aggregate master is left open after call to component_master_add_*(). This leads to problems when the master does further managed allocations on its own. When any participating driver calls component_del(), this leads to immediate release of resources. This came up when investigating a page fault occurring with i915 DRM driver unbind with 5.15-rc1 kernel. The following sequence occurs: i915_pci_remove() -> intel_display_driver_unregister() -> i915_audio_component_cleanup() -> component_del() -> component.c:take_down_master() -> hdac_component_master_unbind() [via master->ops->unbind()] -> devres_release_group(master->parent, NULL) With older kernels this has not caused issues, but with audio driver moving to use managed interfaces for more of its allocations, this no longer works. Devres log shows following to occur: component_master_add_with_match() [ 126.886032] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000323ccdc5 devm_component_match_release (24 bytes) [ 126.886045] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000865cdb29 grp< (0 bytes) [ 126.886049] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 grp< (0 bytes) audio driver completes its PCI probe() [ 126.892238] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 pcim_iomap_release (48 bytes) component_del() called() at DRM/i915 unbind() [ 137.579422] i915 0000:00:02.0: DEVRES REL 00000000ef44c293 grp< (0 bytes) [ 137.579445] snd_hda_intel 0000:00:1f.3: DEVRES REL 00000000865cdb29 grp< (0 bytes) [ 137.579458] snd_hda_intel 0000:00:1f.3: DEVRES REL 000000001b480725 pcim_iomap_release (48 bytes) So the "devres_release_group(master->parent, NULL)" ends up freeing the pcim_iomap allocation. Upon next runtime resume, the audio driver will cause a page fault as the iomap alloc was released without the driver knowing about it. Fix this issue by using the "struct master" pointer as identifier for the devres group, and by closing the devres group after the master->ops->bind() call is done. This allows devres allocations done by the driver acting as master to be isolated from the binding state of the aggregate driver. This modifies the logic originally introduced in commit 9e1ccb4a7700 ("drivers/base: fix devres handling for master device") Cc: stable(a)vger.kernel.org BugLink: https://gitlab.freedesktop.org/drm/intel/-/issues/4136 Fixes: 9e1ccb4a7700 ("drivers/base: fix devres handling for master device") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Acked-by: Imre Deak <imre.deak(a)intel.com> Acked-by: Russell King (Oracle) <rmk+kernel(a)armlinux.org.uk> --- drivers/base/component.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) V3 changes: - address feedback from Greg KH, add a Fixes tag and cc stable V2 changes: - after review form Imre and Russell, removing RFC tag - rebased on top of 5.15-rc2 (V1 was on drm-tip) - CI test results for V1 show that this patch fixes multiple failures in i915 unbind and module reload tests: https://patchwork.freedesktop.org/series/94889/ diff --git a/drivers/base/component.c b/drivers/base/component.c index 5e79299f6c3f..870485cbbb87 100644 --- a/drivers/base/component.c +++ b/drivers/base/component.c @@ -246,7 +246,7 @@ static int try_to_bring_up_master(struct master *master, return 0; } - if (!devres_open_group(master->parent, NULL, GFP_KERNEL)) + if (!devres_open_group(master->parent, master, GFP_KERNEL)) return -ENOMEM; /* Found all components */ @@ -258,6 +258,7 @@ static int try_to_bring_up_master(struct master *master, return ret; } + devres_close_group(master->parent, NULL); master->bound = true; return 1; } @@ -282,7 +283,7 @@ static void take_down_master(struct master *master) { if (master->bound) { master->ops->unbind(master->parent); - devres_release_group(master->parent, NULL); + devres_release_group(master->parent, master); master->bound = false; } } base-commit: 9e1ff307c779ce1f0f810c7ecce3d95bbae40896 -- 2.33.0

1 0

[PATCH 1/2] ASoC: dt-bindings: max98520: Add initial bindings
by George Song 13 Oct '21

13 Oct '21

From: George Song <george.song(a)analog.com> Add initial bindings for max98520 audio amplifier Signed-off-by: George Song <george.song(a)analog.com> --- .../bindings/sound/maxim,max98520.yaml | 35 +++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..d1f2b5ee775a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <George.song(a)analog.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; -- 2.25.1

1 1

[PATCH v2] component: do not leave master devres group open after bind
by Kai Vehmanen 13 Oct '21

13 Oct '21

In current code, the devres group for aggregate master is left open after call to component_master_add_*(). This leads to problems when the master does further managed allocations on its own. When any participating driver calls component_del(), this leads to immediate release of resources. This came up when investigating a page fault occurring with i915 DRM driver unbind with 5.15-rc1 kernel. The following sequence occurs: i915_pci_remove() -> intel_display_driver_unregister() -> i915_audio_component_cleanup() -> component_del() -> component.c:take_down_master() -> hdac_component_master_unbind() [via master->ops->unbind()] -> devres_release_group(master->parent, NULL) With older kernels this has not caused issues, but with audio driver moving to use managed interfaces for more of its allocations, this no longer works. Devres log shows following to occur: component_master_add_with_match() [ 126.886032] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000323ccdc5 devm_component_match_release (24 bytes) [ 126.886045] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000865cdb29 grp< (0 bytes) [ 126.886049] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 grp< (0 bytes) audio driver completes its PCI probe() [ 126.892238] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 pcim_iomap_release (48 bytes) component_del() called() at DRM/i915 unbind() [ 137.579422] i915 0000:00:02.0: DEVRES REL 00000000ef44c293 grp< (0 bytes) [ 137.579445] snd_hda_intel 0000:00:1f.3: DEVRES REL 00000000865cdb29 grp< (0 bytes) [ 137.579458] snd_hda_intel 0000:00:1f.3: DEVRES REL 000000001b480725 pcim_iomap_release (48 bytes) So the "devres_release_group(master->parent, NULL)" ends up freeing the pcim_iomap allocation. Upon next runtime resume, the audio driver will cause a page fault as the iomap alloc was released without the driver knowing about it. Fix this issue by using the "struct master" pointer as identifier for the devres group, and by closing the devres group after the master->ops->bind() call is done. This allows devres allocations done by the driver acting as master to be isolated from the binding state of the aggregate driver. This modifies the logic originally introduced in commit 9e1ccb4a7700 ("drivers/base: fix devres handling for master device") BugLink: https://gitlab.freedesktop.org/drm/intel/-/issues/4136 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Acked-by: Imre Deak <imre.deak(a)intel.com> Acked-by: Russell King (Oracle) <rmk+kernel(a)armlinux.org.uk> --- drivers/base/component.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) V2 changes: - after review form Imre and Russell, removing RFC tag - rebased on top of 5.15-rc2 (V1 was on drm-tip) - CI test results for V1 show that this patch fixes multiple failures in i915 unbind and module reload tests: https://patchwork.freedesktop.org/series/94889/ diff --git a/drivers/base/component.c b/drivers/base/component.c index 5e79299f6c3f..870485cbbb87 100644 --- a/drivers/base/component.c +++ b/drivers/base/component.c @@ -246,7 +246,7 @@ static int try_to_bring_up_master(struct master *master, return 0; } - if (!devres_open_group(master->parent, NULL, GFP_KERNEL)) + if (!devres_open_group(master->parent, master, GFP_KERNEL)) return -ENOMEM; /* Found all components */ @@ -258,6 +258,7 @@ static int try_to_bring_up_master(struct master *master, return ret; } + devres_close_group(master->parent, NULL); master->bound = true; return 1; } @@ -282,7 +283,7 @@ static void take_down_master(struct master *master) { if (master->bound) { master->ops->unbind(master->parent); - devres_release_group(master->parent, NULL); + devres_release_group(master->parent, master); master->bound = false; } } base-commit: e4e737bb5c170df6135a127739a9e6148ee3da82 -- 2.32.0

3 5

[PATCH v3] ASoC: rt1011: Fix 'I2S Reference' enum control caused error
by Peter Ujfalusi 13 Oct '21

13 Oct '21

Access to 'I2S Reference' enum causes alsamixer to fail to load: $ alsamixer cannot load mixer controls: Invalid argument cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow The reason is that the original patch adding the code was using ucontrol->value.integer.value[0] instead the correct ucontrol->value.enumerated.item[0] for an ENUM control. Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v2: - Fix typo in commit message s/Is@/I2S Changes since v1: - Correct the ENUM declaration as well Regards, Peter sound/soc/codecs/rt1011.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 508597866dff..bdfcbb81fa19 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,12 +1311,11 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref[] = { +static const char * const rt1011_i2s_ref_texts[] = { "None", "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, - rt1011_i2s_ref); +static SOC_ENUM_SINGLE_EXT_DECL(rt1011_i2s_ref_enum, rt1011_i2s_ref_texts); static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1325,7 +1324,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.integer.value[0]; + int i2s_ref_ch = ucontrol->value.enumerated.item[0]; switch (i2s_ref_ch) { case RT1011_I2S_REF_LEFT_CH: @@ -1344,7 +1343,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, dev_info(component->dev, "I2S Reference: Do nothing\n"); } - rt1011->i2s_ref = ucontrol->value.integer.value[0]; + rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; return 0; } @@ -1357,7 +1356,7 @@ static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - ucontrol->value.integer.value[0] = rt1011->i2s_ref; + ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; return 0; } -- 2.33.0

2 6

[PATCH] alsaloop: fixed parsing value 'auto' of option 'sync'
by Pavel Hofman 13 Oct '21

13 Oct '21

The value 'auto' was not checked which resulted in using SYNC_TYPE_NONE instead of SYNC_TYPE_AUTO. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 4192712..f5f2e37 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -528,6 +528,8 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, arg_sync = SYNC_TYPE_PLAYRATESHIFT; else if (optarg[0] == 'r') arg_sync = SYNC_TYPE_SAMPLERATE; + else if (optarg[0] == 'a') + arg_sync = SYNC_TYPE_AUTO; else arg_sync = atoi(optarg); if (arg_sync < 0 || arg_sync > SYNC_TYPE_LAST) -- 2.25.1

2 1

Re: [v2 2/2] ASoC : max98520 : Added AMplifier Driver Signed-off-by: George Song <george.song@maximintegrated.com>
by Mark Brown 13 Oct '21

13 Oct '21

On Wed, Oct 13, 2021 at 04:16:16PM +0900, George Song wrote: > From: "george.song" <george.song(a)analog.com> > > --- > sound/soc/codecs/Kconfig | 5 + You've not provided a Signed-off-by for this so I can't do anything with it, please see Documentation/process/submitting-patches.rst for details on what this is and why it's important. You've also sent two different versions of patch 2 but only one version of patch 1, please don't do this - if you need to send a new version please resend the whole thread otherwise it gets very hard to follow what's going on.

1 0

[PATCH v3] alsaloop: added option prateshift for PLAYSHIFT ctl elem used in PLAYSHIFT
by Pavel Hofman 13 Oct '21

13 Oct '21

If snd-aloop device is on playback side, the required sync mode is PLAYSHIFT. That means Loopback ctl elem "PCM Rate Shift 100000" of the corresponding capture side of the Loopback pipe must be controlled (by a reciprocal). ASCII name of the playback rate shift ctl elem is specified with newly added option -x/--prateshift, e.g.: -P hw:Loopback,0 -S playshift \ -x iface=PCM,name='PCM Rate Shift 100000',device=1 Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- Notes: * using snd_ctl_elem_id_alloca instead of snd_ctl_elem_id_malloc * removed corresponding snd_ctl_elem_id_free calls alsaloop/alsaloop.1 | 11 +++++-- alsaloop/alsaloop.c | 11 ++++++- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 77 +++++++++++++++++++++++++++++++++------------ 4 files changed, 77 insertions(+), 23 deletions(-) diff --git a/alsaloop/alsaloop.1 b/alsaloop/alsaloop.1 index 6bacfd5..006494d 100644 --- a/alsaloop/alsaloop.1 +++ b/alsaloop/alsaloop.1 @@ -57,6 +57,11 @@ Use given CTL device for playback. Use given CTL device for capture. +.TP +\fI\-x <ctl_ascii_name>\fP | \fI\-\-prateshift=<ctl_ascii_name>\fP + +Specify ctl ascii name for playshift sync mode (see the Examples section). + .TP \fI\-l <latency>\fP | \fI\-\-latency=<frames>\fP @@ -195,10 +200,12 @@ Verbose xrun profiling. Set process wake timeout. .SH EXAMPLES - -.TP +.nf \fBalsaloop \-C hw:0,0 \-P hw:1,0 \-t 50000\fR +\fBalsaloop \-C hw:soundcard,0 \-P hw:Loopback,0 \-t 50000 \-S playshift \\ +\-x iface=PCM,name='PCM Rate Shift 100000',device=1\fR +.ne .SH BUGS None known. .SH AUTHOR diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 06ffadf..4192712 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -175,6 +175,7 @@ void help(void) "-C,--cdevice capture device\n" "-X,--pctl playback ctl device\n" "-Y,--cctl capture ctl device\n" +"-x,--prateshift playback 'PCM Rate Shift 100000' ascii ctl name\n" "-l,--latency requested latency in frames\n" "-t,--tlatency requested latency in usec (1/1000000sec)\n" "-f,--format sample format\n" @@ -362,6 +363,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, {"cdevice", 1, NULL, 'C'}, {"pctl", 1, NULL, 'X'}, {"cctl", 1, NULL, 'Y'}, + {"prateshift", 1, NULL, 'x'}, {"latency", 1, NULL, 'l'}, {"tlatency", 1, NULL, 't'}, {"format", 1, NULL, 'f'}, @@ -391,6 +393,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, char *arg_cdevice = NULL; char *arg_pctl = NULL; char *arg_cctl = NULL; + char *arg_prateshift = NULL; unsigned int arg_latency_req = 0; unsigned int arg_latency_reqtime = 10000; snd_pcm_format_t arg_format = SND_PCM_FORMAT_S16_LE; @@ -420,7 +423,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, while (1) { int c; if ((c = getopt_long(argc, argv, - "hdg:P:C:X:Y:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", + "hdg:P:C:X:Y:x:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", long_option, NULL)) < 0) break; switch (c) { @@ -446,6 +449,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, case 'Y': arg_cctl = strdup(optarg); break; + case 'x': + arg_prateshift = strdup(optarg); + break; case 'l': err = atoi(optarg); arg_latency_req = err >= 4 ? err : 4; @@ -627,6 +633,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, logit(LOG_CRIT, "Unable to add ossmixer controls.\n"); exit(EXIT_FAILURE); } + if (arg_prateshift) + play->prateshift_name = arg_prateshift; + #ifdef USE_SAMPLERATE loop->src_enable = arg_samplerate > 0; if (loop->src_enable) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 1dbcefe..7a98ef3 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -127,6 +127,7 @@ struct loopback_handle { snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; snd_ctl_elem_value_t *ctl_channels; + char *prateshift_name; /* ascii name for the playback rate shift ctl elem */ }; struct loopback { diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 845ab82..13fa595 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1101,7 +1101,8 @@ void update_pitch(struct loopback *loop) #endif } else if (loop->sync == SYNC_TYPE_PLAYRATESHIFT) { - set_rate_shift(loop->play, pitch); + // pitch is capture-based, playback side requires reciprocal + set_rate_shift(loop->play, 1 / pitch); #ifdef USE_SAMPLERATE if (loop->use_samplerate) { loop->src_data.src_ratio = @@ -1172,44 +1173,80 @@ static int get_channels(struct loopback_handle *lhandle) return snd_ctl_elem_value_get_integer(lhandle->ctl_channels, 0); } -static void openctl_elem(struct loopback_handle *lhandle, - int device, int subdevice, - const char *name, - snd_ctl_elem_value_t **elem) +static int openctl_elem_id(struct loopback_handle *lhandle, snd_ctl_elem_id_t *id, + snd_ctl_elem_value_t **elem) { int err; if (snd_ctl_elem_value_malloc(elem) < 0) { *elem = NULL; + return -ENOMEM; + } + snd_ctl_elem_value_set_id(*elem, id); + snd_ctl_elem_value_set_interface(*elem, SND_CTL_ELEM_IFACE_PCM); + err = snd_ctl_elem_read(lhandle->ctl, *elem); + if (err < 0) { + snd_ctl_elem_value_free(*elem); + *elem = NULL; + return err; } else { - snd_ctl_elem_value_set_interface(*elem, - SND_CTL_ELEM_IFACE_PCM); - snd_ctl_elem_value_set_device(*elem, device); - snd_ctl_elem_value_set_subdevice(*elem, subdevice); - snd_ctl_elem_value_set_name(*elem, name); - err = snd_ctl_elem_read(lhandle->ctl, *elem); - if (err < 0) { - snd_ctl_elem_value_free(*elem); - *elem = NULL; - } else { - if (verbose) - snd_output_printf(lhandle->loopback->output, - "Opened PCM element %s of %s, device %d, subdevice %d\n", - name, snd_ctl_name(lhandle->ctl), device, subdevice); - } + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + snd_ctl_elem_id_get_name(id), snd_ctl_name(lhandle->ctl), + snd_ctl_elem_id_get_device(id), + snd_ctl_elem_id_get_subdevice(id)); + return 0; } } +static int openctl_elem(struct loopback_handle *lhandle, + int device, int subdevice, + const char *name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_id_set_device(id, device); + snd_ctl_elem_id_set_subdevice(id, subdevice); + snd_ctl_elem_id_set_name(id, name); + return openctl_elem_id(lhandle, id, elem); +} + +static int openctl_elem_ascii(struct loopback_handle *lhandle, char *ascii_name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + snd_ctl_elem_id_alloca(&id); + if (snd_ctl_ascii_elem_id_parse(id, ascii_name)) { + fprintf(stderr, "Wrong control identifier: %s\n", ascii_name); + return -EINVAL; + } + return openctl_elem_id(lhandle, id, elem); +} + static int openctl(struct loopback_handle *lhandle, int device, int subdevice) { int err; lhandle->ctl_rate_shift = NULL; if (lhandle->loopback->play == lhandle) { + // play only + if (lhandle->prateshift_name) { + err = openctl_elem_ascii(lhandle, lhandle->prateshift_name, + &lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Unable to open playback PCM Rate Shift elem '%s'.\n", + lhandle->prateshift_name); + exit(EXIT_FAILURE); + } + } if (lhandle->loopback->controls) goto __events; return 0; } + // capture only openctl_elem(lhandle, device, subdevice, "PCM Notify", &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", -- 2.25.1

2 1

The default jack detection type overridden by sof_sdw_quirk
by Chris Chiu 13 Oct '21

13 Oct '21

Hi, I have an Intel AlderLake machine which uses the sof-soundwire driver for audio functions. When the headset is plugged in, it will incorrectly trigger the event KEY_PLAYPAUSE and never stops. To fix the problem, I used to apply the quirk RT711_JD2 proposed in sound/soc/intel/boards/sof_sdw.c and the jack detect function will work as expected. However, when I look into the code of sound/soc/codecs/rt711.c and rt711-sdca.c, the default value for jd_src (https://github.com/torvalds/linux/blob/master/sound/soc/codecs/rt711.c#L1209) is already RT711_JD2, which means it is overridden by other drivers. After digging deeper, the jd_src value is overridden by rt711_add_codec_device_props() with the value RT711_JD1 comes from sof_sdw_quirk. Is there any reason why sof_sdw_quirk is assigned with RT711_JD1 by default? Can I simply fix it by assigning the sof_sdw_quirk to RT711_JD_NULL as follows? -unsigned long sof_sdw_quirk = RT711_JD1; +unsigned long sof_sdw_quirk = RT711_JD_NULL; Please let me know if there's any potential problem I didn't notice and suggest if there's any better solution. Thanks Chris

1 0

[PATCH] ASoC: atmel: add COMPILE_TEST for SND_ATMEL_SOC_PDC
by Randy Dunlap 13 Oct '21

13 Oct '21

Geert pointed out that since sound/soc has the soc_dummy_driver for NO_DMA platforms, it is possible (desirable) to have drivers that depend on HAS_DMA to alternately depend on COMPILE_TEST. This means that SND_ATMEL_SOC_PDC can depend on HAS_DMA || COMPIE_TEST. Fixes: 6c5c659dfe3f ("ASoC: atmel: ATMEL drivers don't need HAS_DMA") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20211001.orig/sound/soc/atmel/Kconfig +++ linux-next-20211001/sound/soc/atmel/Kconfig @@ -11,6 +11,7 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool + depends on HAS_DMA || COMPILE_TEST config SND_ATMEL_SOC_DMA bool

3 4

[PATCH 1/2] ASoC: dt-bindings: max98520: Add initial bindings
by George Song 13 Oct '21

13 Oct '21

Signed-off-by: George Song <george.song(a)analog.com> --- .../bindings/sound/maxim,max98520.yaml | 35 +++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..d1f2b5ee775a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <George.song(a)analog.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; -- 2.25.1

1 0

[PATCH v2] alsaloop: added option prateshift for PLAYSHIFT ctl elem used in PLAYSHIFT
by Pavel Hofman 13 Oct '21

13 Oct '21

If snd-aloop device is on playback side, the required sync mode is PLAYSHIFT. That means Loopback ctl elem "PCM Rate Shift 100000" of the corresponding capture side of the Loopback pipe must be controlled (by pitch reciprocal). ASCII name of the playback rate shift ctl elem is specified with newly added option -x/--prateshift, e.g.: -P hw:Loopback,0 -S playshift \ -x iface=PCM,name='PCM Rate Shift 100000',device=1 Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.1 | 11 ++++-- alsaloop/alsaloop.c | 11 +++++- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 83 ++++++++++++++++++++++++++++++++++----------- 4 files changed, 83 insertions(+), 23 deletions(-) diff --git a/alsaloop/alsaloop.1 b/alsaloop/alsaloop.1 index 6bacfd5..006494d 100644 --- a/alsaloop/alsaloop.1 +++ b/alsaloop/alsaloop.1 @@ -57,6 +57,11 @@ Use given CTL device for playback. Use given CTL device for capture. +.TP +\fI\-x <ctl_ascii_name>\fP | \fI\-\-prateshift=<ctl_ascii_name>\fP + +Specify ctl ascii name for playshift sync mode (see the Examples section). + .TP \fI\-l <latency>\fP | \fI\-\-latency=<frames>\fP @@ -195,10 +200,12 @@ Verbose xrun profiling. Set process wake timeout. .SH EXAMPLES - -.TP +.nf \fBalsaloop \-C hw:0,0 \-P hw:1,0 \-t 50000\fR +\fBalsaloop \-C hw:soundcard,0 \-P hw:Loopback,0 \-t 50000 \-S playshift \\ +\-x iface=PCM,name='PCM Rate Shift 100000',device=1\fR +.ne .SH BUGS None known. .SH AUTHOR diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 06ffadf..4192712 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -175,6 +175,7 @@ void help(void) "-C,--cdevice capture device\n" "-X,--pctl playback ctl device\n" "-Y,--cctl capture ctl device\n" +"-x,--prateshift playback 'PCM Rate Shift 100000' ascii ctl name\n" "-l,--latency requested latency in frames\n" "-t,--tlatency requested latency in usec (1/1000000sec)\n" "-f,--format sample format\n" @@ -362,6 +363,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, {"cdevice", 1, NULL, 'C'}, {"pctl", 1, NULL, 'X'}, {"cctl", 1, NULL, 'Y'}, + {"prateshift", 1, NULL, 'x'}, {"latency", 1, NULL, 'l'}, {"tlatency", 1, NULL, 't'}, {"format", 1, NULL, 'f'}, @@ -391,6 +393,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, char *arg_cdevice = NULL; char *arg_pctl = NULL; char *arg_cctl = NULL; + char *arg_prateshift = NULL; unsigned int arg_latency_req = 0; unsigned int arg_latency_reqtime = 10000; snd_pcm_format_t arg_format = SND_PCM_FORMAT_S16_LE; @@ -420,7 +423,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, while (1) { int c; if ((c = getopt_long(argc, argv, - "hdg:P:C:X:Y:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", + "hdg:P:C:X:Y:x:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", long_option, NULL)) < 0) break; switch (c) { @@ -446,6 +449,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, case 'Y': arg_cctl = strdup(optarg); break; + case 'x': + arg_prateshift = strdup(optarg); + break; case 'l': err = atoi(optarg); arg_latency_req = err >= 4 ? err : 4; @@ -627,6 +633,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, logit(LOG_CRIT, "Unable to add ossmixer controls.\n"); exit(EXIT_FAILURE); } + if (arg_prateshift) + play->prateshift_name = arg_prateshift; + #ifdef USE_SAMPLERATE loop->src_enable = arg_samplerate > 0; if (loop->src_enable) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 1dbcefe..7a98ef3 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -127,6 +127,7 @@ struct loopback_handle { snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; snd_ctl_elem_value_t *ctl_channels; + char *prateshift_name; /* ascii name for the playback rate shift ctl elem */ }; struct loopback { diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 845ab82..8658fa6 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1101,7 +1101,8 @@ void update_pitch(struct loopback *loop) #endif } else if (loop->sync == SYNC_TYPE_PLAYRATESHIFT) { - set_rate_shift(loop->play, pitch); + // pitch is capture-based, playback side requires reciprocal + set_rate_shift(loop->play, 1 / pitch); #ifdef USE_SAMPLERATE if (loop->use_samplerate) { loop->src_data.src_ratio = @@ -1172,44 +1173,86 @@ static int get_channels(struct loopback_handle *lhandle) return snd_ctl_elem_value_get_integer(lhandle->ctl_channels, 0); } -static void openctl_elem(struct loopback_handle *lhandle, - int device, int subdevice, - const char *name, - snd_ctl_elem_value_t **elem) +static int openctl_elem_id(struct loopback_handle *lhandle, snd_ctl_elem_id_t *id, + snd_ctl_elem_value_t **elem) { int err; if (snd_ctl_elem_value_malloc(elem) < 0) { *elem = NULL; + snd_ctl_elem_id_free(id); + return -ENOMEM; + } + snd_ctl_elem_value_set_id(*elem, id); + snd_ctl_elem_value_set_interface(*elem, SND_CTL_ELEM_IFACE_PCM); + err = snd_ctl_elem_read(lhandle->ctl, *elem); + if (err < 0) { + snd_ctl_elem_id_free(id); + snd_ctl_elem_value_free(*elem); + *elem = NULL; + return err; } else { - snd_ctl_elem_value_set_interface(*elem, - SND_CTL_ELEM_IFACE_PCM); - snd_ctl_elem_value_set_device(*elem, device); - snd_ctl_elem_value_set_subdevice(*elem, subdevice); - snd_ctl_elem_value_set_name(*elem, name); - err = snd_ctl_elem_read(lhandle->ctl, *elem); - if (err < 0) { - snd_ctl_elem_value_free(*elem); - *elem = NULL; - } else { - if (verbose) - snd_output_printf(lhandle->loopback->output, - "Opened PCM element %s of %s, device %d, subdevice %d\n", - name, snd_ctl_name(lhandle->ctl), device, subdevice); - } + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + snd_ctl_elem_id_get_name(id), snd_ctl_name(lhandle->ctl), + snd_ctl_elem_id_get_device(id), + snd_ctl_elem_id_get_subdevice(id)); + return 0; } } +static int openctl_elem(struct loopback_handle *lhandle, + int device, int subdevice, + const char *name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + if (snd_ctl_elem_id_malloc(&id)) + return -ENOMEM; + snd_ctl_elem_id_set_device(id, device); + snd_ctl_elem_id_set_subdevice(id, subdevice); + snd_ctl_elem_id_set_name(id, name); + return openctl_elem_id(lhandle, id, elem); +} + +static int openctl_elem_ascii(struct loopback_handle *lhandle, char *ascii_name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + if (snd_ctl_elem_id_malloc(&id)) { + return -ENOMEM; + } + if (snd_ctl_ascii_elem_id_parse(id, ascii_name)) { + snd_ctl_elem_id_free(id); + fprintf(stderr, "Wrong control identifier: %s\n", ascii_name); + return -EINVAL; + } + return openctl_elem_id(lhandle, id, elem); +} + static int openctl(struct loopback_handle *lhandle, int device, int subdevice) { int err; lhandle->ctl_rate_shift = NULL; if (lhandle->loopback->play == lhandle) { + // play only + if (lhandle->prateshift_name) { + err = openctl_elem_ascii(lhandle, lhandle->prateshift_name, + &lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Unable to open playback PCM Rate Shift elem '%s'.\n", + lhandle->prateshift_name); + exit(EXIT_FAILURE); + } + } if (lhandle->loopback->controls) goto __events; return 0; } + // capture only openctl_elem(lhandle, device, subdevice, "PCM Notify", &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", -- 2.25.1

2 1

Alsaloop: sync mode PLAYSHIFT + Loopback on playback side
by Pavel Hofman 13 Oct '21

13 Oct '21

Hi, I added support for "Playback Pitch 1000000" ctl elem to UAC2 gadget (not submitted to USB yet) and now I am working on alsaloop support for this ctl elem. The changes are simple (tested to work perfectly, patch to follow), but during the work I hit the following issue with playback Loopback "PCM Rate Shift 100000". If the snd-aloop device is on playback side (i.e. capture from soundcard -> Loopback), the required sync mode is PLAYSHIFT. That means Loopback ctl elem "PCM Rate Shift 100000" should be controlled (by a reciprocal). That is simple by a patch like this: diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 845ab82..619bf35 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1061,7 +1061,13 @@ static int set_rate_shift(struct loopback_handle *lhandle, double pitch) int err; if (lhandle->ctl_rate_shift) { - snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); + long value; + if (lhandle->loopback->play == lhandle) + // playback => reciprocal + value = 1/(pitch) * 100000; + else + value = pitch * 100000; + snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, value); err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); } else if (lhandle->capt_pitch) { snd_ctl_elem_value_set_integer(lhandle->capt_pitch, 0, (1 / pitch) * 1000000); @@ -1205,15 +1211,18 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) int err; lhandle->ctl_rate_shift = NULL; + // both play and capture + openctl_elem(lhandle, device, subdevice, "PCM Notify", + &lhandle->ctl_notify); + openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", + &lhandle->ctl_rate_shift); if (lhandle->loopback->play == lhandle) { + // play only if (lhandle->loopback->controls) goto __events; return 0; } - openctl_elem(lhandle, device, subdevice, "PCM Notify", - &lhandle->ctl_notify); - openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", - &lhandle->ctl_rate_shift); + // capture only openctl_elem(lhandle, device, subdevice, "Capture Pitch 1000000", &lhandle->capt_pitch); set_rate_shift(lhandle, 1); However, IIUC how the Loopback device works, the "PCM Rate Shift 100000" ctl elem applicable to device=0 on playback side is that of the capture side, i.e. for device=1. The patch above would pick the playback-side device=0 ctl elem in pcmjob.c:openctl_elem. Hard-coding the device=0 -> device=1 is possible, but Loopback supports more devices. Please what solution for picking the correct "PCM Rate Shift 100000" ctl elem for the PLAYSHIFT sync mode would you recommend? Thanks a lot, Pavel.

2 6

Re: [PATCH] ASoC: wm8960: Fix clock configuration on slave mode
by S.j. Wang 13 Oct '21

13 Oct '21

> On Tue, Sep 07, 2021 at 05:11:09PM +0800, Shengjiu Wang wrote: > > There is a noise issue for 8kHz sample rate on slave mode. > > Compared with master mode, the difference is the DACDIV setting, after > > correcting the DACDIV, the noise is gone. > > > > There is no noise issue for 48kHz sample rate, because the default > > value of DACDIV is correct for 48kHz. > > > > So wm8960_configure_clocking() should be functional for ADC and DAC > > function even if it is slave mode. > > > > In order to be compatible for old use case, just add condition for > > checking that sysclk is zero with slave mode. > > > > Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its > > bit clock and frame clock") > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/codecs/wm8960.c | 2 +- > > 1 file changed, 1 insertion(+), 1 deletion(-) > > > > diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c > > index 9e621a254392..9c6af76a60fd 100644 > > --- a/sound/soc/codecs/wm8960.c > > +++ b/sound/soc/codecs/wm8960.c > > @@ -742,7 +742,7 @@ static int wm8960_configure_clocking(struct > snd_soc_component *component) > > int i, j, k; > > int ret; > > > > - if (!(iface1 & (1<<6))) { > > + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { > > dev_dbg(component->dev, > > "Codec is slave mode, no need to configure > > clock\n"); > > Looking through the datasheet it just looks like this if statement has always > been non-sense, it looks pretty clear the clocking should still be configured in > slave mode (apart from BCLKDIV which is presumably ignored in slave mode). > > I would be slightly inclined to suggest it would be better to just fixup any > systems not setting sysclk for slave mode, but I am assuming you are talking > primarily about out of tree systems. So I think we need to at least update the > message here as well, it should probably change to a warning and state that > the we are proceeding with no clock configuration, rather than erroneously > saying it doesn't need one. > > Thanks, > Charles Sorry for missing your email. I will update it in v2. Thanks. Best regards Wang Shengjiu

1 0

[RESEND PATCH] ASoC: wm8960: Fix clock configuration on slave mode
by Shengjiu Wang 13 Oct '21

13 Oct '21

There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392..9c6af76a60fd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,7 +742,7 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { dev_dbg(component->dev, "Codec is slave mode, no need to configure clock\n"); return 0; -- 2.17.1

3 2

[PATCH] ALSA: usb-audio: Add quirk for VF0770
by Jonas Hahnfeld 12 Oct '21

12 Oct '21

The device advertises 8 formats, but only a rate of 48kHz is honored by the hardware and 24 bits give chopped audio, so only report the one working combination. This fixes out-of-the-box audio experience with PipeWire which otherwise attempts to choose S24_3LE (while PulseAudio defaulted to S16_LE). Signed-off-by: Jonas Hahnfeld <hahnjo(a)hahnjo.de> --- sound/usb/quirks-table.h | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e03043f7dad3..be5c2bc26a1d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -77,6 +77,45 @@ /* E-Mu 0204 USB */ { USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) }, +/* + * Creative Technology, Ltd Live! Cam Sync HD [VF0770] + * The device advertises 8 formats, but only a rate of 48kHz is honored by the + * hardware and 24 bits give chopped audio, so only report the one working + * combination. + */ +{ + USB_DEVICE(0x041e, 0x4095), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .fmt_bits = 16, + .iface = 3, + .altsetting = 4, + .altset_idx = 4, + .endpoint = 0x82, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 }, + }, + }, + }, + }, +}, + /* * HP Wireless Audio * When not ignored, causes instability issues for some users, forcing them to -- 2.33.0

2 5

[PATCH v2 0/3] ASoC: SOF: Intel: power optimizations with HDaudio SPIB register
by Pierre-Louis Bossart 12 Oct '21

12 Oct '21

This patchset was initially provided in a larger series that was split in two [1]. This part only provides support for the SPIB register support, added on Intel platforms since Skylake (2015). The use of the SPIB register helps reduce power consumption - though to a smaller degree than DMI_L1. This hardware capability is however incompatible with userspace-initiated rewinds typically used by PulseAudio. In the past (2015..2017) Intel suggested an API extension to let applications disable rewinds. At the time the feedback was that such a capability was too Intel-specific and SPIB remained unused except for loading DSP code. We now see devices with smaller batteries being released, and it's time to revisit Linux support for SPIB to extend battery life. In this update the rewinds are disabled via an opt-in kernel parameter. In the previous reviews, there was consensus that a Kconfig option was too complicated for distributions to set, and we are missing a TBD API to expose such capabilities to user-space. The debate on whether or not to use rewinds, and the impact of disabling rewinds, will likely be closed when Intel releases the 'deep-buffer' support, currently under development [2][3]. With this solution, rewinds will not be needed, ever. When an application deals with content that is not latency-sensitive (e.g. music playback), it will be able to reduce power consumption by selecting a different PCM device with increased buffering capabilities. Low-latency streams will be handled by the 'regular' path. In other words, the impossible compromise between power and latency will be handled with different PCM devices/profiles for the same endpoint, and we can push the design of capability negotiation to a later time when all the building blocks (firmware topology, kernel, userspace) are ready - we still have firmware xruns, DPCM race conditions to solve, and a need to describe these alternate PCM devices with UCM using 'modifiers'. [1] https://lore.kernel.org/alsa-devel/20210610205326.1176400-1-pierre-louis.bo… [2] https://github.com/thesofproject/linux/pull/3146 [3] https://github.com/thesofproject/sof/pull/4611 Pierre-Louis Bossart (1): ALSA: pcm: introduce INFO_NO_REWINDS flag Ranjani Sridharan (2): ASOC: SOF: pcm: add .ack callback support ASoC: SOF: Intel: add .ack support for HDaudio platforms include/uapi/sound/asound.h | 2 +- sound/core/pcm_native.c | 2 ++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-pcm.c | 41 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda-stream.c | 2 ++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 1 + sound/soc/sof/ops.h | 10 ++++++++ sound/soc/sof/pcm.c | 9 +++++++ sound/soc/sof/sof-priv.h | 3 +++ 11 files changed, 70 insertions(+), 3 deletions(-) -- 2.25.1

2 14

[PATCH] ALSA: hda: avoid write to STATESTS if controller is in reset
by Kai Vehmanen 12 Oct '21

12 Oct '21

The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 062da7a7a586..f7bd6e2db085 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -421,8 +421,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); base-commit: 0fc27129ed268b619135b99d365ce44daaf85602 -- 2.33.0

2 3

[RFC PATCH v2 0/5] ASoC: soc-pcm: fix trigger race conditions with shared BE
by Pierre-Louis Bossart 12 Oct '21

12 Oct '21

We've been adding a 'deep buffer' PCM device to several SOF topologies in order to reduce power consumption. The typical use-case would be music playback over a headset: this additional PCM device provides more buffering and longer latencies, leaving the rest of the system sleep for longer periods. Notifications and 'regular' low-latency audio playback would still use the 'normal' PCM device and be mixed with the 'deep buffer' before rendering on the headphone endpoint. The tentative direction would be to expose this alternate device to PulseAudio/PipeWire/CRAS via the UCM SectionModifier definitions. That seemed a straightforward topology change until our automated validation stress tests started reporting issues on SoundWire platforms, when e.g. two START triggers might be send and conversely the STOP trigger is never sent. The SoundWire stream state management flagged inconsistent states when the two 'normal' and 'deep buffer' devices are used concurrently with rapid play/stop/pause monkey testing. Looking at the soc-pcm.c code, it seems that the BE state management needs a lot of love. a) there is no consistent protection for the BE state. In some parts of the code, the state updates are protected by a spinlock but in the trigger they are not. When we open/play/close the two PCM devices in stress tests, we end-up testing a state that is being modified. That can't be good. b) there is a conceptual deadlock: on stop we check the FE states to see if a shared BE can be stopped, but since we trigger the BE first the FE states have not been modified yet, so the TRIGGER_STOP is never sent. This patchset suggests a transition from a spinlock to a mutex, an extended protection when walking through the BE list, and the use of a refcount to decide when to trigger the BE. With these patches I am able to run our entire validation suite without any issues with this new 'deep buffer' topology, and no regressions on existing solutions [1] One might ask 'how come we didn't see this earlier'? The answer is probably that the .trigger callbacks in most implementations seems to perform DAPM operations, and sending the triggers multiple times is not an issue. In the case of SoundWire, we do use the .trigger callback to reconfigure the bus using the 'bank switch' mechanism. It could be acceptable to tolerate a trigger multiple times, but the deadlock on stop cannot be fixed at the SoundWire layer alone. I chose to send this patchset as an RFCv2 to gather more feedback and make use others know about DPCM issues. We're going to spend more time on this but if others can provide feedback/test results it would be greatly appreciated. Opens: 1) is this the right solution? The DPCM code is far from simple, has notions such as SND_SOC_DPCM_UPDATE_NO and 'trigger_pending' that I have no background on. 2) There are other reports of kernel oopses [2] that seem related to the lack of protection. I'd be good to confirm if this patchset solve these problems as well. [1] https://github.com/thesofproject/linux/pull/3146 [2] https://lore.kernel.org/alsa-devel/002f01d7b4f5$c030f4a0$4092dde0$@samsung.… changes since RFC v1: Removed unused function Removed exported symbols only used in soc-pcm.c, used static instead Use a mutex instead of a spinlock Protect all for_each_dpcm_be() loops Fix bugs introduced in the refcount Pierre-Louis Bossart (5): ASoC: soc-pcm: remove snd_soc_dpcm_fe_can_update() ASoC: soc-pcm: don't export local functions, use static ASoC: soc-pcm: replace dpcm_lock with dpcm_mutex ASoC: soc-pcm: protect for_each_dpcm_be() loops with dpcm_mutex ASoC: soc-pcm: test refcount before triggering include/sound/soc-dpcm.h | 17 +---- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 153 ++++++++++++++++++++++++++------------- 4 files changed, 108 insertions(+), 66 deletions(-) -- 2.25.1

3 27

[PATCH v2 0/9] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 12 Oct '21

12 Oct '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. This patch set depends on: -- https://patchwork.kernel.org/project/alsa-devel/list/?series=549577 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=548765 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Chnages Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (9): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoc: qcom: lpass: Add suspend and resume for sc7280 platform ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 209 +++++++ sound/soc/qcom/lpass-cpu.c | 274 +++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 497 +++++++++++++++-- sound/soc/qcom/lpass-sc7280.c | 602 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 156 ++++++ 9 files changed, 1870 insertions(+), 57 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 12

[PATCH] ALSA: hda/realtek: Fix the mic type detection issue for ASUS G551JW
by Hui Wang 12 Oct '21

12 Oct '21

We need to define the codec pin 0x1b to be the mic, but somehow the mic doesn't support hot plugging detection, and Windows also has this issue, so we set it to phantom headset-mic. Also the determine_headset_type() often returns the omtp type by a mistake when we plug a ctia headset, this makes the mic can't record sound at all. Because most of the headset are ctia type nowadays and some machines have the fixed ctia type audio jack, it is possible this machine has the fixed ctia jack too. Here we set this mic jack to fixed ctia type, this could avoid the mic type detection mistake and make the ctia headset work stable. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214537 Reported-and-tested-by: msd <msd.mmq(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bca5830ff706..22d27b12c4e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10197,6 +10197,9 @@ enum { ALC671_FIXUP_HP_HEADSET_MIC2, ALC662_FIXUP_ACER_X2660G_HEADSET_MODE, ALC662_FIXUP_ACER_NITRO_HEADSET_MODE, + ALC668_FIXUP_ASUS_NO_HEADSET_MIC, + ALC668_FIXUP_HEADSET_MIC, + ALC668_FIXUP_MIC_DET_COEF, }; static const struct hda_fixup alc662_fixups[] = { @@ -10580,6 +10583,29 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_USI_FUNC }, + [ALC668_FIXUP_ASUS_NO_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x04a1112c }, + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MIC + }, + [ALC668_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + .chained = true, + .chain_id = ALC668_FIXUP_MIC_DET_COEF + }, + [ALC668_FIXUP_MIC_DET_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x15 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0d60 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -10615,6 +10641,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x185d, "ASUS G551JW", ALC668_FIXUP_ASUS_NO_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), -- 2.25.1

2 1

[PATCH] ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets
by Richard Fitzgerald 12 Oct '21

12 Oct '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Ensure the default 0dB playback path is always used. The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was spurious, and resulted in a -6dB attenuation being accidentally inserted into the playback path. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d484ec09eb2e..8de23e4732b3 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -922,7 +922,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +992,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.11.0

2 1

[PATCH 0/8] arm: ep93xx: CCF conversion
by Nikita Shubin 12 Oct '21

12 Oct '21

This series series of patches converts ep93xx to Common Clock Framework. It consists of preparation patches to use clk_prepare_enable where it is needed, instead of clk_enable used in ep93xx drivers prior to CCF and a patch converting mach-ep93xx/clock.c to CCF. Link: https://lore.kernel.org/patchwork/cover/1445563/ Link: https://lore.kernel.org/patchwork/patch/1435884/ Alexander Sverdlin (7): iio: ep93xx: Prepare clock before using it spi: spi-ep93xx: Prepare clock before using it Input: ep93xx_keypad: Prepare clock before using it video: ep93xx: Prepare clock before using it dmaengine: ep93xx: Prepare clock before using it ASoC: cirrus: i2s: Prepare clock before using it pwm: ep93xx: Prepare clock before using it Nikita Shubin (1): ep93xx: clock: convert in-place to COMMON_CLK arch/arm/Kconfig | 2 +- arch/arm/mach-ep93xx/clock.c | 975 ++++++++++++++----------- arch/arm/mach-ep93xx/core.c | 2 +- arch/arm/mach-ep93xx/soc.h | 42 +- drivers/dma/ep93xx_dma.c | 6 +- drivers/iio/adc/ep93xx_adc.c | 6 +- drivers/input/keyboard/ep93xx_keypad.c | 4 +- drivers/pwm/pwm-ep93xx.c | 12 +- drivers/spi/spi-ep93xx.c | 4 +- drivers/video/fbdev/ep93xx-fb.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 12 +- 11 files changed, 605 insertions(+), 464 deletions(-) base-commit: 64376a981a0e2e57c46efa63197c2ebb7dab35df -- 2.26.2

4 9

[PATCH v3 0/5] Update Lpass digital codec macro drivers
by Srinivasa Rao Mandadapu 12 Oct '21

12 Oct '21

This patch set is to add support for lpass sc7280 based targets. Upadate compatible name and change of bulk clock voting to optional clock voting in digital codecs va, rx, tx macro drivers. Changes Since V2: -- Add Tx macro deafults for lpass sc7280 Changes Since V1: -- Removed individual clock voting and used bulk clock optional. -- Removed volatile changes and fixed default values. -- Typo errors. Srinivasa Rao Mandadapu (5): ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Enable tx top soundwire mic clock ASoC: codecs: tx-macro: Update tx default values ASoC: codecs: Change bulk clock voting to optional voting in digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +++- sound/soc/codecs/lpass-rx-macro.c | 3 ++- sound/soc/codecs/lpass-tx-macro.c | 25 +++++++++++++++++++--- sound/soc/codecs/lpass-va-macro.c | 3 ++- sound/soc/codecs/lpass-wsa-macro.c | 1 + 8 files changed, 39 insertions(+), 9 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 10

Add ALSA UCM support for Pinebook Pro's audio
by Jian-Hong Pan 12 Oct '21

12 Oct '21

Hi, I prepared the pull request "ucm2: Add UCM support for rockchip_es8316 on Pinebook Pro" [1] to enable Pinebook Pro's audio generically. Also, the corresponding validator pull request "ucm-validator: Upload rockchip_es8316's dummy json and alsa-info" [2]. The original information comes from Manjaro ARM's pinebookpro-post-install's asound.state [3] and pinebookpro-audio's audio_jack_plugged_in.sh [4]. To make it available on most distros, I transfer the asound.state and the shell script into ALSA UCM conf. It will be appreciated if someone can review the pull requests and give some feedback. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/112 [2]: https://github.com/alsa-project/alsa-tests/pull/22 [3]: https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-post-… [4]: https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-audio… Jian-Hong Pan

3 2

[PATCH v2 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 12 Oct '21

12 Oct '21

Changes since v1: - Fixes Kconfig indent as per kernel coding style guidelines. - Updated _GPL suffix for all exported symbol. - As suggested added separate machine ops for each codecs in machine driver - Add sound card support for newer codecs Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 50 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 340 +++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 596 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 315 ++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 131 ++++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1988 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

2 11

[PATCH] ASoC: simple-card: Fill in driver name
by Guido Günther 12 Oct '21

12 Oct '21

alsa-ucm groups by driver name so fill that in as well. Otherwise the presented information is redundant and doesn't reflect the used driver. We can't just use 'asoc-simple-card' since the driver name is restricted to 15 characters. Before: # cat /proc/asound/cards 0 [Devkit ]: Librem_5_Devkit - Librem 5 Devkit Librem 5 Devkit After: 0 [Devkit ]: simple-card - Librem 5 Devkit Librem 5 Devkit Signed-off-by: Guido Günther <agx(a)sigxcpu.org> --- This came out of a discussion about adding alsa-ucm profiles for the Librem 5 Devkit at https://github.com/alsa-project/alsa-ucm-conf/pull/102 sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0015f534d42d..a3a7990b5cb6 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -621,6 +621,7 @@ static int asoc_simple_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->probe = simple_soc_probe; + card->driver_name = "simple-card"; li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); if (!li) -- 2.30.2

5 8

[PATCH v3 0/2] ASoC: rt9120: Add Richtek RT9120 supprot
by cy_huang 12 Oct '21

12 Oct '21

From: ChiYuan Huang <cy_huang(a)richtek.com> This patch series Add the Richtek RT9120 support. In v3: - Add dvdd regulator binding to check the dvdd voltage domain. - Refine sdo_select_text. - Use switch case in 'internal_power_event' function. - Remove the volume and mute initially write in component probe. - Remove the mute API. It's no need by HW design. In v2: - Add missing #sound-dai-cells property. ChiYuan Huang (2): ASoC: dt-bindings: rt9120: Add initial bindings ASoC: rt9120: Add rt9210 audio amplifier support .../devicetree/bindings/sound/richtek,rt9120.yaml | 59 +++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt9120.c | 489 +++++++++++++++++++++ 4 files changed, 560 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml create mode 100644 sound/soc/codecs/rt9120.c -- 2.7.4

3 5

[PATCH] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
by Yang Yingliang 11 Oct '21

11 Oct '21

'component' is allocated in snd_soc_register_component(), but component->list is not initalized, this may cause snd_soc_del_component_unlocked() deref null ptr in the error handing case. KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007] RIP: 0010:__list_del_entry_valid+0x81/0xf0 Call Trace: snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core] snd_soc_add_component.cold+0x54/0x6c [snd_soc_core] snd_soc_register_component+0x70/0x90 [snd_soc_core] devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core] tas2552_probe+0x265/0x320 [snd_soc_tas2552] ? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552] i2c_device_probe+0xa31/0xbe0 Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize(). Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c830e96afba2..80ca260595fd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,6 +2599,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); -- 2.25.1

2 1

[PATCH v2 0/4] ASoC: Intel: bytcr_rt5640: few cleanups
by Andy Shevchenko 11 Oct '21

11 Oct '21

The small set of cleanups against bytcr_rt5640 board file. In v2: - added commit message to patch 2 (Joe, Pierre) - added cover letter (Pierre) - added Hans to Cc list (Hans) Andy Shevchenko (4): ASoC: Intel: bytcr_rt5640: Get platform data via dev_get_platdata() ASoC: Intel: bytcr_rt5640: Use temporary variable for struct device ASoC: Intel: bytcr_rt5640: use devm_clk_get_optional() for mclk ASoC: Intel: bytcr_rt5640: Utilize dev_err_probe() to avoid log saturation sound/soc/intel/boards/bytcr_rt5640.c | 116 +++++++++++--------------- 1 file changed, 49 insertions(+), 67 deletions(-) -- 2.33.0

4 7

[PATCH] ASoC: amd: Kconfig: Select fch clock support with machine driver
by Ajit Kumar Pandey 11 Oct '21

11 Oct '21

We are using fch clock controller as parent mclk source for rt5682 codec. Add config to enable clock framework support for 48MHz fixed clock when machine driver config is selected. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/amd/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 49ff5e73e9ba..a2cb50d09980 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -6,6 +6,7 @@ config SND_SOC_AMD_ACP config SND_SOC_AMD_CZ_DA7219MX98357_MACH tristate "AMD CZ support for DA7219, RT5682 and MAX9835" + select CLK_FIXED_FCH select SND_SOC_DA7219 select SND_SOC_RT5682_I2C select SND_SOC_MAX98357A @@ -30,6 +31,7 @@ config SND_SOC_AMD_ACP3x config SND_SOC_AMD_RV_RT5682_MACH tristate "AMD RV support for RT5682" + select CLK_FIXED_FCH select SND_SOC_RT5682_I2C select SND_SOC_MAX98357A select SND_SOC_CROS_EC_CODEC -- 2.25.1

2 1

[PATCH v2 0/4] ASoC: Intel: bytcr_rt5651: few cleanups
by Andy Shevchenko 11 Oct '21

11 Oct '21

The small set of cleanups against bytcr_rt5651 board file. In v2: - added commit message to patch 2 (Joe, Pierre) - added cover letter (Pierre) - added Hans to Cc list (Hans) Andy Shevchenko (4): ASoC: Intel: bytcr_rt5651: Get platform data via dev_get_platdata() ASoC: Intel: bytcr_rt5651: Use temporary variable for struct device ASoC: Intel: bytcr_rt5651: use devm_clk_get_optional() for mclk ASoC: Intel: bytcr_rt5651: Utilize dev_err_probe() to avoid log saturation sound/soc/intel/boards/bytcr_rt5651.c | 118 +++++++++++--------------- 1 file changed, 50 insertions(+), 68 deletions(-) -- 2.33.0

4 7

[PATCH] sh: Use modern ASoC DAI format terminology
by Mark Brown 11 Oct '21

11 Oct '21

The SH machine drivers have some ASoC DAI format specifications that use older defines based on outdated terminology which we're trying to retire, update to the new bindings. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- arch/sh/boards/mach-ecovec24/setup.c | 2 +- arch/sh/boards/mach-se/7724/setup.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index bab91a99124e..97c5703b1818 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -886,7 +886,7 @@ static struct asoc_simple_card_info fsi_da7210_info = { .card = "FSIB-DA7210", .codec = "da7210.0-001a", .platform = "sh_fsi.0", - .daifmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, + .daifmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBP_CFP, .cpu_dai = { .name = "fsib-dai", }, diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 8d6541ba0186..d9b31d4560c0 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -305,7 +305,7 @@ static struct asoc_simple_card_info fsi_ak4642_info = { .card = "FSIA-AK4642", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM, + .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBP_CFP, .cpu_dai = { .name = "fsia-dai", }, -- 2.20.1

1 1

[PATCH] ALSA: pcm: Workaround for a wrong offset in SYNC_PTR compat ioctl
by Takashi Iwai 11 Oct '21

11 Oct '21

Michael Forney reported an incorrect padding type that was defined in the commit 80fe7430c708 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control") for PCM control mmap data. His analysis is correct, and this caused the misplacements of PCM control data on 32bit arch and 32bit compat mode. The bug is that the __pad2 definition in __snd_pcm_mmap_control64 struct was wrongly with __pad_before_uframe, which should have been __pad_after_uframe instead. This struct is used in SYNC_PTR ioctl and control mmap. Basically this bug leads to two problems: - The offset of avail_min field becomes wrong, it's placed right after appl_ptr without padding on little-endian - When appl_ptr and avail_min are read as 64bit values in kernel side, the values become either zero or corrupted (mixed up) One good news is that, because both user-space and kernel misunderstand the wrong offset, at least, 32bit application running on 32bit kernel works as is. Also, 64bit applications are unaffected because the padding size is zero. The remaining problem is the 32bit compat mode; as mentioned in the above, avail_min is placed right after appl_ptr on little-endian archs, 64bit kernel reads bogus values for appl_ptr updates, which may lead to streaming bugs like jumping, XRUN or whatever unexpected. (However, we haven't heard any serious bug reports due to this over years, so practically seen, it's fairly safe to assume that the impact by this bug is limited.) Ideally speaking, we should correct the wrong mmap status control definition. But this would cause again incompatibility with the existing binaries, and fixing it (e.g. by renumbering ioctls) would be really messy. So, as of this patch, we only correct the behavior of 32bit compat mode and keep the rest as is. Namely, the SYNC_PTR ioctl is now handled differently in compat mode to read/write the 32bit values at the right offsets. The control mmap of 32bit apps on 64bit kernels has been already disabled (which is likely rather an overlook, but this worked fine at this time :), so covering SYNC_PTR ioctl should suffice as a fallback. Fixes: 80fe7430c708 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control") Reported-by: Michael Forney <mforney(a)mforney.org> Cc: <stable(a)vger.kernel.org> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Rich Felker <dalias(a)libc.org> Link: https://lore.kernel.org/r/29QBMJU8DE71E.2YZSH8IHT5HMH@mforney.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_compat.c | 72 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index a59de24695ec..dfe5a64e19d2 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -468,6 +468,76 @@ static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream, } #endif /* CONFIG_X86_X32 */ +#ifdef __BIG_ENDIAN +typedef char __pad_before_u32[4]; +typedef char __pad_after_u32[0]; +#else +typedef char __pad_before_u32[0]; +typedef char __pad_after_u32[4]; +#endif + +/* PCM 2.0.15 API definition had a bug in mmap control; it puts the avail_min + * at the wrong offset due to a typo in padding type. + * The bug hits only 32bit. + * A workaround for incorrect read/write is needed only in 32bit compat mode. + */ +struct __snd_pcm_mmap_control64_buggy { + __pad_before_u32 __pad1; + __u32 appl_ptr; + __pad_before_u32 __pad2; /* SiC! here is the bug */ + __pad_before_u32 __pad3; + __u32 avail_min; + __pad_after_uframe __pad4; +}; + +static int snd_pcm_ioctl_sync_ptr_buggy(struct snd_pcm_substream *substream, + struct snd_pcm_sync_ptr __user *_sync_ptr) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_sync_ptr sync_ptr; + struct __snd_pcm_mmap_control64_buggy *sync_cp; + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + int err; + + memset(&sync_ptr, 0, sizeof(sync_ptr)); + sync_cp = (struct __snd_pcm_mmap_control64_buggy *)&sync_ptr.c.control; + if (get_user(sync_ptr.flags, (unsigned __user *)&(_sync_ptr->flags))) + return -EFAULT; + if (copy_from_user(sync_cp, &(_sync_ptr->c.control), sizeof(*sync_cp))) + return -EFAULT; + status = runtime->status; + control = runtime->control; + if (sync_ptr.flags & SNDRV_PCM_SYNC_PTR_HWSYNC) { + err = snd_pcm_hwsync(substream); + if (err < 0) + return err; + } + snd_pcm_stream_lock_irq(substream); + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, sync_cp->appl_ptr); + if (err < 0) { + snd_pcm_stream_unlock_irq(substream); + return err; + } + } else { + sync_cp->appl_ptr = control->appl_ptr; + } + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = sync_cp->avail_min; + else + sync_cp->avail_min = control->avail_min; + sync_ptr.s.status.state = status->state; + sync_ptr.s.status.hw_ptr = status->hw_ptr; + sync_ptr.s.status.tstamp = status->tstamp; + sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; + snd_pcm_stream_unlock_irq(substream); + if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) + return -EFAULT; + return 0; +} + /* */ enum { @@ -537,7 +607,7 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l if (in_x32_syscall()) return snd_pcm_ioctl_sync_ptr_x32(substream, argp); #endif /* CONFIG_X86_X32 */ - return snd_pcm_common_ioctl(file, substream, cmd, argp); + return snd_pcm_ioctl_sync_ptr_buggy(substream, argp); case SNDRV_PCM_IOCTL_HW_REFINE32: return snd_pcm_ioctl_hw_params_compat(substream, 1, argp); case SNDRV_PCM_IOCTL_HW_PARAMS32: -- 2.31.1

2 3

[PATCH] ASoC: rt1011: Fix 'I2S Reference' enum control caused error
by Peter Ujfalusi 11 Oct '21

11 Oct '21

Access to 'Is@ Reference' enum causes alsamixer to fail to load: $ alsamixer cannot load mixer controls: Invalid argument cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow The reason is that the original patch adding the code was using ucontrol->value.integer.value[0] instead the correct ucontrol->value.enumerated.item[0] for an ENUM control. Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/codecs/rt1011.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 508597866dff..3adaff6f0141 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1325,7 +1325,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.integer.value[0]; + int i2s_ref_ch = ucontrol->value.enumerated.item[0]; switch (i2s_ref_ch) { case RT1011_I2S_REF_LEFT_CH: @@ -1344,7 +1344,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, dev_info(component->dev, "I2S Reference: Do nothing\n"); } - rt1011->i2s_ref = ucontrol->value.integer.value[0]; + rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; return 0; } @@ -1357,7 +1357,7 @@ static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - ucontrol->value.integer.value[0] = rt1011->i2s_ref; + ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; return 0; } -- 2.33.0

1 1

[PATCH v2] ASoC: rt1011: Fix 'I2S Reference' enum control caused error
by Peter Ujfalusi 11 Oct '21

11 Oct '21

Access to 'Is@ Reference' enum causes alsamixer to fail to load: $ alsamixer cannot load mixer controls: Invalid argument cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow The reason is that the original patch adding the code was using ucontrol->value.integer.value[0] instead the correct ucontrol->value.enumerated.item[0] for an ENUM control. Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v1: - Correct the ENUM declaration as well Regards, Peter sound/soc/codecs/rt1011.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 508597866dff..bdfcbb81fa19 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,12 +1311,11 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref[] = { +static const char * const rt1011_i2s_ref_texts[] = { "None", "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, - rt1011_i2s_ref); +static SOC_ENUM_SINGLE_EXT_DECL(rt1011_i2s_ref_enum, rt1011_i2s_ref_texts); static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1325,7 +1324,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.integer.value[0]; + int i2s_ref_ch = ucontrol->value.enumerated.item[0]; switch (i2s_ref_ch) { case RT1011_I2S_REF_LEFT_CH: @@ -1344,7 +1343,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, dev_info(component->dev, "I2S Reference: Do nothing\n"); } - rt1011->i2s_ref = ucontrol->value.integer.value[0]; + rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; return 0; } @@ -1357,7 +1356,7 @@ static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - ucontrol->value.integer.value[0] = rt1011->i2s_ref; + ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; return 0; } -- 2.33.0

1 0

Re: Fwd: [PATCH 4/8] ASoC: amd: acp: Add generic machine driver support for ACP cards
by Ajit Kumar Pandey 11 Oct '21

11 Oct '21

Apologies for corrupted reply format, some issue with thunderbird. We will update all below review comments in v2 patch series > On Thu, Sep 30, 2021 at 06:54:14PM +0530, Ajit Kumar Pandey wrote: > > A couple of things here, most of these are probably fine for now > other than the EXPORT_SYMBOL but I think you're likely to run > into issues going forward and need to refactor. > >> + switch (drvdata->hs_codec_id) { >> + case RT5682: >> + pll_id = RT5682_PLL2; >> + pll_src = RT5682_PLL2_S_MCLK; >> + freq_in = PCO_PLAT_CLK; >> + freq_out = RT5682_PLL_FREQ; >> + clk_id = RT5682_SCLK_S_PLL2; >> + clk_freq = RT5682_PLL_FREQ; >> + wclk_name = "rt5682-dai-wclk"; >> + bclk_name = "rt5682-dai-bclk"; >> + drvdata->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF >> + | SND_SOC_DAIFMT_CBP_CFP; >> + snd_soc_dapm_add_routes(&rtd->card->dapm, rt5682_map, ARRAY_SIZE(rt5682_map)); >> + break; > > It feels like this is going to run into scaling issues going > forward and you're likely to need separate operations for > different CODECs rather than just different IDs. Similar issues > apply for the amps, it feels like you want to be passing separate > ops in rather than having these switch statements. Ok will change to use separate ops in v2 patch series instead of switch statements. > >> + /* Do nothing for dummy codec */ >> + if (!drvdata->hs_codec_id && drvdata->amp_codec_id) >> + return; > > Wha the test seems to say is do nothing if there's no CODEC but > there is an amp... > >> + >> + clk_disable_unprepare(drvdata->wclk); >> +} > > ...though I'd expect that given that the clock API accepts NULL > clocks you could just remove these checks and unconditionally use > the clocks. Ok will remove this check in v2 patch series. > >> + return 0; >> +} >> +EXPORT_SYMBOL_NS(acp_legacy_dai_links_create, SND_SOC_AMD_MACH); > > EXPORT_SYMBOL_GPL_NS() - ASoC is all EXPORT_SYMBOL_GPL > sure will update to use NS_GPL() in v2 patch chain

1 0

When will we have a new release?
by GitHub issues - opened 11 Oct '21

11 Oct '21

alsa-project/alsa-lib issue #185 was opened from leiserfg: There was an issue in the last release that affects pipewire while using laptops with `Comet Lake PCH` that was solved in alsa after the release. I'm currently using `master` via AUR but will love to be able to use a release version (so I don't have to re-build with every change on master) Issue URL : https://github.com/alsa-project/alsa-lib/issues/185 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/13] Add Yellow Carp platform ASoC driver
by Vijendar Mukunda 11 Oct '21

11 Oct '21

Yellow Carp platform is new APU series based on acp6.x design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Yellow Carp APU with DMIC endpoint support. Vijendar Mukunda (13): ASoC: amd: add Yellow Carp ACP6x IP register header ASoC: amd: add Yellow Carp ACP PCI driver ASoC: amd: add acp6x init/de-init functions ASoC: amd: add platform devices for acp6x pdm driver and dmic driver ASoC: amd: add acp6x pdm platform driver ASoC: amd: add acp6x irq handler ASoC: amd: add acp6x pdm driver dma ops ASoC: amd: add acp6x pci driver pm ops ASoC: amd: add acp6x pdm driver pm ops ASoC: amd: enable Yellow carp acp6x drivers build ASoC: amd: create platform device for acp6x machine driver ASoC: amd: add YC machine driver using dmic ASoC: amd: enable yc machine driver build sound/soc/amd/Kconfig | 13 + sound/soc/amd/Makefile | 1 + sound/soc/amd/yc/Makefile | 9 + sound/soc/amd/yc/acp6x-mach.c | 76 ++++ sound/soc/amd/yc/acp6x-pdm-dma.c | 447 ++++++++++++++++++++++ sound/soc/amd/yc/acp6x.h | 107 ++++++ sound/soc/amd/yc/acp6x_chip_offset_byte.h | 444 +++++++++++++++++++++ sound/soc/amd/yc/pci-acp6x.c | 343 +++++++++++++++++ 8 files changed, 1440 insertions(+) create mode 100644 sound/soc/amd/yc/Makefile create mode 100644 sound/soc/amd/yc/acp6x-mach.c create mode 100644 sound/soc/amd/yc/acp6x-pdm-dma.c create mode 100644 sound/soc/amd/yc/acp6x.h create mode 100644 sound/soc/amd/yc/acp6x_chip_offset_byte.h create mode 100644 sound/soc/amd/yc/pci-acp6x.c -- 2.25.1

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[PATCH] ALSA: usb-audio: Less restriction for low-latency playback mode
by Takashi Iwai 11 Oct '21

11 Oct '21

The recent support for the improved low-latency playback mode applied the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this was a slight overkill. The use of the flag above disables effectively both PCM status and control mmaps, while basically what we want to track is only about the appl_ptr update. For less restriction, use a more proper flag, SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control mmap. Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3bb095a3c9b6..95ec8eec1bb0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1102,7 +1102,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream) /* need an explicit sync to catch applptr update in low-latency mode */ if (direction == SNDRV_PCM_STREAM_PLAYBACK && as->chip->lowlatency) - runtime->hw.info |= SNDRV_PCM_INFO_EXPLICIT_SYNC; + runtime->hw.info |= SNDRV_PCM_INFO_SYNC_APPLPTR; runtime->private_data = subs; subs->pcm_substream = substream; /* runtime PM is also done there */ -- 2.31.1

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