January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: simple-card: add asoc_simple_card_get_conversion()" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card: add asoc_simple_card_get_conversion() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7e5e1f8bbaa82e6877d8bc121cb1b44cb1ce7ddf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:42 +0900 Subject: [PATCH] ASoC: simple-card: add asoc_simple_card_get_conversion() simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_simple_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001cf9cd1..52048069b25a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,6 +165,21 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_simple_card_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); + + asoc_simple_card_parse_convert(dev, top, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, NULL, adata); + asoc_simple_card_parse_convert(dev, np, NULL, adata); + + of_node_put(node); +} + static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *node, struct device_node *np, @@ -260,9 +275,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, "prefix"); } - asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); - asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + asoc_simple_card_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) -- 2.20.1

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[alsa-devel] Applied "ASoC: simple-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: simple-card: add 1 CPU : 1 Codec support again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7adee60ee2732f23f703ff83ee35caad561490ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:47 +0900 Subject: [PATCH] ASoC: simple-card: add 1 CPU : 1 Codec support again simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 52048069b25a..b15651409c7f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -453,6 +453,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *node; struct device_node *np; struct device_node *codec; + struct asoc_simple_card_data adata; bool is_fe; int ret, loop; int dai_idx, link_idx, conf_idx; @@ -480,8 +481,13 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + /* DPCM */ - if (of_get_child_count(node) > 2) { + if (of_get_child_count(node) > 2 || + adata.convert_rate || adata.convert_channels) { for_each_child_of_node(node, np) { codec = of_get_child_by_name(node, loop ? "codec" : @@ -495,14 +501,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) top, node, np, codec, priv, &dai_idx, link_idx++, &conf_idx, is_fe, !loop); + if (ret < 0) + return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, &dai_idx, link_idx++, !loop); + if (ret < 0) + return ret; } - if (ret < 0) - return ret; node = of_get_next_child(top, node); } while (loop && node); @@ -523,6 +531,8 @@ static void asoc_simple_card_get_dais_count(struct device *dev, { struct device_node *top = dev->of_node; struct device_node *node; + struct device_node *np; + struct asoc_simple_card_data adata; int loop; int num; @@ -562,6 +572,15 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ if (!top) { (*link_num) = 1; @@ -578,9 +597,14 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + num = of_get_child_count(node); (*dais_num) += num; - if (num > 2) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { (*link_num) += num; (*ccnf_num)++; } else { -- 2.20.1

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[alsa-devel] Applied "ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8c3590de0a378c2449fc1aec127cc693632458e4 Mon Sep 17 00:00:00 2001 From: Yizhuo <yzhai003(a)ucr.edu> Date: Thu, 3 Jan 2019 13:59:12 -0800 Subject: [PATCH] ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized Inside function rt274_i2c_probe(), if regmap_read() function returns -EINVAL, then local variable "val" leaves uninitialized but used in if statement. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt274.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0ef966d56bac..e2855ab9a2c6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); -- 2.20.1

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[alsa-devel] [PATCH] Variable "val" in function rt274_i2c_probe() could be uninitialized
by Yizhuo 04 Jan '19

04 Jan '19

Inside function rt274_i2c_probe(), if regmap_read() function returns -EINVAL, then local variable "val" leaves uninitialized but used in if statement. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/soc/codecs/rt274.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 8f92e5c4dd9d..cd048df76232 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); -- 2.17.1

2 1

Re: [alsa-devel] [PATCH v5 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Rafael J. Wysocki 04 Jan '19

04 Jan '19

On Thu, Jan 3, 2019 at 5:08 AM Chandan Rajendra <chandan(a)linux.ibm.com> wrote: > > On Thursday, January 3, 2019 5:20:27 AM IST Pierre-Louis Bossart wrote: > > > > On 1/2/19 4:58 PM, Sinan Kaya wrote: > > > On Wed, Jan 2, 2019 at 10:50 PM Pierre-Louis Bossart > > > <pierre-louis.bossart(a)linux.intel.com> wrote: > > >> > > >>> This is pointing to a kconfig issue on ia64 arch. > > >>> > > >>> arch/ia64/Kconfig:128:error: recursive dependency detected! > > >>> arch/ia64/Kconfig:128: choice <choice> contains symbol IA64_HP_SIM > > >>> arch/ia64/Kconfig:202: symbol IA64_HP_SIM is part of choice PM > > >>> > > >>> IA64_HP_SIM is both a choice and is selected. > > >>> > > >>> I did allyesconfig and disabled PCI afterwards to find all the issues > > >>> on this patchset. > > >> Are you saying there's a newer series that fixes this issue for both > > >> allyesconfig and allmodconfig? > > >> > > >> if yes, then we're good. > > > > > > No, I haven't fixed ia64 kconfig issue. That's somebody else's job. I > > > used allyesconfig to find out all compilation issues on x86 arch to > > > come up with this patchset. > > > > Nothing makes me cringe more than "somebody else's job" statements. In > > this case, there is obviously a correlation with your ACPI changes since > > the circular dependency happens because of the ACPI symbol. > > > > arch/ia64/Kconfig:126:error: recursive dependency detected! > > arch/ia64/Kconfig:126: choice <choice> contains symbol IA64_HP_SIM > > arch/ia64/Kconfig:200: symbol IA64_HP_SIM is part of choice PM > > kernel/power/Kconfig:144: symbol PM is selected by PM_SLEEP > > kernel/power/Kconfig:104: symbol PM_SLEEP depends on HIBERNATE_CALLBACKS > > kernel/power/Kconfig:31: symbol HIBERNATE_CALLBACKS is selected by > > HIBERNATION > > kernel/power/Kconfig:34: symbol HIBERNATION depends on SWAP > > init/Kconfig:250: symbol SWAP depends on BLOCK > > block/Kconfig:5: symbol BLOCK is selected by UBIFS_FS > > fs/ubifs/Kconfig:1: symbol UBIFS_FS depends on MISC_FILESYSTEMS > > fs/Kconfig:220: symbol MISC_FILESYSTEMS is selected by ACPI_APEI > > drivers/acpi/apei/Kconfig:8: symbol ACPI_APEI depends on ACPI > > drivers/acpi/Kconfig:9: symbol ACPI depends on ARCH_SUPPORTS_ACPI > > <<<< LOOK HERE > > drivers/acpi/Kconfig:6: symbol ARCH_SUPPORTS_ACPI is selected by > > IA64_HP_SIM > > arch/ia64/Kconfig:200: symbol IA64_HP_SIM is part of choice <choice> > > > > At any rate, a 3 mn git bisect tells me the circular dependency is > > exposed by this change: > > > > f3fd6cd74fedf99b6060f75df00943fda13b65f2 is the first bad commit > > commit f3fd6cd74fedf99b6060f75df00943fda13b65f2 > > Author: Chandan Rajendra <chandan(a)linux.vnet.ibm.com> > > Date: Sat Dec 8 12:21:38 2018 +0530 > > > > fscrypt: remove filesystem specific build config option > > > > In order to have a common code base for fscrypt "post read" processing > > for all filesystems which support encryption, this commit removes > > filesystem specific build config option (e.g. > > CONFIG_EXT4_FS_ENCRYPTION) > > and replaces it with a build option (i.e. CONFIG_FS_ENCRYPTION) whose > > value affects all the filesystems making use of fscrypt. > > > > Signed-off-by: Chandan Rajendra <chandan(a)linux.vnet.ibm.com> > > Signed-off-by: Theodore Ts'o <tytso(a)mit.edu> > > > > FWIW, The patch at https://patchwork.kernel.org/patch/10725883/ fixes this > problem by removing "select BLOCK if FS_ENCRYPTION" from fs/ubifs/Kconfig. OK Pierre-Louis, can you check if this patch makes the issue go away for you, please?

3 2

[alsa-devel] [RFC PATCH] ASoC: fsl: Add Audio Mixer CPU DAI driver
by Viorel Suman 04 Jan '19

04 Jan '19

This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs. The Audio Mixer is a on-chip functional module that allows mixing of two audio streams into a single audio stream. Audio Mixer datasheet is available here: https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- .../devicetree/bindings/sound/fsl,amix.txt | 45 ++ sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/fsl_amix.c | 554 +++++++++++++++++++++ sound/soc/fsl/fsl_amix.h | 101 ++++ 5 files changed, 710 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,amix.txt create mode 100644 sound/soc/fsl/fsl_amix.c create mode 100644 sound/soc/fsl/fsl_amix.h diff --git a/Documentation/devicetree/bindings/sound/fsl,amix.txt b/Documentation/devicetree/bindings/sound/fsl,amix.txt new file mode 100644 index 0000000..049144f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,amix.txt @@ -0,0 +1,45 @@ +NXP Audio Mixer (AMIX). + +The Audio Mixer is a on-chip functional module that allows mixing of two +audio streams into a single audio stream. Audio Mixer has two input serial +audio interfaces. These are driven by two Synchronous Audio interface +modules (SAI). Each input serial interface carries 8 audio channels in its +frame in TDM manner. Mixer mixes audio samples of corresponding channels +from two interfaces into a single sample. Before mixing, audio samples of +two inputs can be attenuated based on configuration. The output of the +Audio Mixer is also a serial audio interface. Like input interfaces it has +the same TDM frame format. This output is used to drive the serial DAC TDM +interface of audio codec and also sent to the external pins along with the +receive path of normal audio SAI module for readback by the CPU. + +The output of Audio mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - Mixed audio + +Mixing operation is independent of audio sample rate but the two audio +input streams must have same audio sample rate with same number of channels +in TDM frame to be eligible for mixing. + +Device driver required properties: +================================= + - compatible : Compatible list, contains "fsl,imx8qm-amix" + + - reg : Offset and length of the register set for the device. + + - clocks : Must contain an entry for each entry in clock-names. + + - clock-names : Must include the "ipg" for register access. + + - power-domains : Must contain the phandle to the AMIX power domain node + +Device driver configuration example: +====================================== + amix: amix@59840000 { + compatible = "fsl,imx8qm-amix"; + reg = <0x0 0x59840000 0x0 0x10000>; + clocks = <&clk IMX8QXP_AUD_AMIX_IPG>; + clock-names = "ipg"; + power-domains = <&pd_amix>; + }; + diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 2e75b5bc..0b08eb2 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_AMIX + tristate "Audio Mixer (AMIX) module support" + select REGMAP_MMIO + help + Say Y if you want to add Audio Mixer (AMIX) + support for the NXP iMX CPUs. + config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index de94fa0..3263634 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-amix-objs := fsl_amix.o snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o @@ -21,6 +22,8 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o + +obj-$(CONFIG_SND_SOC_FSL_AMIX) += snd-soc-fsl-amix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o diff --git a/sound/soc/fsl/fsl_amix.c b/sound/soc/fsl/fsl_amix.c new file mode 100644 index 0000000..d752029 --- /dev/null +++ b/sound/soc/fsl/fsl_amix.c @@ -0,0 +1,554 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * NXP AMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "fsl_amix.h" + +#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \ + SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts) + +static const char + *tdm_sel[] = { "TDM1", "TDM2", }, + *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", }, + *width_sel[] = { "16b", "18b", "20b", "24b", "32b", }, + *pol_sel[] = { "Positive edge", "Negative edge", }, + *endis_sel[] = { "Disabled", "Enabled", }, + *updn_sel[] = { "Downward", "Upward", }, + *mask_sel[] = { "Unmask", "Mask", }; + +static const struct soc_enum fsl_amix_enum[] = { +/* FSL_AMIX_CTR enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MIXCLK_SHIFT, tdm_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTSRC_SHIFT, mode_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTWIDTH_SHIFT, width_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTCKPOL_SHIFT, pol_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MASKRTDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MASKCKDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_SYNCMODE_SHIFT, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_SYNCSRC_SHIFT, tdm_sel), +/* FSL_AMIX_ATCR0 enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR0, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR0, 1, updn_sel), +/* FSL_AMIX_ATCR1 enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR1, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR1, 1, updn_sel), +}; + +struct fsl_amix_state { + u8 tdms; + u8 clk; + char msg[64]; +}; + +static const struct fsl_amix_state prms[4][4] = {{ + /* DIS->DIS, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* DIS->TDM1*/ + { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" }, + /* DIS->TDM2*/ + { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" }, + /* DIS->MIX */ + { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" } +}, { /* TDM1->DIS */ + { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" }, + /* TDM1->TDM1, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM1->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" }, + /* TDM1->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" } +}, { /* TDM2->DIS */ + { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" }, + /* TDM2->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" }, + /* TDM2->TDM2, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM2->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" } +}, { /* MIX->DIS */ + { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" }, + /* MIX->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" }, + /* MIX->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" }, + /* MIX->MIX, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" } +}, }; + +static int fsl_amix_state_trans(struct snd_soc_component *comp, + unsigned int *mask, unsigned int *ctr, + const struct fsl_amix_state prm) +{ + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + /* Enforce all required TDMs are started */ + if ((priv->tdms & prm.tdms) != prm.tdms) { + dev_dbg(comp->dev, prm.msg); + return -EINVAL; + } + + switch (prm.clk) { + case 1: + case 2: + /* Set mix clock */ + (*mask) |= FSL_AMIX_CTR_MIXCLK_MASK; + (*ctr) |= FSL_AMIX_CTR_MIXCLK(prm.clk - 1); + break; + default: + break; + } + + return 0; +} + +static int fsl_amix_put_mix_clk_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int reg_val, val, mix_clk; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AMIX_CTR, &reg_val); + if (ret) + return ret; + + mix_clk = ((reg_val & FSL_AMIX_CTR_MIXCLK_MASK) + >> FSL_AMIX_CTR_MIXCLK_SHIFT); + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + if (!(priv->tdms & BIT(val))) { + dev_err(comp->dev, + "The selected clock source has no TDM%d enabled!\n", + val + 1); + return -EINVAL; + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static int fsl_amix_put_out_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + u32 out_src, mix_clk; + unsigned int reg_val, val, mask = 0, ctr = 0; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AMIX_CTR, &reg_val); + if (ret) + return ret; + + /* "From" state */ + out_src = ((reg_val & FSL_AMIX_CTR_OUTSRC_MASK) + >> FSL_AMIX_CTR_OUTSRC_SHIFT); + mix_clk = ((reg_val & FSL_AMIX_CTR_MIXCLK_MASK) + >> FSL_AMIX_CTR_MIXCLK_SHIFT); + + /* "To" state */ + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /* Check if state is changing ... */ + if (out_src == val) + return 0; + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + /* Check state transition constraints */ + ret = fsl_amix_state_trans(comp, &mask, &ctr, prms[out_src][val]); + if (ret) + return ret; + + /* Complete transition to new state */ + mask |= FSL_AMIX_CTR_OUTSRC_MASK; + ctr |= FSL_AMIX_CTR_OUTSRC(val); + + return snd_soc_component_update_bits(comp, FSL_AMIX_CTR, mask, ctr); +} + +static const struct snd_kcontrol_new fsl_amix_snd_controls[] = { + /* FSL_AMIX_CTR controls */ + SOC_ENUM_EXT("Mixing Clock Source", fsl_amix_enum[0], + snd_soc_get_enum_double, fsl_amix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_amix_enum[1], + snd_soc_get_enum_double, fsl_amix_put_out_src), + SOC_ENUM("Output Width", fsl_amix_enum[2]), + SOC_ENUM("Output Clock Polarity", fsl_amix_enum[3]), + SOC_ENUM("Frame Rate Diff Error", fsl_amix_enum[4]), + SOC_ENUM("Clock Freq Diff Error", fsl_amix_enum[5]), + SOC_ENUM("Sync Mode Config", fsl_amix_enum[6]), + SOC_ENUM("Sync Mode Clk Source", fsl_amix_enum[7]), + /* TDM1 Attenuation controls */ + SOC_ENUM("TDM1 Attenuation", fsl_amix_enum[8]), + SOC_ENUM("TDM1 Attenuation Direction", fsl_amix_enum[9]), + SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AMIX_ATCR0, + 2, 0x00fff, 0), + SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AMIX_ATIVAL0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AMIX_ATSTPUP0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AMIX_ATSTPDN0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AMIX_ATSTPTGT0, + 0, 0x3ffff, 0), + /* TDM2 Attenuation controls */ + SOC_ENUM("TDM2 Attenuation", fsl_amix_enum[10]), + SOC_ENUM("TDM2 Attenuation Direction", fsl_amix_enum[11]), + SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AMIX_ATCR1, + 2, 0x00fff, 0), + SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AMIX_ATIVAL1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AMIX_ATSTPUP1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AMIX_ATSTPDN1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AMIX_ATSTPTGT1, + 0, 0x3ffff, 0), +}; + +static int fsl_amix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + u32 mask = 0, ctr = 0; + + /* AMIX is working in DSP_A format only */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + /* For playback the AMIX is slave, and for record is master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + /* Output data will be written on positive edge of the clock */ + ctr |= FSL_AMIX_CTR_OUTCKPOL(0); + break; + case SND_SOC_DAIFMT_NB_NF: + /* Output data will be written on negative edge of the clock */ + ctr |= FSL_AMIX_CTR_OUTCKPOL(1); + break; + default: + return -EINVAL; + } + + mask |= FSL_AMIX_CTR_OUTCKPOL_MASK; + + return snd_soc_component_update_bits(comp, FSL_AMIX_CTR, mask, ctr); +} + +static int fsl_amix_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_amix *priv = snd_soc_dai_get_drvdata(dai); + + /* Capture stream shall not be handled */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->tdms |= BIT(dai->driver->id); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->tdms &= ~BIT(dai->driver->id); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops fsl_amix_dai_ops = { + .set_fmt = fsl_amix_dai_set_fmt, + .trigger = fsl_amix_dai_trigger, +}; + +static struct snd_soc_dai_driver fsl_amix_dai[] = { + { + .id = 0, + .name = "amix-0", + .playback = { + .stream_name = "AMIX-Playback-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .capture = { + .stream_name = "AMIX-Capture-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .ops = &fsl_amix_dai_ops, + }, + { + .id = 1, + .name = "amix-1", + .playback = { + .stream_name = "AMIX-Playback-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .capture = { + .stream_name = "AMIX-Capture-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .ops = &fsl_amix_dai_ops, + }, +}; + +static const struct snd_soc_component_driver fsl_amix_component = { + .name = "fsl-amix-dai", + .controls = fsl_amix_snd_controls, + .num_controls = ARRAY_SIZE(fsl_amix_snd_controls), +}; + +static bool fsl_amix_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AMIX_CTR: + case FSL_AMIX_STR: + case FSL_AMIX_ATCR0: + case FSL_AMIX_ATIVAL0: + case FSL_AMIX_ATSTPUP0: + case FSL_AMIX_ATSTPDN0: + case FSL_AMIX_ATSTPTGT0: + case FSL_AMIX_ATTNVAL0: + case FSL_AMIX_ATSTP0: + case FSL_AMIX_ATCR1: + case FSL_AMIX_ATIVAL1: + case FSL_AMIX_ATSTPUP1: + case FSL_AMIX_ATSTPDN1: + case FSL_AMIX_ATSTPTGT1: + case FSL_AMIX_ATTNVAL1: + case FSL_AMIX_ATSTP1: + return true; + default: + return false; + } +} + +static bool fsl_amix_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AMIX_CTR: + case FSL_AMIX_ATCR0: + case FSL_AMIX_ATIVAL0: + case FSL_AMIX_ATSTPUP0: + case FSL_AMIX_ATSTPDN0: + case FSL_AMIX_ATSTPTGT0: + case FSL_AMIX_ATCR1: + case FSL_AMIX_ATIVAL1: + case FSL_AMIX_ATSTPUP1: + case FSL_AMIX_ATSTPDN1: + case FSL_AMIX_ATSTPTGT1: + return true; + default: + return false; + } +} + +static const struct reg_default fsl_amix_reg[] = { + { FSL_AMIX_CTR, 0x00060 }, + { FSL_AMIX_STR, 0x00003 }, + { FSL_AMIX_ATCR0, 0x00000 }, + { FSL_AMIX_ATIVAL0, 0x3FFFF }, + { FSL_AMIX_ATSTPUP0, 0x2AAAA }, + { FSL_AMIX_ATSTPDN0, 0x30000 }, + { FSL_AMIX_ATSTPTGT0, 0x00010 }, + { FSL_AMIX_ATTNVAL0, 0x00000 }, + { FSL_AMIX_ATSTP0, 0x00000 }, + { FSL_AMIX_ATCR1, 0x00000 }, + { FSL_AMIX_ATIVAL1, 0x3FFFF }, + { FSL_AMIX_ATSTPUP1, 0x2AAAA }, + { FSL_AMIX_ATSTPDN1, 0x30000 }, + { FSL_AMIX_ATSTPTGT1, 0x00010 }, + { FSL_AMIX_ATTNVAL1, 0x00000 }, + { FSL_AMIX_ATSTP1, 0x00000 }, +}; + +static const struct regmap_config fsl_amix_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = FSL_AMIX_ATSTP1, + .reg_defaults = fsl_amix_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_amix_reg), + .readable_reg = fsl_amix_readable_reg, + .writeable_reg = fsl_amix_writeable_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int fsl_amix_probe(struct platform_device *pdev) +{ + struct fsl_amix *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->pdev = pdev; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs, + &fsl_amix_regmap_config); + if (IS_ERR(priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(priv->regmap); + } + + priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(priv->ipg_clk); + } + + platform_set_drvdata(pdev, priv); + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_amix_component, + fsl_amix_dai, + ARRAY_SIZE(fsl_amix_dai)); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + return 0; +} + +#ifdef CONFIG_PM +static int fsl_amix_runtime_resume(struct device *dev) +{ + struct fsl_amix *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->ipg_clk); + if (ret) { + dev_err(dev, "Failed to enable IPG clock: %d\n", ret); + return ret; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + return regcache_sync(priv->regmap); +} + +static int fsl_amix_runtime_suspend(struct device *dev) +{ + struct fsl_amix *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + + clk_disable_unprepare(priv->ipg_clk); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_amix_pm = { + SET_RUNTIME_PM_OPS(fsl_amix_runtime_suspend, + fsl_amix_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id fsl_amix_ids[] = { + { .compatible = "fsl,imx8qm-amix", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_amix_ids); + +static struct platform_driver fsl_amix_driver = { + .probe = fsl_amix_probe, + .driver = { + .name = "fsl-amix", + .of_match_table = fsl_amix_ids, + .pm = &fsl_amix_pm, + }, +}; +module_platform_driver(fsl_amix_driver); + +MODULE_DESCRIPTION("NXP AMIX ASoC DAI driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman(a)nxp.com>"); +MODULE_ALIAS("platform:fsl-amix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_amix.h b/sound/soc/fsl/fsl_amix.h new file mode 100644 index 0000000..5312683 --- /dev/null +++ b/sound/soc/fsl/fsl_amix.h @@ -0,0 +1,101 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * NXP AMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#ifndef __FSL_AMIX_H +#define __FSL_AMIX_H + +#define FSL_AMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +/* AMIX Registers */ +#define FSL_AMIX_CTR 0x200 /* Control */ +#define FSL_AMIX_STR 0x204 /* Status */ + +#define FSL_AMIX_ATCR0 0x208 /* Attenuation Control */ +#define FSL_AMIX_ATIVAL0 0x20c /* Attenuation Initial Value */ +#define FSL_AMIX_ATSTPUP0 0x210 /* Attenuation step up factor */ +#define FSL_AMIX_ATSTPDN0 0x214 /* Attenuation step down factor */ +#define FSL_AMIX_ATSTPTGT0 0x218 /* Attenuation step target */ +#define FSL_AMIX_ATTNVAL0 0x21c /* Attenuation Value */ +#define FSL_AMIX_ATSTP0 0x220 /* Attenuation step number */ + +#define FSL_AMIX_ATCR1 0x228 /* Attenuation Control */ +#define FSL_AMIX_ATIVAL1 0x22c /* Attenuation Initial Value */ +#define FSL_AMIX_ATSTPUP1 0x230 /* Attenuation step up factor */ +#define FSL_AMIX_ATSTPDN1 0x234 /* Attenuation step down factor */ +#define FSL_AMIX_ATSTPTGT1 0x238 /* Attenuation step target */ +#define FSL_AMIX_ATTNVAL1 0x23c /* Attenuation Value */ +#define FSL_AMIX_ATSTP1 0x240 /* Attenuation step number */ + +/* AMIX Control Register */ +#define FSL_AMIX_CTR_MIXCLK_SHIFT 0 +#define FSL_AMIX_CTR_MIXCLK_MASK BIT(FSL_AMIX_CTR_MIXCLK_SHIFT) +#define FSL_AMIX_CTR_MIXCLK(i) ((i) << FSL_AMIX_CTR_MIXCLK_SHIFT) +#define FSL_AMIX_CTR_OUTSRC_SHIFT 1 +#define FSL_AMIX_CTR_OUTSRC_MASK (0x3 << FSL_AMIX_CTR_OUTSRC_SHIFT) +#define FSL_AMIX_CTR_OUTSRC(i) (((i) << FSL_AMIX_CTR_OUTSRC_SHIFT) \ + & FSL_AMIX_CTR_OUTSRC_MASK) +#define FSL_AMIX_CTR_OUTWIDTH_SHIFT 3 +#define FSL_AMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AMIX_CTR_OUTWIDTH_SHIFT) +#define FSL_AMIX_CTR_OUTWIDTH(i) (((i) << FSL_AMIX_CTR_OUTWIDTH_SHIFT) \ + & FSL_AMIX_CTR_OUTWIDTH_MASK) +#define FSL_AMIX_CTR_OUTCKPOL_SHIFT 6 +#define FSL_AMIX_CTR_OUTCKPOL_MASK BIT(FSL_AMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AMIX_CTR_OUTCKPOL(i) ((i) << FSL_AMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AMIX_CTR_MASKRTDF_SHIFT 7 +#define FSL_AMIX_CTR_MASKRTDF_MASK BIT(FSL_AMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AMIX_CTR_MASKRTDF(i) ((i) << FSL_AMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AMIX_CTR_MASKCKDF_SHIFT 8 +#define FSL_AMIX_CTR_MASKCKDF_MASK BIT(FSL_AMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AMIX_CTR_MASKCKDF(i) ((i) << FSL_AMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AMIX_CTR_SYNCMODE_SHIFT 9 +#define FSL_AMIX_CTR_SYNCMODE_MASK BIT(FSL_AMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AMIX_CTR_SYNCMODE(i) ((i) << FSL_AMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AMIX_CTR_SYNCSRC_SHIFT 10 +#define FSL_AMIX_CTR_SYNCSRC_MASK BIT(FSL_AMIX_CTR_SYNCSRC_SHIFT) +#define FSL_AMIX_CTR_SYNCSRC(i) ((i) << FSL_AMIX_CTR_SYNCSRC_SHIFT) + +/* AMIX Status Register */ +#define FSL_AMIX_STR_RATEDIFF BIT(0) +#define FSL_AMIX_STR_CLKDIFF BIT(1) +#define FSL_AMIX_STR_MIXSTAT_SHIFT 2 +#define FSL_AMIX_STR_MIXSTAT_MASK (0x3 << FSL_AMIX_STR_MIXSTAT_SHIFT) +#define FSL_AMIX_STR_MIXSTAT(i) ((i & FSL_AMIX_STR_MIXSTAT_MASK) \ + >> FSL_AMIX_STR_MIXSTAT_SHIFT) +/* AMIX Attenuation Control Register */ +#define FSL_AMIX_ATCR_AT_EN BIT(0) +#define FSL_AMIX_ATCR_AT_UPDN BIT(1) +#define FSL_AMIX_ATCR_ATSTPDIF_SHIFT 2 +#define FSL_AMIX_ATCR_ATSTPDFI_MASK (0xfff << FSL_AMIX_ATCR_ATSTPDIF_SHIFT) + +/* AMIX Attenuation Initial Value Register */ +#define FSL_AMIX_ATIVAL_ATINVAL_MASK 0x3FFFF + +/* AMIX Attenuation Step Up Factor Register */ +#define FSL_AMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF + +/* AMIX Attenuation Step Down Factor Register */ +#define FSL_AMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF + +/* AMIX Attenuation Step Target Register */ +#define FSL_AMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF + +/* AMIX Attenuation Value Register */ +#define FSL_AMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF + +/* AMIX Attenuation Step Number Register */ +#define FSL_AMIX_ATSTP_STPCTR_MASK 0x3FFFF + +#define FSL_AMIX_MAX_DAIS 2 +struct fsl_amix { + struct platform_device *pdev; + struct regmap *regmap; + struct clk *ipg_clk; + u8 tdms; +}; + +#endif /* __FSL_AMIX_H */ -- 2.7.4

4 8

[alsa-devel] [PATCH] ASoC: rt5682: Fix recording no sound issue
by shumingf＠realtek.com 03 Jan '19

03 Jan '19

From: Shuming Fan <shumingf(a)realtek.com> The ADC mixer setting needs to restore to default value after calibration. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 34cfaf8..89c43b2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); mutex_unlock(&rt5682->calibrate_mutex); -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: virtuoso: add de-emphasis control
by tom.ty89＠gmail.com 03 Jan '19

03 Jan '19

From: Tom Yan <tom.ty89(a)gmail.com> Add control for the de-emphasis filter in the PCM179x DACs Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> diff --git a/sound/pci/oxygen/pcm1796.h b/sound/pci/oxygen/pcm1796.h index 34d07dd2d22e..d5dcb09e44cd 100644 --- a/sound/pci/oxygen/pcm1796.h +++ b/sound/pci/oxygen/pcm1796.h @@ -10,7 +10,6 @@ #define PCM1796_MUTE 0x01 #define PCM1796_DME 0x02 #define PCM1796_DMF_MASK 0x0c -#define PCM1796_DMF_DISABLED 0x00 #define PCM1796_DMF_48 0x04 #define PCM1796_DMF_441 0x08 #define PCM1796_DMF_32 0x0c diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 24109d37ca09..a1c6b98b191e 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -331,7 +331,7 @@ static void pcm1796_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = - PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; + PCM1796_FMT_24_I2S | PCM1796_ATLD; if (!data->broken_i2c) data->pcm1796_regs[0][18 - PCM1796_REG_BASE] |= PCM1796_MUTE; data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = @@ -621,6 +621,23 @@ static void update_pcm1796_oversampling(struct oxygen *chip) pcm1796_write_cached(chip, i, 20, reg); } +static void update_pcm1796_deemph(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 reg; + + reg = data->pcm1796_regs[0][18 - PCM1796_REG_BASE] & ~PCM1796_DMF_MASK; + if (data->current_rate == 48000) + reg |= PCM1796_DMF_48; + else if (data->current_rate == 44100) + reg |= PCM1796_DMF_441; + else if (data->current_rate == 32000) + reg |= PCM1796_DMF_32; + for (i = 0; i < data->dacs; ++i) + pcm1796_write_cached(chip, i, 18, reg); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -629,6 +646,7 @@ static void set_pcm1796_params(struct oxygen *chip, msleep(1); data->current_rate = params_rate(params); update_pcm1796_oversampling(chip); + update_pcm1796_deemph(chip); } static void update_pcm1796_volume(struct oxygen *chip) @@ -653,9 +671,11 @@ static void update_pcm1796_mute(struct oxygen *chip) unsigned int i; u8 value; - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; + value = data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; if (chip->dac_mute) value |= PCM1796_MUTE; + else + value &= ~PCM1796_MUTE; for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 18, value); } @@ -777,6 +797,49 @@ static const struct snd_kcontrol_new rolloff_control = { .put = rolloff_put, }; +static int deemph_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.integer.value[0] = + !!(data->pcm1796_regs[0][18 - PCM1796_REG_BASE] & PCM1796_DME); + return 0; +} + +static int deemph_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; + if (!value->value.integer.value[0]) + reg &= ~PCM1796_DME; + else + reg |= PCM1796_DME; + changed = reg != data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; + if (changed) { + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 18, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new deemph_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "De-emphasis Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = deemph_get, + .put = deemph_put, +}; + static const struct snd_kcontrol_new hdav_hdmi_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "HDMI Playback Switch", @@ -1011,6 +1074,10 @@ static int add_pcm1796_controls(struct oxygen *chip) snd_ctl_new1(&rolloff_control, chip)); if (err < 0) return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&deemph_control, chip)); + if (err < 0) + return err; } return 0; } -- 2.20.1

3 2

[alsa-devel] Applied "ASoC: Intel: atom: Make PCI dependency explicit" to the asoc tree
by Mark Brown 03 Jan '19

03 Jan '19

The patch ASoC: Intel: atom: Make PCI dependency explicit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6a444da7ddc6e478637a189a9c2898f2c3e0d399 Mon Sep 17 00:00:00 2001 From: Sinan Kaya <okaya(a)kernel.org> Date: Wed, 2 Jan 2019 18:10:35 +0000 Subject: [PATCH] ASoC: Intel: atom: Make PCI dependency explicit After 'commit 5d32a66541c4 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were satisfied implicitly through dependencies on CONFIG_ACPI have to be specified directly. This code relies on IOSF_MBI and IOSF_MBI depends on PCI. For this reason, add a direct dependency on CONFIG_PCI to the IOSF_MBI driver. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 99a62ba409df..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.20.1

1 0

Re: [alsa-devel] [PATCH] In function rt274_i2c_probe(), if the regmap_read fails. The variable "val" could leave uninitialized but used in if statement.
by Mark Brown 03 Jan '19

03 Jan '19

On Wed, Jan 02, 2019 at 06:38:39PM -0800, Yizhuo wrote: > int ret; > - unsigned int val; > + unsigned int val = ~0; > This just shuts warnings up which may mask further problem, it is better to just leave it and fix the users. > - regmap_read(rt274->regmap, > + ret = regmap_read(rt274->regmap, > RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); > - if (val != RT274_VENDOR_ID) { > + if (ret || val != RT274_VENDOR_ID) { > dev_err(&i2c->dev, > "Device with ID register %#x is not rt274\n", val); > return -ENODEV; If the read returned an error we should pass back the error code we got from it, not mask it with another error code. This is especially true here where the error message would be very misleading if there's a problem with I/O.

1 0