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

hdajackretask doesn't check for script's exit code
by GitHub issues - opened 06 Sep '21

06 Sep '21

alsa-project/alsa-tools issue #8 was opened from kroq-gar78: `hdajackretask` creates a script in `/tmp` before executing it. When `/tmp` is mounted with `noexec`, running that script will fail, but the GUI gives no indication that it failed. The GUI should check for the exit code of the script, and notify the user appropriately if the script failed to execute. Also, if this isn't the appropriate place to report a bug, please let me know! Issue URL : https://github.com/alsa-project/alsa-tools/issues/8 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

alsa-lib 1.2.5 and 1.2.5.1 fail to build with --enable-static
by GitHub issues - edited 06 Sep '21

06 Sep '21

alsa-project/alsa-lib issue #177 was edited from pjbroad: This build successfully: `./configure && make clean && make` This fails to build: `./configure --enable-shared=no --enable-static=yes && make clean && make` ``` make[1]: Entering directory '/home/paul/el/el-build-scripts/linux-static/libs/alsa-lib-1.2.5.1/aserver' CC aserver.o CCLD aserver /usr/bin/ld: ../src/.libs/libasound.a(control_symbols.o):(.data.rel+0x8): undefined reference to `_snd_module_control_empty' collect2: error: ld returned 1 exit status make[1]: *** [Makefile:398: aserver] Error 1 make[1]: Leaving directory '/home/paul/el/el-build-scripts/linux-static/libs/alsa-lib-1.2.5.1/aserver' make: *** [Makefile:405: all-recursive] Error 1 ``` `gcc (Ubuntu 11.2.0-3ubuntu1) 11.2.0` Building static worked find for 1.2.4. Hopefully someone can help resolve this. Let me know if you want me to try anything. Issue URL : https://github.com/alsa-project/alsa-lib/issues/177 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib 1.2.5 and 1.2.5.1 fail to build with --enable-static
by GitHub issues - opened 06 Sep '21

06 Sep '21

alsa-project/alsa-lib issue #177 was opened from pjbroad: This build successfully: ``` ./configure && make clean && make ``` This fails to build: ``` ./configure --enable-shared=no --enable-static=yes && make clean && make make[1]: Entering directory '/home/paul/el/el-build-scripts/linux-static/libs/alsa-lib-1.2.5.1/aserver' CC aserver.o CCLD aserver /usr/bin/ld: ../src/.libs/libasound.a(control_symbols.o):(.data.rel+0x8): undefined reference to `_snd_module_control_empty' collect2: error: ld returned 1 exit status make[1]: *** [Makefile:398: aserver] Error 1 make[1]: Leaving directory '/home/paul/el/el-build-scripts/linux-static/libs/alsa-lib-1.2.5.1/aserver' make: *** [Makefile:405: all-recursive] Error 1 ``` Building static worked find for 1.2.4. Hopefully someone can help resolve this. Let me know if you want me to try anything. Issue URL : https://github.com/alsa-project/alsa-lib/issues/177 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Asus Xonar SE front-panel audio support quirks snd_usb_audio
by smesgr 05 Sep '21

05 Sep '21

Greetings, I have bourght a Asus Xonar SE (90YA00T0-M0UA00) which is only working partially. While the rear connectors seems to work the front header is not. I found some reports scattered around the internet with the same issue but without a solution. It is a PCIe card with an pcie-to-usb interface. Thus the card is reported as a usb device: ID 0b05:189d ASUSTek Computer, Inc. Xonar SoundCard The card has several audio Endpoints EP 5 to EP 9 and an additonal HID EP 4. All the rear connectors seems to work. Can't test the SPDIF but found a SE specfic quirk in the alsa USB configuration for the SDPIF PCM thus assume this issue was already fixed. EP 5 -> Audio Out EP 6 -> SDPIF EP 8 -> Line In The rear connector don't have a plug-in-detector, the front-panel has and is working: node hw:1, #16 (0,0,0,Headphone - Output Jack,0) VALUE node hw:1, #31 (0,0,0,Mic - Input Jack,1) VALUE In stereo configuration the card has "headphones" and "speaker". The Front Panel input nor output is working. If headphones is selected the rear is disabled but the front header is not working. Side note the default Microsoft driver has the same issue so I assume the card usb configuration is faulty. I installed the ASUS driver and tracked the USB messages with wireshark in a Windows VM. I noticed the Asus driver uses those additional posts EP 7 -> Front Audio Out EP 9 -> Front Audio In Neither the default Windows, nor alsa use those EP. Is there a way to force the alsa driver to use EP7/9 to further analyse this issue? Best Regards Smesgr Sent with ProtonMail Secure Email.

2 1

pulseaudio: a52 sink doesn't stream audio from bluez source
by GitHub issues - edited 05 Sep '21

05 Sep '21

alsa-project/alsa-plugins issue #32 was edited from quequotion: Sorry to beat a dead horse, and double if this is in fact a pulseaudio issue. I recently configured bluez and pulseaudio to receive streams from devices over bluetooth (eg, to play audio from a phone over the PC's speaker system). This works when the PC's sound card is set to profiles *other than* the A52 plugin (ie, pulseaudio's provided digital stereo output and also the [DTS alsa plugin](https://github.com/darealshinji/dcaenc)). Only when set to use the AC3 output profile, audio from the bluez source is not produced. pavucontrol shows the bluez _source_ is producing sound (the level bounces appropriately), but no audio comes out of the _sink_ (no level bounce in pavucontrol either). Any ideas? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/32 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

No sound after kernel 5.8.15-301
by Сурков Павел 05 Sep '21

05 Sep '21

Hello! Â I lost sound from laptop speakers after updating the kernel. Sound via HDMI, AUX or Bluetooth works. Details on the links: [1]https://bbs.archlinux.org/viewtopic.php?id=269301 [2]https://ask.fedoraproject.org/t/no-sound-after-kernel-5-8-15-301/164 41 Â Help me please. --Â Pavel Surkov Â References 1. https://bbs.archlinux.org/viewtopic.php?id=269301 2. https://ask.fedoraproject.org/t/no-sound-after-kernel-5-8-15-301/16441

2 4

[PATCH v3 0/2] Add reset-gpios handling for max98927
by Alejandro 04 Sep '21

04 Sep '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) require hardware-resetting the codec by driving a reset-gpio. This series add support for it through an optional reset-gpios property. Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 14 ++++++++++++++ 2 files changed, 17 insertions(+) -- 2.33.0

5 10

[PATCH v5 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 04 Sep '21

04 Sep '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v5 changes: Reverse xmas-tree style for declarations At probe, regulator enable before handle pdata At probe, set pdata before component registration Remove several DT properties and implement as mixer controls Remove some unnecessary NULL inits in otp_unpack Revise license header style David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 162 ++ include/sound/cs35l41.h | 35 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 115 ++ sound/soc/codecs/cs35l41-spi.c | 144 ++ sound/soc/codecs/cs35l41-tables.c | 611 +++++++ sound/soc/codecs/cs35l41.c | 1587 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 774 ++++++++ 9 files changed, 3444 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

5 10

[PATCH v3] ASoC: rockchip: i2s: Fix concurrency between tx/rx
by Sugar Zhang 03 Sep '21

03 Sep '21

This patch adds lock to fix comcurrency between tx/rx to fix 'rockchip-i2s ff070000.i2s; fail to clear' Considering the situation; tx stream rx stream | | | disable enable | | reset After this patch: lock | tx stream | enable | unlock -------- --------- lock | rx stream | disable | reset | unlock Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v3: - Rebase and fix conflicts. Changes in v2: - Fix compile error. sound/soc/rockchip/rockchip_i2s.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 53e0b4a..7e89f5b 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -15,6 +15,7 @@ #include <linux/clk.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/spinlock.h> #include <sound/pcm_params.h> #include <sound/dmaengine_pcm.h> @@ -53,6 +54,7 @@ struct rk_i2s_dev { bool is_master_mode; const struct rk_i2s_pins *pins; unsigned int bclk_ratio; + spinlock_t lock; /* tx/rx lock */ }; static int i2s_runtime_suspend(struct device *dev) @@ -96,6 +98,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) unsigned int val = 0; int retry = 10; + spin_lock(&i2s->lock); if (on) { regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); @@ -136,6 +139,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } + spin_unlock(&i2s->lock); } static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) @@ -143,6 +147,7 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) unsigned int val = 0; int retry = 10; + spin_lock(&i2s->lock); if (on) { regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); @@ -183,6 +188,7 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } + spin_unlock(&i2s->lock); } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -684,6 +690,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; + spin_lock_init(&i2s->lock); i2s->dev = &pdev->dev; i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); -- 2.7.4

2 1

[PATCH v2] ASoC: rockchip: i2s: Fix concurrency between tx/rx
by Sugar Zhang 03 Sep '21

03 Sep '21

This patch adds lock to fix comcurrency between tx/rx to fix 'rockchip-i2s ff070000.i2s; fail to clear' Considering the situation; tx stream rx stream | | | disable enable | | reset After this patch: lock | tx stream | enable | unlock -------- --------- lock | rx stream | disable | reset | unlock Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - Fix compile error. sound/soc/rockchip/rockchip_i2s.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index c7dc350..88e340f 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -15,6 +15,7 @@ #include <linux/clk.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/spinlock.h> #include <sound/pcm_params.h> #include <sound/dmaengine_pcm.h> @@ -49,6 +50,7 @@ struct rk_i2s_dev { bool rx_start; bool is_master_mode; const struct rk_i2s_pins *pins; + spinlock_t lock; /* tx/rx lock */ }; static int i2s_runtime_suspend(struct device *dev) @@ -92,6 +94,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) unsigned int val = 0; int retry = 10; + spin_lock(&i2s->lock); if (on) { regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); @@ -132,6 +135,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } + spin_unlock(&i2s->lock); } static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) @@ -139,6 +143,7 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) unsigned int val = 0; int retry = 10; + spin_lock(&i2s->lock); if (on) { regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); @@ -179,6 +184,7 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } + spin_unlock(&i2s->lock); } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -589,6 +595,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; + spin_lock_init(&i2s->lock); i2s->dev = &pdev->dev; i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); -- 2.7.4

2 2

[PATCH v2 02/52] ASoC: dt-bindings: Convert Bluetooth SCO Link binding to a schema
by Maxime Ripard 03 Sep '21

03 Sep '21

Bluetooth SCO Link are supported by Linux with a matching device tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that driver over to a YAML schema. Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Samuel Holland <samuel(a)sholland.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Changes from v1: - Accept either a 0 or 1 #sound-dai-cells --- .../devicetree/bindings/sound/bt-sco.txt | 13 ------- .../bindings/sound/linux,bt-sco.yaml | 38 +++++++++++++++++++ 2 files changed, 38 insertions(+), 13 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/linux,bt-sco.yaml diff --git a/Documentation/devicetree/bindings/sound/bt-sco.txt b/Documentation/devicetree/bindings/sound/bt-sco.txt deleted file mode 100644 index 641edf75e184..000000000000 --- a/Documentation/devicetree/bindings/sound/bt-sco.txt +++ /dev/null @@ -1,13 +0,0 @@ -Bluetooth-SCO audio CODEC - -This device support generic Bluetooth SCO link. - -Required properties: - - - compatible : "delta,dfbmcs320" or "linux,bt-sco" - -Example: - -codec: bt_sco { - compatible = "delta,dfbmcs320"; -}; diff --git a/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml new file mode 100644 index 000000000000..e3a1f485f664 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml @@ -0,0 +1,38 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bluetooth SCO Audio Codec Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + '#sound-dai-cells': + enum: + - 0 + + # For Wideband PCM + - 1 + + compatible: + enum: + - delta,dfbmcs320 + - linux,bt-sco + +required: + - '#sound-dai-cells' + - compatible + +additionalProperties: false + +examples: + - | + codec { + #sound-dai-cells = <0>; + compatible = "linux,bt-sco"; + }; + +... -- 2.31.1

2 1

DSD Recording on RME 2-PRO FS/BE
by David Janello 03 Sep '21

03 Sep '21

Greetings, I am testing native DSD recording using arecord with the RME 2-PRO FS/BE audio interface. using command line arecord -c 2 -Dhw:CARD=Pro53791340 -r 176400 -f DSD_U32_BE audio.dsf gives output of: Recording WAVE 'audio.dsf' : Direct Stream Digital, 4-byte (x32), big endian, oldest bits in MSB, Rate 176400 Hz, Stereo arecord: set_params:1368: Sample format non available Available formats: - S32_LE Using S32_LE parameter arecord -c 2 -Dhw:CARD=Pro53791340 -r 176400 -f S32_LE audio.dsf gives output of Recording WAVE 'audio.dsf' : Signed 32 bit Little Endian, Rate 176400 Hz, Stereo Which appears to be a PCM recording and not DSD. What is the correct command line to arecord to capture DSD audio streams and write them to a dsd file? Thanks David

1 0

[PATCH] ASoC: Intel: boards: Fix CONFIG_SND_SOC_SDW_MOCKUP select
by Nathan Chancellor 03 Sep '21

03 Sep '21

When CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH is enabled without CONFIG_EXPERT, there is a Kconfig warning about unmet dependencies: WARNING: unmet direct dependencies detected for SND_SOC_SDW_MOCKUP Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && EXPERT [=n] && SOUNDWIRE [=y] Selected by [y]: - SND_SOC_INTEL_SOUNDWIRE_SOF_MACH [=y] && ... Selecting a symbol does not account for dependencies so if symbol A selects symbol B which depends on symbol C, symbol B or its select of symbol A should depend on symbol C as well. Make the CONFIG_SND_SOC_SDW_MOCKUP select in CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH depend on CONFIG_EXPERT as the help text for CONFIG_SND_SOC_SDW_MOCKUP indicates it is intended to be a development option. Fixes: 0ccac3bcf356 ("ASoC: Intel: boards: sof_sdw: add SoundWire mockup codecs for tests") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- An alternative here is if CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH wants to unconditionally select CONFIG_SND_SOC_SDW_MOCKUP, the "depends on EXPERT" can be moved to the prompt (tristate "..." if EXPERT). I am happy to send a new patch if that is what is desired. 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 046955bf717c..8284c46d7435 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -602,7 +602,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_MAXIM_COMMON - select SND_SOC_SDW_MOCKUP + select SND_SOC_SDW_MOCKUP if EXPERT help Add support for Intel SoundWire-based platforms connected to MAX98373, RT700, RT711, RT1308 and RT715 base-commit: 170c0d7460fc4aa522995ae4096b5a442f50a1fc -- 2.32.0.264.g75ae10bc75

3 9

[PATCH] ASoC: mt8195: correct the dts parsing logic about DPTX and HDMITX
by Trevor Wu 03 Sep '21

03 Sep '21

According to the description in dt-bindings, phandle assignment of HDMI TX and DP TX are not required properties, but driver regards them as required properties. In real use case, it's expected that DP TX and HDMI TX are optional features, so correct the behavior in driver. Fixes: 40d605df0a7b ("ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 22 +++++++++---------- 1 file changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index 5dc217f59bd6..c97ace7387b4 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -1018,13 +1018,12 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) of_parse_phandle(pdev->dev.of_node, "mediatek,dptx-codec", 0); if (!dai_link->codecs->of_node) { - dev_err(&pdev->dev, "Property 'dptx-codec' missing or invalid\n"); - return -EINVAL; + dev_dbg(&pdev->dev, "No property 'dptx-codec'\n"); + } else { + dai_link->codecs->name = NULL; + dai_link->codecs->dai_name = "i2s-hifi"; + dai_link->init = mt8195_dptx_codec_init; } - - dai_link->codecs->name = NULL; - dai_link->codecs->dai_name = "i2s-hifi"; - dai_link->init = mt8195_dptx_codec_init; } if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { @@ -1032,13 +1031,12 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) of_parse_phandle(pdev->dev.of_node, "mediatek,hdmi-codec", 0); if (!dai_link->codecs->of_node) { - dev_err(&pdev->dev, "Property 'hdmi-codec' missing or invalid\n"); - return -EINVAL; + dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n"); + } else { + dai_link->codecs->name = NULL; + dai_link->codecs->dai_name = "i2s-hifi"; + dai_link->init = mt8195_hdmi_codec_init; } - - dai_link->codecs->name = NULL; - dai_link->codecs->dai_name = "i2s-hifi"; - dai_link->init = mt8195_hdmi_codec_init; } } -- 2.18.0

2 1

[PATCH 0/2] Add code to manage DSP related clocks
by Daniel Baluta 03 Sep '21

03 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> DSP node on the Linux kernel side must also take care of enabling DAI/DMA related clocks. By design we choose to manage DAI/DMA clocks from the kernel side because of the architecture of some i.MX8 boards. Clocks are handled by a special M4 core which runs a special firmware called SCFW (System Controler firmware). This communicates with A cores running Linux via a special Messaging Unit and implements a custom API which is already implemented by the Linux kernel i.MX clocks implementation. Note that these clocks are optional. We can use the DSP without them. First patch was already sent to review on SOF Github: https://github.com/thesofproject/linux/pull/3131 Daniel Baluta (2): ASoC: SOF: imx: Add code to manage DSP related clocks dt-bindings: dsp: fsl: Add DSP optional clocks documentation .../devicetree/bindings/dsp/fsl,dsp.yaml | 33 ++++++++ sound/soc/sof/imx/imx-common.c | 77 +++++++++++++++++++ sound/soc/sof/imx/imx-common.h | 16 ++++ sound/soc/sof/imx/imx8.c | 32 ++++++++ sound/soc/sof/imx/imx8m.c | 33 ++++++++ 5 files changed, 191 insertions(+) -- 2.27.0

4 6

Re: [PATCH 1/2] ASoC: rt5682s: Add driver for ALC5682I-VS codec
by Mark Brown 03 Sep '21

03 Sep '21

On Fri, Aug 27, 2021 at 05:59:30AM +0000, Derek [方德義] wrote: > > > +#ifdef CONFIG_COMMON_CLK > > > + /* Check if MCLK provided */ > > > + rt5682s->mclk = devm_clk_get(component->dev, "mclk"); > > > > Perhaps factor the clock init out into a _probe_clks() function and then have a > > stub function rather than the two ifdefs? > We want to factor the codes like below to reduce ifdefs. > Could you give any comments? That looks fine.

1 0

[PATCH v2 01/52] ASoC: dt-bindings: Add WM8978 Binding
by Maxime Ripard 03 Sep '21

03 Sep '21

Even though we had the wm8978 driver for some time and a number of boards using it already, we never had a binding for it. Let's add it based on what the driver expects and the boards are providing. Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: patches(a)opensource.cirrus.com Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Changes from v1: - Changed the maintainers of the bindings --- .../devicetree/bindings/sound/wlf,wm8978.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8978.yaml diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000000..944e5859c962 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec Device Tree Bindings + +maintainers: + - patches(a)opensource.cirrus.com + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + }; + }; + +... -- 2.31.1

3 4

[PATCH v2 0/3] Convert name-prefix doc to json-schema
by Sameer Pujar 03 Sep '21

03 Sep '21

Following are the changes: - Add json-schema for 'sound-name-prefix' documentation under 'name-perfix.yaml' - Use schema references wherever needed. - Remove txt based doc Changelog ========= v1 -> v2 -------- * Provide top reference to name-prefix.yaml as suggested by Rob for patch 2/3 * Dropped couple of unreachable email ids from Cc list in commit message of patch 2/3 * No changes in remaining patches Sameer Pujar (3): ASoC: Add json-schema documentation for sound-name-prefix ASoC: Use schema reference for sound-name-prefix ASoC: Remove name-prefix.txt .../devicetree/bindings/sound/name-prefix.txt | 24 --------------- .../devicetree/bindings/sound/name-prefix.yaml | 35 ++++++++++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 9 ++---- .../bindings/sound/nvidia,tegra210-dmic.yaml | 9 ++---- .../bindings/sound/nvidia,tegra210-i2s.yaml | 9 ++---- .../devicetree/bindings/sound/nxp,tfa989x.yaml | 9 ++---- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- .../bindings/sound/simple-audio-mux.yaml | 9 ++---- 8 files changed, 51 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.yaml -- 2.7.4

3 11

Re: linux-next: Tree for Sep 1 [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko]
by Randy Dunlap 03 Sep '21

03 Sep '21

On 9/1/21 1:17 AM, Stephen Rothwell wrote: > Hi all, > > Please do not add any v5.16 related code to your linux-next included > branches until after v5.15-rc1 has been released. > > Changes since 20210831: > on x86_64: ERROR: modpost: "clkdev_add" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! ERROR: modpost: "clkdev_drop" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! ERROR: modpost: "clk_unregister_gate" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! ERROR: modpost: "clk_register_gate" [sound/soc/mediatek/mt8195/snd-soc-mt8195-afe.ko] undefined! Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

4 6

[PATCH] ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX8ULP
by Shengjiu Wang 02 Sep '21

02 Sep '21

Add compatible string for i.MX8ULP platform which support audio function through rpmsg audio channel on M core. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index 61802a11baf4..d370c98a62c7 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -21,6 +21,7 @@ properties: - fsl,imx8mn-rpmsg-audio - fsl,imx8mm-rpmsg-audio - fsl,imx8mp-rpmsg-audio + - fsl,imx8ulp-rpmsg-audio model: $ref: /schemas/types.yaml#/definitions/string -- 2.17.1

2 1

config issue with SoundWire mockup device support
by Fabio Aiuto 02 Sep '21

02 Sep '21

Dear Pierre and all sound developers, I work for some tests on a Lenovo ideapad Miix 300-10IBY, with fedora 34 workstation intalled on. After having pulled the last changes from staging tree I get the following issue by typing: $ make olddefconfig WARNING: unmet direct dependencies detected for SND_SOC_SDW_MOCKUP Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && EXPERT [=n] && SOUNDWIRE [=y] Selected by [m]: - SND_SOC_INTEL_SOUNDWIRE_SOF_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_INTEL_MACH [=y] && SND_SOC_SOF_INTEL_SOUNDWIRE [=m] && I2C [=y] && ACPI [=y] && GPIOLIB [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) && (SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES [=y] || COMPILE_TEST [=n]) && SOUNDWIRE [=y] && SND_HDA_CODEC_HDMI [=m] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] WARNING: unmet direct dependencies detected for SND_SOC_SDW_MOCKUP Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && EXPERT [=n] && SOUNDWIRE [=y] Selected by [m]: - SND_SOC_INTEL_SOUNDWIRE_SOF_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_INTEL_MACH [=y] && SND_SOC_SOF_INTEL_SOUNDWIRE [=m] && I2C [=y] && ACPI [=y] && GPIOLIB [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) && (SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES [=y] || COMPILE_TEST [=n]) && SOUNDWIRE [=y] && SND_HDA_CODEC_HDMI [=m] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] # # No change to .config <--- this one gets written the first time # this is an n-th run output Attached the config file I took from my own fedora distribution (just lockdep was enabled). Please tell me if you need more information on this issue. Thank you, fabio

2 2

[PATCH RESEND 0/2] Add reset-gpios handling for max98927
by Alejandro Tafalla 02 Sep '21

02 Sep '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) require hardware-resetting the codec by driving a reset-gpio. This series add support for it through an optional reset-gpios property. Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 16 ++++++++++++++++ sound/soc/codecs/max98927.h | 1 + 3 files changed, 20 insertions(+) -- 2.32.0

4 6

Changes in ALSA firewire stack at Linux kernel v5.14 release
by Takashi Sakamoto 02 Sep '21

02 Sep '21

Dear all, I'm a maintainer of ALSA firewire stack. Linux kernel v5.14 was out a few days ago[1], including some changes in ALSA firewire stack. The changes bring improvement for usage of including drivers by solving some issues. I appreciate the users cooperating for it[2]. This message includes two topics about solved issues in the release: 1. get rid of playback noise by recovering media clock 2. allow some applications to run without periodical hardware interrupts and another topic: 3. device aggregation Let me describe the two topics first. 1. get rid of playback noise by recovering media clock Many users had been reporting playback noise since the initial version of each driver in ALSA firewire stack. The cause of the issue is complicated to explain, but let me roughly summarize it to a point below: * mismatch between audio sample count in playback stream and the one expected by hardware Since the initial stage of ALSA firewire stack, included drivers transfer audio data frames per second the exact count as sampling frequency, which is configured via ALSA PCM interface; e.g. 44.1 kHz. But it is figured out that it is not suitable for many models. For recent years, I've measured actual packets from/to various models with Windows and OS X drivers[3], and realized the below phenomena. Here, the configured frequency is called 'nominal', and the measured frequency is called 'effective'. * the effective frequency is not the same as the nominal frequency, less or greater by several audio data frames (<= 10 frames) * the effective frequency is not even in successive seconds for some models The phenomena mean that it is not achieved to transfer samples for playback sound by nominal frequency, and computation for even number of samples per second for some models. Additionally, in isochronous communication of IEEE 1394, part of models support time stamp per isochronous packet[4]. When parsing the sequence of time stamp and compare it to frequency of samples included in the packets, I realize the phenomena below: * the phase of sample based on computed time stamp shifts during long packet streaming * before and after configuring source of sampling clock to external signal input such as S/PDIF, neither the effective frequency of samples in packets nor the sequence of time stamp in packets have difference. The phenomena give us some insights. At least, the phase of samples is not deterministic somehow in driver side. It is required to recover the timing to put audio data frame into packet according to packets transferred by the hardware. This is called 'media clock recovery'[5]. In engineering field, many method of media clock recovery has been invented for each type of media. The way which ALSA firewire stack in v5.14 uses is the simplest one. It is to replay the sequence in transferred packets[6][7][8]. The result looks better. As long as I tested, it can cover all of models supported by ALSA firewire stack. 2. allow applications to run independently of periodical hardware interrupts ALSA PCM interface has hardware parameter for runtime of PCM substream to process audio data frame without scheduling periodical hardware interrupts[9]. PulseAudio and PipeWire use the function. All of drivers[10] in ALSA firewire stack now support it. Linux FireWire subsystem has function to flush queued packet till the most recent isochronous cycle. The function is available in process context without waiting for scheduled hardware interrupts, and allows drivers to achieve the topic. In most cases, it's done by calling ioctl(2) with SNDRV_PCM_IOCTL_HWSYNC. The call is the part of routine in usual ALSA PCM application, thus users do not need to take extra care of it. I think these improvements and configurable size of PCM buffer added in Linux kernel v5.5 brings better experience to users. The rest of topic comes from comparison to what existent userspace driver, libffado2[11], does. 3. device aggregation Some user pointed that it is not available with drivers in ALSA firewire stack to aggregate several audio data stream into one stream. It is what libffado2 does. Let me describe my opinion about it. At first, let me start with fundamental attribute of audio data frame. In my understanding about ALSA PCM interface, audio data frame is a unit for audio data transmission. The audio data frame includes the specific number of audio data depending on hardware; e.g. 2 samples for usual sound device. The fundamental attribute of audio data frame is to include the set of audio data sampled at the same time. The goal of aggregating audio data stream is to construct an audio data frame from some audio data frames for several devices. It means that one audio data frame consists of audio data sampled at different time. As I describe the phenomena about nominal and effective frequency, each hardware seems to run own unique effective frequency time to time[12], at least over IEEE 1394 bus. Additionally, we have the experience that the hardware is not aware of sequence of packet with nominal frequency for sample synchronization. It might be legitimate that we can not pick up audio data frame which consists of audio data sampled at the same time even if they are transferred at the same isochronous cycle[13]. When achieving the aggregation, we would need to loosen up the fundamental attribute of audio data frame, by accepting the range of sampling time for audio data in the frame, or need to implement one of resampling methods to adjust phase of audio data to the frame. Although the aggregation is itself superficially useful, it seems not to be a requirement to device driver in hardware abstraction layer of general purpose operating system. It may be over engineering. [1] Linux 5.14 https://lore.kernel.org/lkml/CAHk-=wh75ELUu99yPkPNt+R166CK=-M4eoV+F62tW3TVg… [2] The cooperation is done in my public repository in github.com: https://github.com/takaswie/snd-firewire-improve [3] The method is described in the message: [alsa-devel] IEEE 1394 isoc library, libhinoko v0.1.0 release https://lore.kernel.org/alsa-devel/20190415153053.GA32090@workstation/ [4] The resolution of time stamp is 24.576 MHz, which is the same as specification of cycle time in IEEE 1394. The method to compute time stamp of packet for samples in the packet is defined by IEC 61883-6. We can see integrated document for it published by industry group: https://web.archive.org/web/20210216003054/http://1394ta.org/wp-content/upl… [5] I borrow the expression from IEEE 1722. We can see specific term, sampling transmission frequency (STF) in IEC 61883-6 to express similar idea of the media clock. [6] [PATCH 0/3] ALSA: firewire: media clock recovery for syt-aware devices https://lore.kernel.org/alsa-devel/20210601081753.9191-1-o-takashi@sakamocc… [7] [PATCH 0/6] ALSA: firewire: media clock recovery for syt-unaware devices https://lore.kernel.org/alsa-devel/20210531025103.17880-1-o-takashi@sakamoc… [8] [PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices https://lore.kernel.org/alsa-devel/20210602013406.26442-1-o-takashi@sakamoc… [9] SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP. When the PCM substream has a flag of SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, it's available. [10] Precisely except for snd-isight. [11] http://www.ffado.org/ [12] Precisely the hardware looks to run own unique media clock over IEEE 1394 bus. [13] For precise discussion, the knowledge about IEC 61883-6 and vendor specific method for packetization is required. Regards Takashi Sakamoto

4 3

[PATCH] ALSA: usb-audio: Add registration quirk for JBL Quantum 800
by Alexander Tsoy 01 Sep '21

01 Sep '21

Add another device ID for JBL Quantum 800. It requires the same quirk as other JBL Quantum devices. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4479a590194f..6ee6d24c847f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1743,6 +1743,7 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ -- 2.31.1

2 1

[PATCH] ASoC: mediatek: mt8195: add MTK_PMIC_WRAP dependency
by Trevor Wu 01 Sep '21

01 Sep '21

randconfig builds show the warning below, WARNING: unmet direct dependencies detected for SND_SOC_MT6359 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MTK_PMIC_WRAP [=n] Selected by [y]: - SND_SOC_MT8195_MT6359_RT1019_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8195 [=y] Add a dependency to avoid it. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index cf567a89f421..f6bf0f9c91e3 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -197,7 +197,7 @@ config SND_SOC_MT8195 config SND_SOC_MT8195_MT6359_RT1019_RT5682 tristate "ASoC Audio driver for MT8195 with MT6359 RT1019 RT5682 codec" depends on I2C - depends on SND_SOC_MT8195 + depends on SND_SOC_MT8195 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C -- 2.18.0

3 2

[PATCH] ASoC: audio-graph: respawn Platform Support
by Kuninori Morimoto 01 Sep '21

01 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch needs Olivier's Tested-by sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.25.1

3 2

[PATCH] ASoC: rt5682: fix headset background noise when S3 state
by shumingf＠realtek.com 01 Sep '21

01 Sep '21

From: Shuming Fan <shumingf(a)realtek.com> Remove CBJ power off setting to avoid floating state. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e822fa1b9d4b..4a64cab99c55 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2942,9 +2942,6 @@ static int rt5682_suspend(struct snd_soc_component *component) break; } - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, - RT5682_PWR_CBJ, 0); - /* enter SAR ADC power saving mode */ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | -- 2.31.1

2 1

[PATCH] ASoC: samsung: s3c24xx_simtec: fix spelling mistake "devicec" -> "device"
by Colin King 01 Sep '21

01 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err error message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/samsung/s3c24xx_simtec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 81a29d12c57d..0cc66774b85d 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -327,7 +327,7 @@ int simtec_audio_core_probe(struct platform_device *pdev, snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + dev_err(&pdev->dev, "failed to alloc soc-audio device\n"); ret = -ENOMEM; goto err_gpio; } -- 2.32.0

3 2

[PATCH] ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK
by Geert Uytterhoeven 01 Sep '21

01 Sep '21

The Mediatek MT8195 sound hardware is only present on Mediatek MT8195 SoCs. Hence add a dependency on ARCH_MEDIATEK, to prevent asking the user about this driver when configuring a kernel without Mediatek SoC support. Fixes: 6746cc858259985a ("ASoC: mediatek: mt8195: add platform driver") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index cf567a89f421b4c0..c8c815f599da90bf 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -187,6 +187,7 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 config SND_SOC_MT8195 tristate "ASoC support for Mediatek MT8195 chip" + depends on ARCH_MEDIATEK || COMPILE_TEST select SND_SOC_MEDIATEK help This adds ASoC platform driver support for Mediatek MT8195 chip -- 2.25.1

4 3

[PATCH] ASoC: dt-bindings: mt8195: remove dependent headers in the example
by Trevor Wu 01 Sep '21

01 Sep '21

Drop the use of the defines including clock and power id, so that we can remove the headers which are not applied in the example. Fixes: b5bac34fcfb4 ("dt-bindings: mediatek: mt8195: add audio afe document") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../bindings/sound/mt8195-afe-pcm.yaml | 40 +++++++++---------- 1 file changed, 19 insertions(+), 21 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml index 53e9434a6d9d..dcf790b053d2 100644 --- a/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml +++ b/Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml @@ -130,36 +130,34 @@ additionalProperties: false examples: - | - #include <dt-bindings/clock/mt8195-clk.h> #include <dt-bindings/interrupt-controller/arm-gic.h> #include <dt-bindings/interrupt-controller/irq.h> - #include <dt-bindings/power/mt8195-power.h> afe: mt8195-afe-pcm@10890000 { compatible = "mediatek,mt8195-audio"; reg = <0x10890000 0x10000>; interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; mediatek,topckgen = <&topckgen>; - power-domains = <&spm MT8195_POWER_DOMAIN_AUDIO>; + power-domains = <&spm 7>; //MT8195_POWER_DOMAIN_AUDIO clocks = <&clk26m>, - <&topckgen CLK_TOP_APLL1>, - <&topckgen CLK_TOP_APLL2>, - <&topckgen CLK_TOP_APLL12_DIV0>, - <&topckgen CLK_TOP_APLL12_DIV1>, - <&topckgen CLK_TOP_APLL12_DIV2>, - <&topckgen CLK_TOP_APLL12_DIV3>, - <&topckgen CLK_TOP_APLL12_DIV9>, - <&topckgen CLK_TOP_A1SYS_HP_SEL>, - <&topckgen CLK_TOP_AUD_INTBUS_SEL>, - <&topckgen CLK_TOP_AUDIO_H_SEL>, - <&topckgen CLK_TOP_AUDIO_LOCAL_BUS_SEL>, - <&topckgen CLK_TOP_DPTX_M_SEL>, - <&topckgen CLK_TOP_I2SO1_M_SEL>, - <&topckgen CLK_TOP_I2SO2_M_SEL>, - <&topckgen CLK_TOP_I2SI1_M_SEL>, - <&topckgen CLK_TOP_I2SI2_M_SEL>, - <&infracfg_ao CLK_INFRA_AO_AUDIO_26M_B>, - <&scp_adsp CLK_SCP_ADSP_AUDIODSP>; + <&topckgen 163>, //CLK_TOP_APLL1 + <&topckgen 166>, //CLK_TOP_APLL2 + <&topckgen 233>, //CLK_TOP_APLL12_DIV0 + <&topckgen 234>, //CLK_TOP_APLL12_DIV1 + <&topckgen 235>, //CLK_TOP_APLL12_DIV2 + <&topckgen 236>, //CLK_TOP_APLL12_DIV3 + <&topckgen 238>, //CLK_TOP_APLL12_DIV9 + <&topckgen 100>, //CLK_TOP_A1SYS_HP_SEL + <&topckgen 33>, //CLK_TOP_AUD_INTBUS_SEL + <&topckgen 34>, //CLK_TOP_AUDIO_H_SEL + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS_SEL + <&topckgen 98>, //CLK_TOP_DPTX_M_SEL + <&topckgen 94>, //CLK_TOP_I2SO1_M_SEL + <&topckgen 95>, //CLK_TOP_I2SO2_M_SEL + <&topckgen 96>, //CLK_TOP_I2SI1_M_SEL + <&topckgen 97>, //CLK_TOP_I2SI2_M_SEL + <&infracfg_ao 50>, //CLK_INFRA_AO_AUDIO_26M_B + <&scp_adsp 0>; //CLK_SCP_ADSP_AUDIODSP clock-names = "clk26m", "apll1_ck", "apll2_ck", -- 2.18.0

3 2

[PATCH v2 0/2] Add reset-gpios handling for max98927
by Alejandro Tafalla 01 Sep '21

01 Sep '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) require hardware-resetting the codec by driving a reset-gpio. This series add support for it through an optional reset-gpios property. Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 15 +++++++++++++++ 2 files changed, 18 insertions(+) -- 2.32.0

3 5

[PATCH 0/3] Convert name-prefix doc to json-schema
by Sameer Pujar 01 Sep '21

01 Sep '21

Following are the changes: - Add json-schema for 'sound-name-prefix' documentation under 'name-perfix.yaml' - Use schema references wherever needed. - Remove txt based doc Sameer Pujar (3): ASoC: Add json-schema documentation for sound-name-prefix ASoC: Use schema reference for sound-name-prefix ASoC: Remove name-prefix.txt .../devicetree/bindings/sound/name-prefix.txt | 24 --------------- .../devicetree/bindings/sound/name-prefix.yaml | 35 ++++++++++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 7 +---- .../bindings/sound/nvidia,tegra210-dmic.yaml | 7 +---- .../bindings/sound/nvidia,tegra210-i2s.yaml | 7 +---- .../devicetree/bindings/sound/nxp,tfa989x.yaml | 5 +--- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- .../bindings/sound/simple-audio-mux.yaml | 5 +--- 8 files changed, 41 insertions(+), 51 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.yaml -- 2.7.4

2 4

[PATCH v4 00/20] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 01 Sep '21

01 Sep '21

Many thanks for reviewing v3. This version addresses all the comments raised as part of v3 review. 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 simpified 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/ Thanks, srini Changes since v3: - Split some dt-bindings patches for easy review. - fixed various cppcheck warnings and ignore some bogus ones. - fixed memory leak in various places in topology - reordered Kconfig and Makefile changes - cleaned Kconfig symbols to fit in properly with existing Q6DSP KConfigs - added lpass suffix to compatible for nodes that are dealing with LPASS IP. - fix few typos. Srinivas Kandagatla (20): soc: dt-bindings: qcom: apr: convert to yaml soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support 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 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/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 209 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 50 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 74 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 202 +++ drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 286 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 203 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1170 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 692 ++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 504 +++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 266 ++++ sound/soc/qcom/qdsp6/q6apm.c | 916 +++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 154 +++ 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 | 256 ++++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1118 ++++++++++++++++ 31 files changed, 7443 insertions(+), 1414 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml 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/soc/qcom,gpr.h 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 24

[PATCH v3 0/7] Patches to update for rockchip pdm
by Sugar Zhang 01 Sep '21

01 Sep '21

These patches fixup or update for rockchip pdm. Changes in v3: - Fix property 'path-map' suggested by Rob Herring. Changes in v2: - Fix yamllint errors. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 -------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 120 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 +++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 232 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

2 11

[PATCH v2 04/52] ASoC: dt-bindings: Convert Simple Amplifier binding to a schema
by Maxime Ripard 01 Sep '21

01 Sep '21

Simple audio amplifiers are supported by Linux with a matching device tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that driver over to a YAML schema. Cc: alsa-devel(a)alsa-project.org Cc: Jerome Brunet <jbrunet(a)baylibre.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../bindings/sound/simple-amplifier.txt | 17 ------- .../sound/simple-audio-amplifier.yaml | 45 +++++++++++++++++++ 2 files changed, 45 insertions(+), 17 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-amplifier.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt deleted file mode 100644 index b1b097cc9b68..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ /dev/null @@ -1,17 +0,0 @@ -Simple Amplifier Audio Driver - -Required properties: -- compatible : "dioo,dio2125" or "simple-audio-amplifier" - -Optional properties: -- enable-gpios : the gpio connected to the enable pin of the simple amplifier -- VCC-supply : power supply for the device, as covered - in Documentation/devicetree/bindings/regulator/regulator.txt - -Example: - -amp: analog-amplifier { - compatible = "simple-audio-amplifier"; - VCC-supply = <&regulator>; - enable-gpios = <&gpio GPIOH_3 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml new file mode 100644 index 000000000000..26379377a7ac --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-amplifier.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Amplifier Device Tree Bindings + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +properties: + compatible: + enum: + - dioo,dio2125 + - simple-audio-amplifier + + enable-gpios: + maxItems: 1 + + VCC-supply: + description: > + power supply for the device + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: > + See ./name-prefix.txt + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/meson8-gpio.h> + + analog-amplifier { + compatible = "simple-audio-amplifier"; + VCC-supply = <&regulator>; + enable-gpios = <&gpio GPIOH_3 0>; + }; + +... -- 2.31.1

1 0

[PATCH v2 03/52] ASoC: dt-bindings: Convert SPDIF Transmitter binding to a schema
by Maxime Ripard 01 Sep '21

01 Sep '21

The SPDIF Transmitter binding is used by Linux with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that driver over to a YAML schema. Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../bindings/sound/linux,spdif-dit.yaml | 32 +++++++++++++++++++ .../bindings/sound/spdif-transmitter.txt | 10 ------ 2 files changed, 32 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml delete mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt diff --git a/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml new file mode 100644 index 000000000000..c6b070e1d014 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml @@ -0,0 +1,32 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,spdif-dit.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dummy SPDIF Transmitter Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + const: linux,spdif-dit + + "#sound-dai-cells": + const: 0 + +required: + - "#sound-dai-cells" + - compatible + +additionalProperties: false + +examples: + - | + spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt b/Documentation/devicetree/bindings/sound/spdif-transmitter.txt deleted file mode 100644 index 55a85841dd85..000000000000 --- a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt +++ /dev/null @@ -1,10 +0,0 @@ -Device-Tree bindings for dummy spdif transmitter - -Required properties: - - compatible: should be "linux,spdif-dit". - -Example node: - - codec: spdif-transmitter { - compatible = "linux,spdif-dit"; - }; -- 2.31.1

1 0

[PATCH] kconfig: forbid symbols that end with '_MODULE'
by Masahiro Yamada 01 Sep '21

01 Sep '21

Kconfig (syncconfig) generates include/generated/autoconf.h to make CONFIG options available to the pre-processor. The macros are suffixed with '_MODULE' for symbols with the value 'm'. Here is a conflict; CONFIG_FOO=m results in '#define CONFIG_FOO_MODULE 1', but CONFIG_FOO_MODULE=y also results in the same define. fixdep always assumes CONFIG_FOO_MODULE comes from CONFIG_FOO=m, so the dependency is not properly tracked for symbols that end with '_MODULE'. This commit makes Kconfig error out if it finds a symbol suffixed with '_MODULE'. This restriction does not exist if the module feature is not supported (at least from the Kconfig perspective). It detected one error: error: SND_SOC_DM365_VOICE_CODEC_MODULE: symbol name must not end with '_MODULE' Rename it to SND_SOC_DM365_VOICE_CODEC_MODULAR. Commit 147162f57515 ("ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies") added it for internal use. So, this renaming has no impact on users. Remove a comment from drivers/net/wireless/intel/iwlwifi/Kconfig since this is a hard error now. Add a comment to include/linux/kconfig.h in order not to worry observant developers. Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- drivers/net/wireless/intel/iwlwifi/Kconfig | 1 - include/linux/kconfig.h | 3 ++ scripts/kconfig/parser.y | 40 +++++++++++++++++++++- sound/soc/ti/Kconfig | 2 +- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/drivers/net/wireless/intel/iwlwifi/Kconfig b/drivers/net/wireless/intel/iwlwifi/Kconfig index 1085afbefba8..5b238243617c 100644 --- a/drivers/net/wireless/intel/iwlwifi/Kconfig +++ b/drivers/net/wireless/intel/iwlwifi/Kconfig @@ -70,7 +70,6 @@ config IWLMVM of the devices that use this firmware is available here: https://wireless.wiki.kernel.org/en/users/drivers/iwlwifi#firmware -# don't call it _MODULE -- will confuse Kconfig/fixdep/... config IWLWIFI_OPMODE_MODULAR bool default y if IWLDVM=m diff --git a/include/linux/kconfig.h b/include/linux/kconfig.h index 20d1079e92b4..54f677e742fe 100644 --- a/include/linux/kconfig.h +++ b/include/linux/kconfig.h @@ -53,6 +53,9 @@ * IS_MODULE(CONFIG_FOO) evaluates to 1 if CONFIG_FOO is set to 'm', 0 * otherwise. CONFIG_FOO=m results in "#define CONFIG_FOO_MODULE 1" in * autoconf.h. + * CONFIG_FOO_MODULE=y would also result in "#define CONFIG_FOO_MODULE 1", + * but Kconfig forbids symbol names that end with '_MODULE', so that would + * not happen. */ #define IS_MODULE(option) __is_defined(option##_MODULE) diff --git a/scripts/kconfig/parser.y b/scripts/kconfig/parser.y index 2af7ce4e1531..b0f73f74ccd3 100644 --- a/scripts/kconfig/parser.y +++ b/scripts/kconfig/parser.y @@ -475,6 +475,37 @@ assign_val: %% +/* + * Symbols suffixed with '_MODULE' would cause a macro conflict in autoconf.h, + * and also confuse the interaction between syncconfig and fixdep. + * Error out if a symbol with the '_MODULE' suffix is found. + */ +static int sym_check_name(struct symbol *sym) +{ + static const char *suffix = "_MODULE"; + static const size_t suffix_len = strlen("_MODULE"); + char *name; + size_t len; + + name = sym->name; + + if (!name) + return 0; + + len = strlen(name); + + if (len < suffix_len) + return 0; + + if (strcmp(name + len - suffix_len, suffix)) + return 0; + + fprintf(stderr, "error: %s: symbol name must not end with '%s'\n", + name, suffix); + + return -1; +} + void conf_parse(const char *name) { struct symbol *sym; @@ -493,8 +524,15 @@ void conf_parse(const char *name) if (yynerrs) exit(1); - if (!modules_sym) + + if (modules_sym) { + for_all_symbols(i, sym) { + if (sym_check_name(sym)) + yynerrs++; + } + } else { modules_sym = sym_find( "n" ); + } if (!menu_has_prompt(&rootmenu)) { current_entry = &rootmenu; diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 698d7bc84dcf..c56a5789056f 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -211,7 +211,7 @@ config SND_SOC_DM365_VOICE_CODEC Say Y if you want to add support for SoC On-chip voice codec endchoice -config SND_SOC_DM365_VOICE_CODEC_MODULE +config SND_SOC_DM365_VOICE_CODEC_MODULAR def_tristate y depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC select MFD_DAVINCI_VOICECODEC -- 2.30.2

4 7

external plugins and openmp
by Matt Flax 01 Sep '21

01 Sep '21

Has anyone succeeded in using openmp with ALSA external plugins ? I compile using the -fopenmp flags however I can't seem to increase the number of threads beyond 1. Matt

1 0

[PATCH v2 0/7] Patches to update for rockchip pdm
by Sugar Zhang 01 Sep '21

01 Sep '21

These patches fixup or update for rockchip pdm. Changes in v2: - Fix yamllint errors. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 --------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 115 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 ++++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 227 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

2 9

[PATCH v3 0/4] Rockchip I2S/TDM controller
by Nicolas Frattaroli 01 Sep '21

01 Sep '21

Hello, this is version 3 of the I2S/TDM driver patchset. A big thanks to everyone who has provided their valuable feedback so far. I've decided to omit the original cover letter for this version, as it got quite long and had some things that are outdated or weren't the case as I've discovered from better understanding the driver code. I've decided not to remove the synced reset code. The udelay matter has been settled, and it is code that most likely works, even if I cannot test it right now. The implementation of half a reset controller is ugly, but specifying a generalised n-reset synchronised reset is a bit more complex to do than just doing it for 2, so I'd prefer putting that work off until later. Changes in v3: driver: - alphabetically sort includes - check pm_runtime_get_sync return value, act on it - remove unnecessary initialisers in set_fmt - use udelay(15) in retry code: 10 retries * 15 = 150, so at worst we wait the full i2s register access delay - fix some weird returns to return directly - use __maybe_unused instead of #ifdef CONFIG_PM_SLEEP, also put __maybe_unused on the runtime callbacks - use (foo) instead of foo in header macros for precedence reasons - when using mclk-calibrate, also turn off/on those clocks during suspend and resume operations - remove mclk_tx and mclk_rx reenablement code in remove - move hclk enablement further down the probe, and disable it on probe failure - make reset controls mandatory, since the bindings state this too - use _exclusive for getting the reset controls - change reset assert/deassert delays to both be 10 usec (thank you Sugar Zhang!) - properly prepare and enable all mclks in probe, especially before calling clk_get_rate on them - if registering PCM fails, also use the cleanup error path instead of returning directly - bring back playback and capture only but in the way Sugar Zhang suggested it: set those modes depending on dma-names - rework clock enablement in general. Probe now always enables these, instead of relying on the pm resume thing - add myself to MAINTAINERS for this driver dt bindings: - fix a description still mentioning clk-trcm in the schema - document rockchip,io-multiplex, a property that describes the hardware as having multiplexed I2S GPIOs so direction needs to be changed dynamically - document rockchip,mclk-calibrate, which allows specifying different clocks for the two sample rate bases and switch between them as needed - dma-names now doesn't have a set order and items can be absent to indicate that the controller doesn't support this mode - add myself to MAINTAINERS for these bindings Changes in v2: - remove ad-hoc writeq and needless (and broken) optimisation in reset assert/deassert. This wouldn't have worked on Big Endian, and would've been pointless on any other platform, as the overhead for saving one write was comparatively big - fix various checkpatch issues - get rid of leftover clk-trcm in schema - set status = "okay" in example in schema instead of "disabled" - change dma-names so rx is first, adjust device trees as necessary - properly reference uint32-array for rx-route and tx-route instead of uint32 - replace trcm-sync with two boolean properties, adjust DT changes accordingly and also get rid of the header file - get rid of rockchip,no-dmaengine. This was only needed for some downstream driver and shouldn't be in the DT - get rid of rockchip,capture-only/playback-only. Rationale being that I have no way to test whether they're needed, and unconditionally setting channels_min to 0 breaks everything - change hclk description in "clocks" Nicolas Frattaroli (4): ASoC: rockchip: add support for i2s-tdm controller dt-bindings: sound: add rockchip i2s-tdm binding arm64: dts: rockchip: add i2s1 on rk356x arm64: dts: rockchip: add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 218 ++ MAINTAINERS | 7 + .../boot/dts/rockchip/rk3566-quartz64-a.dts | 35 +- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 26 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1845 +++++++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 398 ++++ 8 files changed, 2541 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h -- 2.33.0

2 3

Sample rate issues with M-Audio M-Track 2x2M
by GitHub issues - edited 31 Aug '21

31 Aug '21

alsa-project/alsa-lib issue #176 was edited from powerjungle: At random points the sample rate changes, but the applications continue to use the previous rate. This results in sound being lower/higher pitch then normal. My own voice when using the application [Mumble](https://github.com/mumble-voip/mumble) consistently becomes higher pitch and sounds pretty funny, friends had a good laugh. I have some diffs of ALSA dumps. My output is to headphones. My input is XLR. Here for example I open the GNOME settings for "Sound" and my output becomes lower pitch: ```diff 6c6 < !!Script ran on: Sun Aug 29 19:56:57 UTC 2021 --- > !!Script ran on: Sun Aug 29 19:58:27 UTC 2021 1287c1287 < Momentary freq = 48000 Hz (0x6.0000) --- > Momentary freq = 44101 Hz (0x5.8338) 1304c1304,1308 < Status: Stop --- > Status: Running > Interface = 2 > Altset = 1 > Packet Size = 72 > Momentary freq = 44100 Hz (0x5.8333) 1346c1350 < crw-rw----+ 1 root audio 116, 12 Aug 29 21:15 /dev/snd/pcmC1D0c --- > crw-rw----+ 1 root audio 116, 12 Aug 29 21:58 /dev/snd/pcmC1D0c 1400c1404 < Subdevices: 1/1 --- > Subdevices: 0/1 2057,2058c2061,2062 < value.0 0 < value.1 0 --- > value.0 3 > value.1 4 ``` Afterwards at a random point the output gets fixed, but the sample rate that gets logged is weird: ```diff 6c6 < !!Script ran on: Sun Aug 29 19:58:27 UTC 2021 --- > !!Script ran on: Sun Aug 29 20:33:14 UTC 2021 78c78 < Running - No --- > Running - Yes 1287c1287 < Momentary freq = 44101 Hz (0x5.8338) --- > Momentary freq = 47999 Hz (0x5.fff8) 1304,1308c1304 < Status: Running < Interface = 2 < Altset = 1 < Packet Size = 72 < Momentary freq = 44100 Hz (0x5.8333) --- > Status: Stop 1351c1347 < crw-rw----+ 1 root audio 116, 11 Aug 29 21:56 /dev/snd/pcmC1D0p --- > crw-rw----+ 1 root audio 116, 11 Aug 29 22:32 /dev/snd/pcmC1D0p 1404c1400 < Subdevices: 0/1 --- > Subdevices: 1/1 2061,2062c2057,2058 < value.0 3 < value.1 4 --- > value.0 0 > value.1 0 ``` Also sometimes when listening to music in Youtube using Firefox, the sound becomes higher pitch, but I cannot reproduce it, can't find the cause. I'm running OpenSUSE Tumbleweed 5.13.13-1-default. Issue URL : https://github.com/alsa-project/alsa-lib/issues/176 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Sample rate issues with M-Audio M-Track 2x2M
by GitHub issues - opened 31 Aug '21

31 Aug '21

alsa-project/alsa-lib issue #176 was opened from powerjungle: At random points the sample rate changes, but the applications continue to use the previous rate. This results in sound being lower/higher pitch then normal. My own voice when using the application [Mumble](https://github.com/mumble-voip/mumble) consistently becomes higher pitch and sounds pretty funny, friends had a good laugh. I have some diffs of ALSA dumps. My output is to headphones. My input is XLR. Here for example I open the GNOME settings for "Sound" and my output becomes lower pitch: ```diff 6c6 < !!Script ran on: Sun Aug 29 19:56:57 UTC 2021 --- > !!Script ran on: Sun Aug 29 19:58:27 UTC 2021 1287c1287 < Momentary freq = 48000 Hz (0x6.0000) --- > Momentary freq = 44101 Hz (0x5.8338) 1304c1304,1308 < Status: Stop --- > Status: Running > Interface = 2 > Altset = 1 > Packet Size = 72 > Momentary freq = 44100 Hz (0x5.8333) 1346c1350 < crw-rw----+ 1 root audio 116, 12 Aug 29 21:15 /dev/snd/pcmC1D0c --- > crw-rw----+ 1 root audio 116, 12 Aug 29 21:58 /dev/snd/pcmC1D0c 1400c1404 < Subdevices: 1/1 --- > Subdevices: 0/1 2057,2058c2061,2062 < value.0 0 < value.1 0 --- > value.0 3 > value.1 4 ``` Afterwards at a random point the output gets fixed, but the sample rate that gets logged is weird: ```diff 6c6 < !!Script ran on: Sun Aug 29 19:58:27 UTC 2021 --- > !!Script ran on: Sun Aug 29 20:33:14 UTC 2021 78c78 < Running - No --- > Running - Yes 1287c1287 < Momentary freq = 44101 Hz (0x5.8338) --- > Momentary freq = 47999 Hz (0x5.fff8) 1304,1308c1304 < Status: Running < Interface = 2 < Altset = 1 < Packet Size = 72 < Momentary freq = 44100 Hz (0x5.8333) --- > Status: Stop 1351c1347 < crw-rw----+ 1 root audio 116, 11 Aug 29 21:56 /dev/snd/pcmC1D0p --- > crw-rw----+ 1 root audio 116, 11 Aug 29 22:32 /dev/snd/pcmC1D0p 1404c1400 < Subdevices: 0/1 --- > Subdevices: 1/1 2061,2062c2057,2058 < value.0 3 < value.1 4 --- > value.0 0 > value.1 0 ``` I'm running OpenSUSE Tumbleweed 5.13.13-1-default. Issue URL : https://github.com/alsa-project/alsa-lib/issues/176 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ASoC: rt5682s: Add driver for ALC5682I-VS codec
by derek.fang＠realtek.com 31 Aug '21

31 Aug '21

From: Derek Fang <derek.fang(a)realtek.com> This is an initial codec driver for Realtek ALC5682I-VS codec. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- include/sound/rt5682s.h | 48 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682s.c | 3231 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5682s.h | 1455 ++++++++++++++++ 5 files changed, 4741 insertions(+) create mode 100644 include/sound/rt5682s.h create mode 100644 sound/soc/codecs/rt5682s.c create mode 100644 sound/soc/codecs/rt5682s.h diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h new file mode 100644 index 000000000000..37ecede2becd --- /dev/null +++ b/include/sound/rt5682s.h @@ -0,0 +1,48 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt5682s.h -- Platform data for RT5682I-VS + * + * Copyright 2021 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT5682S_H +#define __LINUX_SND_RT5682S_H + +enum rt5682s_dmic1_data_pin { + RT5682S_DMIC1_NULL, + RT5682S_DMIC1_DATA_GPIO2, + RT5682S_DMIC1_DATA_GPIO5, +}; + +enum rt5682s_dmic1_clk_pin { + RT5682S_DMIC1_CLK_GPIO1, + RT5682S_DMIC1_CLK_GPIO3, +}; + +enum rt5682s_jd_src { + RT5682S_JD_NULL, + RT5682S_JD1, +}; + +enum rt5682s_dai_clks { + RT5682S_DAI_WCLK_IDX, + RT5682S_DAI_BCLK_IDX, + RT5682S_DAI_NUM_CLKS, +}; + +struct rt5682s_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5682s_dmic1_data_pin dmic1_data_pin; + enum rt5682s_dmic1_clk_pin dmic1_clk_pin; + enum rt5682s_jd_src jd_src; + unsigned int dmic_clk_rate; + unsigned int dmic_delay; + bool dmic_clk_driving_high; + + const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; +}; + +#endif + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..6b3c2cf153db 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,6 +180,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT5677 imply SND_SOC_RT5682_I2C imply SND_SOC_RT5682_SDW + imply SND_SOC_RT5682S imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW imply SND_SOC_RT711_SDCA_SDW @@ -1249,6 +1250,10 @@ config SND_SOC_RT5682_SDW select SND_SOC_RT5682 select REGMAP_SOUNDWIRE +config SND_SOC_RT5682S + tristate + depends on I2C + config SND_SOC_RT700 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..42d00aa4ee46 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -198,6 +198,7 @@ snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt5682-i2c-objs := rt5682-i2c.o +snd-soc-rt5682s-objs := rt5682s.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o @@ -526,6 +527,7 @@ obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_RT5682_I2C) += snd-soc-rt5682-i2c.o obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o +obj-$(CONFIG_SND_SOC_RT5682S) += snd-soc-rt5682s.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT711_SDCA_SDW) += snd-soc-rt711-sdca.o diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c new file mode 100644 index 000000000000..86df1dabf103 --- /dev/null +++ b/sound/soc/codecs/rt5682s.c @@ -0,0 +1,3231 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt5682s.c -- RT5682I-VS ALSA SoC audio component driver +// +// Copyright 2021 Realtek Semiconductor Corp. +// Author: Derek Fang <derek.fang(a)realtek.com> +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5682s.h> + +#include "rt5682s.h" + +#define DEVICE_ID 0x6749 + +static const struct rt5682s_platform_data i2s_default_platform_data = { + .dmic1_data_pin = RT5682S_DMIC1_DATA_GPIO2, + .dmic1_clk_pin = RT5682S_DMIC1_CLK_GPIO3, + .jd_src = RT5682S_JD1, + .dai_clk_names[RT5682S_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682S_DAI_BCLK_IDX] = "rt5682-dai-bclk", +}; + +const char *rt5682s_supply_names[RT5682S_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", +}; + +static const struct reg_sequence patch_list[] = { + {RT5682S_I2C_CTRL, 0x0007}, + {RT5682S_DIG_IN_CTRL_1, 0x0000}, + {RT5682S_CHOP_DAC_2, 0x2020}, + {RT5682S_VREF_REC_OP_FB_CAP_CTRL_2, 0x0101}, + {RT5682S_VREF_REC_OP_FB_CAP_CTRL_1, 0x80c0}, + {RT5682S_HP_CALIB_CTRL_9, 0x0002}, + {RT5682S_DEPOP_1, 0x0000}, + {RT5682S_HP_CHARGE_PUMP_2, 0x3c15}, + {RT5682S_DAC1_DIG_VOL, 0xfefe}, + {RT5682S_SAR_IL_CMD_2, 0xac00}, + {RT5682S_SAR_IL_CMD_3, 0x024c}, + {RT5682S_CBJ_CTRL_6, 0x0804}, +}; + +void rt5682s_apply_patch_list(struct rt5682s_priv *rt5682s, + struct device *dev) +{ + int ret; + + ret = regmap_multi_reg_write(rt5682s->regmap, patch_list, ARRAY_SIZE(patch_list)); + if (ret) + dev_warn(dev, "Failed to apply regmap patch: %d\n", ret); +} + +const struct reg_default rt5682s_reg[] = { + {0x0000, 0x0001}, + {0x0002, 0x8080}, + {0x0003, 0x0001}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x8007}, + {0x000b, 0x0000}, + {0x000f, 0x4000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x0000}, + {0x0013, 0x1200}, + {0x0014, 0x200a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0017, 0x05a4}, + {0x0019, 0xffff}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x08c0}, + {0x0044, 0x1818}, + {0x004b, 0x00c0}, + {0x004c, 0x0000}, + {0x004d, 0x0000}, + {0x0061, 0x00c0}, + {0x0062, 0x008a}, + {0x0063, 0x0800}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x000c}, + {0x0068, 0x0000}, + {0x0069, 0x0000}, + {0x006a, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0810}, + {0x006f, 0xe4de}, + {0x0070, 0x3320}, + {0x0071, 0x0000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x007f, 0x0000}, + {0x0080, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008e, 0x0060}, + {0x008f, 0x4da1}, + {0x0091, 0x1c15}, + {0x0092, 0x0425}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0095, 0x008f}, + {0x0096, 0x0000}, + {0x0097, 0x0000}, + {0x0098, 0x0000}, + {0x0099, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x0000}, + {0x009f, 0x0009}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0000}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6749}, + {0x0100, 0xa000}, + {0x010b, 0x0066}, + {0x010c, 0x6666}, + {0x010d, 0x2202}, + {0x010e, 0x6666}, + {0x010f, 0xa800}, + {0x0110, 0x0006}, + {0x0111, 0x0460}, + {0x0112, 0x2000}, + {0x0113, 0x0200}, + {0x0117, 0x8000}, + {0x0118, 0x0303}, + {0x0125, 0x0020}, + {0x0132, 0x5026}, + {0x0136, 0x8000}, + {0x0139, 0x0005}, + {0x013a, 0x3030}, + {0x013b, 0xa000}, + {0x013c, 0x4110}, + {0x013f, 0x0000}, + {0x0145, 0x0022}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0156, 0x0022}, + {0x0157, 0x0303}, + {0x0158, 0x2222}, + {0x0159, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0001}, + {0x0190, 0x4131}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01ac, 0x0000}, + {0x01ad, 0x0000}, + {0x01ae, 0x0000}, + {0x01af, 0x2000}, + {0x01b0, 0x0000}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0017}, + {0x01b4, 0x004b}, + {0x01b5, 0x0000}, + {0x01b6, 0x03e8}, + {0x01b7, 0x0000}, + {0x01b8, 0x0000}, + {0x01b9, 0x0400}, + {0x01ba, 0xb5b6}, + {0x01bb, 0x9124}, + {0x01bc, 0x4924}, + {0x01bd, 0x0009}, + {0x01be, 0x0018}, + {0x01bf, 0x002a}, + {0x01c0, 0x004c}, + {0x01c1, 0x0097}, + {0x01c2, 0x01c3}, + {0x01c3, 0x03e9}, + {0x01c4, 0x1389}, + {0x01c5, 0xc351}, + {0x01c6, 0x02a0}, + {0x01c7, 0x0b0f}, + {0x01c8, 0x402f}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0000}, + {0x01d8, 0x0162}, + {0x01d9, 0x0007}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0020}, + {0x01e0, 0x04c1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e5, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa004}, + {0x0212, 0x0365}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x021d, 0x024c}, + {0x02fa, 0x0000}, + {0x02fb, 0x0000}, + {0x02fc, 0x0000}, + {0x03fe, 0x0000}, + {0x03ff, 0x0000}, + {0x0500, 0x0000}, + {0x0600, 0x0000}, + {0x0610, 0x6666}, + {0x0611, 0xa9aa}, + {0x0620, 0x6666}, + {0x0621, 0xa9aa}, + {0x0630, 0x6666}, + {0x0631, 0xa9aa}, + {0x0640, 0x6666}, + {0x0641, 0xa9aa}, + {0x07fa, 0x0000}, + {0x08fa, 0x0000}, + {0x08fb, 0x0000}, + {0x0d00, 0x0000}, + {0x1100, 0x0000}, + {0x1101, 0x0000}, + {0x1102, 0x0000}, + {0x1103, 0x0000}, + {0x1104, 0x0000}, + {0x1105, 0x0000}, + {0x1106, 0x0000}, + {0x1107, 0x0000}, + {0x1108, 0x0000}, + {0x1109, 0x0000}, + {0x110a, 0x0000}, + {0x110b, 0x0000}, + {0x110c, 0x0000}, + {0x1111, 0x0000}, + {0x1112, 0x0000}, + {0x1113, 0x0000}, + {0x1114, 0x0000}, + {0x1115, 0x0000}, + {0x1116, 0x0000}, + {0x1117, 0x0000}, + {0x1118, 0x0000}, + {0x1119, 0x0000}, + {0x111a, 0x0000}, + {0x111b, 0x0000}, + {0x111c, 0x0000}, + {0x1401, 0x0404}, + {0x1402, 0x0007}, + {0x1403, 0x0365}, + {0x1404, 0x0210}, + {0x1405, 0x0365}, + {0x1406, 0x0210}, + {0x1407, 0x0000}, + {0x1408, 0x0000}, + {0x1409, 0x0000}, + {0x140a, 0x0000}, + {0x140b, 0x0000}, + {0x140c, 0x0000}, + {0x140d, 0x0000}, + {0x140e, 0x0000}, + {0x140f, 0x0000}, + {0x1410, 0x0000}, + {0x1411, 0x0000}, + {0x1801, 0x0004}, + {0x1802, 0x0000}, + {0x1803, 0x0000}, + {0x1804, 0x0000}, + {0x1805, 0x00ff}, + {0x2c00, 0x0000}, + {0x3400, 0x0200}, + {0x3404, 0x0000}, + {0x3405, 0x0000}, + {0x3406, 0x0000}, + {0x3407, 0x0000}, + {0x3408, 0x0000}, + {0x3409, 0x0000}, + {0x340a, 0x0000}, + {0x340b, 0x0000}, + {0x340c, 0x0000}, + {0x340d, 0x0000}, + {0x340e, 0x0000}, + {0x340f, 0x0000}, + {0x3410, 0x0000}, + {0x3411, 0x0000}, + {0x3412, 0x0000}, + {0x3413, 0x0000}, + {0x3414, 0x0000}, + {0x3415, 0x0000}, + {0x3424, 0x0000}, + {0x3425, 0x0000}, + {0x3426, 0x0000}, + {0x3427, 0x0000}, + {0x3428, 0x0000}, + {0x3429, 0x0000}, + {0x342a, 0x0000}, + {0x342b, 0x0000}, + {0x342c, 0x0000}, + {0x342d, 0x0000}, + {0x342e, 0x0000}, + {0x342f, 0x0000}, + {0x3430, 0x0000}, + {0x3431, 0x0000}, + {0x3432, 0x0000}, + {0x3433, 0x0000}, + {0x3434, 0x0000}, + {0x3435, 0x0000}, + {0x3440, 0x6319}, + {0x3441, 0x3771}, + {0x3500, 0x0002}, + {0x3501, 0x5728}, + {0x3b00, 0x3010}, + {0x3b01, 0x3300}, + {0x3b02, 0x2200}, + {0x3b03, 0x0100}, +}; + +bool rt5682s_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682S_RESET: + case RT5682S_CBJ_CTRL_2: + case RT5682S_I2S1_F_DIV_CTRL_2: + case RT5682S_I2S2_F_DIV_CTRL_2: + case RT5682S_INT_ST_1: + case RT5682S_GPIO_ST: + case RT5682S_IL_CMD_1: + case RT5682S_4BTN_IL_CMD_1: + case RT5682S_AJD1_CTRL: + case RT5682S_VERSION_ID...RT5682S_DEVICE_ID: + case RT5682S_STO_NG2_CTRL_1: + case RT5682S_STO_NG2_CTRL_5...RT5682S_STO_NG2_CTRL_7: + case RT5682S_STO1_DAC_SIL_DET: + case RT5682S_HP_IMP_SENS_CTRL_1...RT5682S_HP_IMP_SENS_CTRL_4: + case RT5682S_HP_IMP_SENS_CTRL_13: + case RT5682S_HP_IMP_SENS_CTRL_14: + case RT5682S_HP_IMP_SENS_CTRL_43...RT5682S_HP_IMP_SENS_CTRL_46: + case RT5682S_HP_CALIB_CTRL_1: + case RT5682S_HP_CALIB_CTRL_10: + case RT5682S_HP_CALIB_ST_1...RT5682S_HP_CALIB_ST_11: + case RT5682S_SAR_IL_CMD_2...RT5682S_SAR_IL_CMD_5: + case RT5682S_SAR_IL_CMD_10: + case RT5682S_SAR_IL_CMD_11: + case RT5682S_VERSION_ID_HIDE: + case RT5682S_VERSION_ID_CUS: + case RT5682S_I2C_TRANS_CTRL: + case RT5682S_DMIC_FLOAT_DET: + case RT5682S_HA_CMP_OP_1: + case RT5682S_NEW_CBJ_DET_CTL_10...RT5682S_NEW_CBJ_DET_CTL_16: + case RT5682S_CLK_SW_TEST_1: + case RT5682S_CLK_SW_TEST_2: + case RT5682S_EFUSE_READ_1...RT5682S_EFUSE_READ_18: + case RT5682S_PILOT_DIG_CTL_1: + return true; + default: + return false; + } +} + +bool rt5682s_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682S_RESET: + case RT5682S_VERSION_ID: + case RT5682S_VENDOR_ID: + case RT5682S_DEVICE_ID: + case RT5682S_HP_CTRL_1: + case RT5682S_HP_CTRL_2: + case RT5682S_HPL_GAIN: + case RT5682S_HPR_GAIN: + case RT5682S_I2C_CTRL: + case RT5682S_CBJ_BST_CTRL: + case RT5682S_CBJ_DET_CTRL: + case RT5682S_CBJ_CTRL_1...RT5682S_CBJ_CTRL_8: + case RT5682S_DAC1_DIG_VOL: + case RT5682S_STO1_ADC_DIG_VOL: + case RT5682S_STO1_ADC_BOOST: + case RT5682S_HP_IMP_GAIN_1: + case RT5682S_HP_IMP_GAIN_2: + case RT5682S_SIDETONE_CTRL: + case RT5682S_STO1_ADC_MIXER: + case RT5682S_AD_DA_MIXER: + case RT5682S_STO1_DAC_MIXER: + case RT5682S_A_DAC1_MUX: + case RT5682S_DIG_INF2_DATA: + case RT5682S_REC_MIXER: + case RT5682S_CAL_REC: + case RT5682S_HP_ANA_OST_CTRL_1...RT5682S_HP_ANA_OST_CTRL_3: + case RT5682S_PWR_DIG_1...RT5682S_PWR_MIXER: + case RT5682S_MB_CTRL: + case RT5682S_CLK_GATE_TCON_1...RT5682S_CLK_GATE_TCON_3: + case RT5682S_CLK_DET...RT5682S_LPF_AD_DMIC: + case RT5682S_I2S1_SDP: + case RT5682S_I2S2_SDP: + case RT5682S_ADDA_CLK_1: + case RT5682S_ADDA_CLK_2: + case RT5682S_I2S1_F_DIV_CTRL_1: + case RT5682S_I2S1_F_DIV_CTRL_2: + case RT5682S_TDM_CTRL: + case RT5682S_TDM_ADDA_CTRL_1: + case RT5682S_TDM_ADDA_CTRL_2: + case RT5682S_DATA_SEL_CTRL_1: + case RT5682S_TDM_TCON_CTRL_1: + case RT5682S_TDM_TCON_CTRL_2: + case RT5682S_GLB_CLK: + case RT5682S_PLL_TRACK_1...RT5682S_PLL_TRACK_6: + case RT5682S_PLL_TRACK_11: + case RT5682S_DEPOP_1: + case RT5682S_HP_CHARGE_PUMP_1: + case RT5682S_HP_CHARGE_PUMP_2: + case RT5682S_HP_CHARGE_PUMP_3: + case RT5682S_MICBIAS_1...RT5682S_MICBIAS_3: + case RT5682S_PLL_TRACK_12...RT5682S_PLL_CTRL_7: + case RT5682S_RC_CLK_CTRL: + case RT5682S_I2S2_M_CLK_CTRL_1: + case RT5682S_I2S2_F_DIV_CTRL_1: + case RT5682S_I2S2_F_DIV_CTRL_2: + case RT5682S_IRQ_CTRL_1...RT5682S_IRQ_CTRL_4: + case RT5682S_INT_ST_1: + case RT5682S_GPIO_CTRL_1: + case RT5682S_GPIO_CTRL_2: + case RT5682S_GPIO_ST: + case RT5682S_HP_AMP_DET_CTRL_1: + case RT5682S_MID_HP_AMP_DET: + case RT5682S_LOW_HP_AMP_DET: + case RT5682S_DELAY_BUF_CTRL: + case RT5682S_SV_ZCD_1: + case RT5682S_SV_ZCD_2: + case RT5682S_IL_CMD_1...RT5682S_IL_CMD_6: + case RT5682S_4BTN_IL_CMD_1...RT5682S_4BTN_IL_CMD_7: + case RT5682S_ADC_STO1_HP_CTRL_1: + case RT5682S_ADC_STO1_HP_CTRL_2: + case RT5682S_AJD1_CTRL: + case RT5682S_JD_CTRL_1: + case RT5682S_DUMMY_1...RT5682S_DUMMY_3: + case RT5682S_DAC_ADC_DIG_VOL1: + case RT5682S_BIAS_CUR_CTRL_2...RT5682S_BIAS_CUR_CTRL_10: + case RT5682S_VREF_REC_OP_FB_CAP_CTRL_1: + case RT5682S_VREF_REC_OP_FB_CAP_CTRL_2: + case RT5682S_CHARGE_PUMP_1: + case RT5682S_DIG_IN_CTRL_1: + case RT5682S_PAD_DRIVING_CTRL: + case RT5682S_CHOP_DAC_1: + case RT5682S_CHOP_DAC_2: + case RT5682S_CHOP_ADC: + case RT5682S_CALIB_ADC_CTRL: + case RT5682S_VOL_TEST: + case RT5682S_SPKVDD_DET_ST: + case RT5682S_TEST_MODE_CTRL_1...RT5682S_TEST_MODE_CTRL_4: + case RT5682S_PLL_INTERNAL_1...RT5682S_PLL_INTERNAL_4: + case RT5682S_STO_NG2_CTRL_1...RT5682S_STO_NG2_CTRL_10: + case RT5682S_STO1_DAC_SIL_DET: + case RT5682S_SIL_PSV_CTRL1: + case RT5682S_SIL_PSV_CTRL2: + case RT5682S_SIL_PSV_CTRL3: + case RT5682S_SIL_PSV_CTRL4: + case RT5682S_SIL_PSV_CTRL5: + case RT5682S_HP_IMP_SENS_CTRL_1...RT5682S_HP_IMP_SENS_CTRL_46: + case RT5682S_HP_LOGIC_CTRL_1...RT5682S_HP_LOGIC_CTRL_3: + case RT5682S_HP_CALIB_CTRL_1...RT5682S_HP_CALIB_CTRL_11: + case RT5682S_HP_CALIB_ST_1...RT5682S_HP_CALIB_ST_11: + case RT5682S_SAR_IL_CMD_1...RT5682S_SAR_IL_CMD_14: + case RT5682S_DUMMY_4...RT5682S_DUMMY_6: + case RT5682S_VERSION_ID_HIDE: + case RT5682S_VERSION_ID_CUS: + case RT5682S_SCAN_CTL: + case RT5682S_HP_AMP_DET: + case RT5682S_BIAS_CUR_CTRL_11: + case RT5682S_BIAS_CUR_CTRL_12: + case RT5682S_BIAS_CUR_CTRL_13: + case RT5682S_BIAS_CUR_CTRL_14: + case RT5682S_BIAS_CUR_CTRL_15: + case RT5682S_BIAS_CUR_CTRL_16: + case RT5682S_BIAS_CUR_CTRL_17: + case RT5682S_BIAS_CUR_CTRL_18: + case RT5682S_I2C_TRANS_CTRL: + case RT5682S_DUMMY_7: + case RT5682S_DUMMY_8: + case RT5682S_DMIC_FLOAT_DET: + case RT5682S_HA_CMP_OP_1...RT5682S_HA_CMP_OP_13: + case RT5682S_HA_CMP_OP_14...RT5682S_HA_CMP_OP_25: + case RT5682S_NEW_CBJ_DET_CTL_1...RT5682S_NEW_CBJ_DET_CTL_16: + case RT5682S_DA_FILTER_1...RT5682S_DA_FILTER_5: + case RT5682S_CLK_SW_TEST_1: + case RT5682S_CLK_SW_TEST_2: + case RT5682S_CLK_SW_TEST_3...RT5682S_CLK_SW_TEST_14: + case RT5682S_EFUSE_MANU_WRITE_1...RT5682S_EFUSE_MANU_WRITE_6: + case RT5682S_EFUSE_READ_1...RT5682S_EFUSE_READ_18: + case RT5682S_EFUSE_TIMING_CTL_1: + case RT5682S_EFUSE_TIMING_CTL_2: + case RT5682S_PILOT_DIG_CTL_1: + case RT5682S_PILOT_DIG_CTL_2: + case RT5682S_HP_AMP_DET_CTL_1...RT5682S_HP_AMP_DET_CTL_4: + return true; + default: + return false; + } +} + +void rt5682s_reset(struct rt5682s_priv *rt5682s) +{ + regmap_write(rt5682s->regmap, RT5682S_RESET, 0); +} + +static int rt5682s_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read(component, RT5682S_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682S_4BTN_IL_CMD_1, val); + dev_dbg(component->dev, "%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_2, + RT5682S_SAR_ADC_PSV_MASK, RT5682S_SAR_ADC_PSV_ENTRY); + + return btn_type; +} + +enum { + SAR_PWR_OFF, + SAR_PWR_NORMAL, + SAR_PWR_SAVING, +}; + +static int rt5682s_sar_power_mode(struct snd_soc_component *component, + int mode, int jd_step) +{ + struct rt5682s_priv *rt5682s = + snd_soc_component_get_drvdata(component); + + mutex_lock(&rt5682s->sar_mutex); + + switch (mode) { + case SAR_PWR_SAVING: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_REG | RT5682S_CTRL_MB2_REG); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK, RT5682S_SAR_BUTDET_EN); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_2, + RT5682S_SAR_ADC_PSV_MASK, RT5682S_SAR_ADC_PSV_ENTRY); + break; + case SAR_PWR_NORMAL: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_FSM | RT5682S_CTRL_MB2_FSM); + if (!jd_step) { + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_SEL_MB1_2_AUTO); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK, + RT5682S_SAR_BUTDET_EN | RT5682S_SAR_BUTDET_POW_NORM); + } + break; + case SAR_PWR_OFF: + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); + break; + default: + break; + } + + mutex_unlock(&rt5682s->sar_mutex); + + return 0; +} + +static void rt5682s_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682S_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK | RT5682S_4BTN_IL_RST_MASK, + RT5682S_4BTN_IL_EN | RT5682S_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682S_IRQ_CTRL_3, + RT5682S_IL_IRQ_MASK, RT5682S_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5682S_IRQ_CTRL_3, + RT5682S_IL_IRQ_MASK, RT5682S_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); + } +} + +/** + * rt5682s_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +int rt5682s_headset_detect(struct snd_soc_component *component, int jack_insert) +{ + unsigned int val, count; + int jack_type = 0; + + if (jack_insert) { + rt5682s_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF1 | RT5682S_PWR_VREF2 | RT5682S_PWR_MB, + RT5682S_PWR_VREF1 | RT5682S_PWR_VREF2 | RT5682S_PWR_MB); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2, + RT5682S_PWR_FV1 | RT5682S_PWR_FV2); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_3, + RT5682S_PWR_CBJ, RT5682S_PWR_CBJ); + snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x0365); + snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, + RT5682S_OSW_L_DIS | RT5682S_OSW_R_DIS); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, + RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 1); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); + usleep_range(45000, 50000); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_HIGH); + + count = 0; + do { + usleep_range(10000, 15000); + val = snd_soc_component_read(component, RT5682S_CBJ_CTRL_2) + & RT5682S_JACK_TYPE_MASK; + count++; + } while (val == 0 && count < 50); + + pr_debug("%s, val=%d, count=%d\n", __func__, val, count); + + switch (val) { + case 0x1: + case 0x2: + jack_type = SND_JACK_HEADSET; + snd_soc_component_write(component, RT5682S_SAR_IL_CMD_3, 0x024c); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_EN); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_MASK, val << RT5682S_SAR_SEL_MB1_2_SFT); + rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 1); + rt5682s_enable_push_button_irq(component, true); + break; + default: + jack_type = SND_JACK_HEADPHONE; + break; + } + snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, + RT5682S_OSW_L_EN | RT5682S_OSW_R_EN); + } else { + rt5682s_sar_power_mode(component, SAR_PWR_OFF, 1); + rt5682s_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); + + if (!snd_soc_dapm_get_pin_status(&component->dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(&component->dapm, "Vref2")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_VREF2, 0); + + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_3, + RT5682S_PWR_CBJ, 0); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_DIS); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_DIS); + jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", jack_type); + + return jack_type; +} + +void rt5682s_jack_detect_handler(struct work_struct *work) +{ + struct rt5682s_priv *rt5682s = + container_of(work, struct rt5682s_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682s->component) + usleep_range(10000, 15000); + + while (!rt5682s->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682s->calibrate_mutex); + + val = snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) + & RT5682S_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682s->jack_type == 0) { + /* jack was out, report jack type */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 1); + rt5682s->irq_work_delay_time = 0; + } else if ((rt5682s->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { + /* jack is already in, report button event */ + rt5682s->jack_type = SND_JACK_HEADSET; + btn_type = rt5682s_button_detect(rt5682s->component); + /** + * rt5682s can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682s->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682s->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682s->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682s->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5682s->component->dev, + "Unexpected button code 0x%04x\n", btn_type); + break; + } + } + } else { + /* jack out */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); + rt5682s->irq_work_delay_time = 50; + } + + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682s->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682s->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + mutex_unlock(&rt5682s->calibrate_mutex); +} + +static void rt5682s_jd_check_handler(struct work_struct *work) +{ + struct rt5682s_priv *rt5682s = + container_of(work, struct rt5682s_priv, jd_check_work.work); + + if (snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) + & RT5682S_JDH_RS_MASK) { + /* jack out */ + rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); + + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682s->jd_check_work, 500); + } +} + +static irqreturn_t rt5682s_irq(int irq, void *data) +{ + struct rt5682s_priv *rt5682s = data; + + mod_delayed_work(system_power_efficient_wq, &rt5682s->jack_detect_work, + msecs_to_jiffies(rt5682s->irq_work_delay_time)); + + return IRQ_HANDLED; +} + +static int rt5682s_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int btndet_delay = 16; + + rt5682s->hs_jack = hs_jack; + + if (!hs_jack) { + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK, RT5682S_JD1_DIS); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_JDH, 0); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + + return 0; + } + + switch (rt5682s->pdata.jd_src) { + case RT5682S_JD1: + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_5, + RT5682S_JD_FAST_OFF_SRC_MASK, RT5682S_JD_FAST_OFF_SRC_JDH); + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_2, + RT5682S_EXT_JD_SRC, RT5682S_EXT_JD_SRC_MANUAL); + regmap_update_bits(rt5682s->regmap, RT5682S_CBJ_CTRL_1, + RT5682S_EMB_JD_MASK | RT5682S_DET_TYPE | + RT5682S_POL_FAST_OFF_MASK | RT5682S_MIC_CAP_MASK, + RT5682S_EMB_JD_EN | RT5682S_DET_TYPE | + RT5682S_POL_FAST_OFF_HIGH | RT5682S_MIC_CAP_HS); + regmap_update_bits(rt5682s->regmap, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_POW_MASK, RT5682S_SAR_POW_EN); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP1_PIN_MASK, RT5682S_GP1_PIN_IRQ); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BGLDO, RT5682S_PWR_BGLDO); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_2, + RT5682S_PWR_JD_MASK, RT5682S_PWR_JD_ENABLE); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_IRQ | RT5682S_POW_JDH, RT5682S_POW_IRQ | RT5682S_POW_JDH); + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK | RT5682S_JD1_POL_MASK, + RT5682S_JD1_EN | RT5682S_JD1_POL_NOR); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_4, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_5, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_6, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + regmap_update_bits(rt5682s->regmap, RT5682S_4BTN_IL_CMD_7, + RT5682S_4BTN_IL_HOLD_WIN_MASK | RT5682S_4BTN_IL_CLICK_WIN_MASK, + (btndet_delay << RT5682S_4BTN_IL_HOLD_WIN_SFT | btndet_delay)); + + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682S_JD_NULL: + regmap_update_bits(rt5682s->regmap, RT5682S_IRQ_CTRL_2, + RT5682S_JD1_EN_MASK, RT5682S_JD1_DIS); + regmap_update_bits(rt5682s->regmap, RT5682S_RC_CLK_CTRL, + RT5682S_POW_JDH, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9450, 150, 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); + +static const char * const rt5682s_sar_mode[] = { + "Normal", "Saving" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_sar_mode_enum, 0, 0, + rt5682s_sar_mode); + +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), + + /* CBJ Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682S_REC_MIXER, + RT5682S_BST_CBJ_SFT, 35, 0, cbj_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE_SFT, RT5682S_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_VOL_SFT + 1, RT5682S_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682S_STO1_ADC_BOOST, + RT5682S_STO1_ADC_L_BST_SFT, RT5682S_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), +}; + +/** + * rt5682s_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682S can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + switch (clk_src) { + case RT5682S_CLK_SEL_SYS: + case RT5682S_CLK_SEL_I2S1_ASRC: + case RT5682S_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682S_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_2, + RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682S_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_3, + RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682s_sel_asrc_clk_src); + +static int rt5682s_div_sel(struct rt5682s_priv *rt5682s, + int target, const int div[], int size) +{ + int i; + + if (rt5682s->sysclk < target) { + dev_err(rt5682s->component->dev, + "sysclk rate %d is too low\n", rt5682s->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + dev_dbg(rt5682s->component->dev, "div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682s->sysclk) + return i; + if (target * div[i + 1] > rt5682s->sysclk) { + dev_dbg(rt5682s->component->dev, + "can't find div for sysclk %d\n", rt5682s->sysclk); + return i; + } + } + + if (target * div[i] < rt5682s->sysclk) + dev_err(rt5682s->component->dev, + "sysclk rate %d is too high\n", rt5682s->sysclk); + + return size - 1; +} + +static int get_clk_info(int sclk, int rate) +{ + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(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 rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int idx, dmic_clk_rate = 3072000; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + if (rt5682s->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682s->pdata.dmic_clk_rate; + + idx = rt5682s_div_sel(rt5682s, dmic_clk_rate, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_CLK_MASK, idx << RT5682S_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(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 rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int ref, val, reg, idx; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read(component, RT5682S_GPIO_CTRL_1) + & RT5682S_GP4_PIN_MASK; + + if (w->shift == RT5682S_PWR_ADC_S1F_BIT && val == RT5682S_GP4_PIN_ADCDAT2) + ref = 256 * rt5682s->lrck[RT5682S_AIF2]; + else + ref = 256 * rt5682s->lrck[RT5682S_AIF1]; + + idx = rt5682s_div_sel(rt5682s, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682S_PWR_ADC_S1F_BIT) + reg = RT5682S_PLL_TRACK_3; + else + reg = RT5682S_PLL_TRACK_2; + + snd_soc_component_update_bits(component, reg, + RT5682S_FILTER_CLK_DIV_MASK, idx << RT5682S_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682s->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_1, + RT5682S_ADC_OSR_MASK | RT5682S_DAC_OSR_MASK, + (idx << RT5682S_ADC_OSR_SFT) | (idx << RT5682S_DAC_OSR_SFT)); + + return 0; +} + +static int set_dmic_power(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 rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int delay = 50, val; + + if (rt5682s->pdata.dmic_delay) + delay = rt5682s->pdata.dmic_delay; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = (snd_soc_component_read(component, RT5682S_GLB_CLK) + & RT5682S_SCLK_SRC_MASK) >> RT5682S_SCLK_SRC_SFT; + if (val == RT5682S_CLK_SRC_PLL1 || val == RT5682S_CLK_SRC_PLL2) + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_MB, + RT5682S_PWR_VREF2 | RT5682S_PWR_MB); + + /*Add delay to avoid pop noise*/ + msleep(delay); + break; + + case SND_SOC_DAPM_POST_PMD: + if (!rt5682s->jack_type) { + if (!snd_soc_dapm_get_pin_status(w->dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(w->dapm, "Vref2")) + snd_soc_component_update_bits(component, + RT5682S_PWR_ANLG_1, RT5682S_PWR_VREF2, 0); + } + break; + } + + return 0; +} + +static int set_i2s_clk(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 rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + int pre_div, id; + unsigned int reg, mask, sft; + + if (event != SND_SOC_DAPM_PRE_PMU) + return 0; + + if (w->shift == RT5682S_PWR_I2S2_BIT) { + id = RT5682S_AIF2; + reg = RT5682S_I2S2_M_CLK_CTRL_1; + mask = RT5682S_I2S2_M_D_MASK; + sft = RT5682S_I2S2_M_D_SFT; + } else { + id = RT5682S_AIF1; + reg = RT5682S_ADDA_CLK_1; + mask = RT5682S_I2S_M_D_MASK; + sft = RT5682S_I2S_M_D_SFT; + } + + if (!rt5682s->master[id]) + return 0; + + pre_div = get_clk_info(rt5682s->sysclk, rt5682s->lrck[id]); + if (pre_div < 0) { + dev_err(component->dev, "get pre_div failed\n"); + return -EINVAL; + } + + dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d master\n", + rt5682s->lrck[id], pre_div, id); + snd_soc_component_update_bits(component, reg, mask, pre_div << sft); + + return 0; +} + +static int is_sys_clk_from_plla(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if ((rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL1) || + (rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL2 && + rt5682s->pll_comb == USE_PLLAB)) + return 1; + + return 0; +} + +static int is_sys_clk_from_pllb(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if (rt5682s->sysclk_src == RT5682S_CLK_SRC_PLL2) + return 1; + + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, sft, val; + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682S_ADC_STO1_ASRC_SFT: + reg = RT5682S_PLL_TRACK_3; + sft = RT5682S_FILTER_CLK_SEL_SFT; + break; + case RT5682S_DAC_STO1_ASRC_SFT: + reg = RT5682S_PLL_TRACK_2; + sft = RT5682S_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read(component, reg) >> sft) & 0xf; + switch (val) { + case RT5682S_CLK_SEL_I2S1_ASRC: + case RT5682S_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } +} + +static int is_headset_type(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + if ((rt5682s->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) + return 1; + + return 0; +} + +static int rt5682s_hp_amp_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); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN); + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(30000, 35000); + snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_11, 0x6666); + snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_12, 0xa82a); + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, + RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | + RT5682S_HPO_SEL_IP_EN_SW, RT5682S_HPO_L_PATH_EN | + RT5682S_HPO_R_PATH_EN | RT5682S_HPO_IP_EN_GATING); + snd_soc_component_write(component, RT5682S_HP_AMP_DET_CTL_1, 0x3050); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, + RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | + RT5682S_HPO_SEL_IP_EN_SW, 0); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | + RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, 0); + snd_soc_component_update_bits(component, RT5682S_DEPOP_1, + RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, 0); + break; + } + + return 0; +} + +static int sar_power_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); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 0); + break; + case SND_SOC_DAPM_POST_PMD: + rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 0); + break; + } + + return 0; +} + +/* Interface data select */ +static const char * const rt5682s_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_if2_adc_enum, + RT5682S_DIG_INF2_DATA, RT5682S_IF2_ADC_SEL_SFT, + rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_01_adc_enum, + RT5682S_TDM_ADDA_CTRL_1, RT5682S_IF1_ADC1_SEL_SFT, + rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_23_adc_enum, + RT5682S_TDM_ADDA_CTRL_1, RT5682S_IF1_ADC2_SEL_SFT, + rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_45_adc_enum, + RT5682S_TDM_ADDA_CTRL_1, RT5682S_IF1_ADC3_SEL_SFT, + rt5682s_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682s_if1_67_adc_enum, + RT5682S_TDM_ADDA_CTRL_1, RT5682S_IF1_ADC4_SEL_SFT, + rt5682s_data_select); + +static const struct snd_kcontrol_new rt5682s_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682s_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682s_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682s_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682s_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682s_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682s_if1_67_adc_enum); + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682s_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682S_STO1_ADC_MIXER, + RT5682S_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682S_AD_DA_MIXER, + RT5682S_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682S_STO1_DAC_MIXER, + RT5682S_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682s_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682S_REC_MIXER, + RT5682S_M_CBJ_RM1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682s_rec1_r_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682S_REC_MIXER, + RT5682S_M_CBJ_RM1_R_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682s_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682s_sto1_adc1l_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC1L_SRC_SFT, rt5682s_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682s_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682s_sto1_adc1r_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC1R_SRC_SFT, rt5682s_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682s_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682s_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682s_sto1_adcl_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADCL_SRC_SFT, rt5682s_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682s_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682s_sto1_adcr_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADCR_SRC_SFT, rt5682s_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682s_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682s_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682s_sto1_adc2l_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC2L_SRC_SFT, rt5682s_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682s_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682s_sto1_adc2r_enum, RT5682S_STO1_ADC_MIXER, + RT5682S_STO1_ADC2R_SRC_SFT, rt5682s_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682s_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682s_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682s_if1_adc_slot_values[] = { + 0, 2, 4, 6, +}; + +static const char * const rt5682s_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_if1_adc_slot_enum, + RT5682S_TDM_CTRL, RT5682S_TDM_ADC_LCA_SFT, RT5682S_TDM_ADC_LCA_MASK, + rt5682s_if1_adc_slot_src, rt5682s_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682s_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682s_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682s_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682s_alg_dac_l1_enum, RT5682S_A_DAC1_MUX, + RT5682S_A_DACL1_SFT, rt5682s_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682s_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682s_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682s_alg_dac_r1_enum, RT5682S_A_DAC1_MUX, + RT5682S_A_DACR1_SFT, rt5682s_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682s_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682s_alg_dac_r1_enum); + +static const unsigned int rt5682s_adcdat_pin_values[] = { + 1, 3, +}; + +static const char * const rt5682s_adcdat_pin_select[] = { + "ADCDAT1", "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, + RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, RT5682S_GP4_PIN_MASK, + rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682s_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682s_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO MB1", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO MB2", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_MB2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO", RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PLL Powers */ + SND_SOC_DAPM_SUPPLY_S("PLLA_LDO", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_LDO", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_LDO_PLLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLA_BIAS", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BIAS_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_BIAS", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_BIAS_PLLB_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLA", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB", 0, RT5682S_PWR_ANLG_3, + RT5682S_PWR_PLLB_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("PLLA_RST", 1, RT5682S_PWR_ANLG_3, + RT5682S_RSTB_PLLA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("PLLB_RST", 1, RT5682S_PWR_ANLG_3, + RT5682S_RSTB_PLLB_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682S_PLL_TRACK_1, + RT5682S_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_EN_SFT, 0, set_dmic_power, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682s_rec1_l_mix, + ARRAY_SIZE(rt5682s_rec1_l_mix)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5682s_rec1_r_mix, + ARRAY_SIZE(rt5682s_rec1_r_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682S_CAL_REC, + RT5682S_PWR_RM1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5682S_CAL_REC, + RT5682S_PWR_RM1_R_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682S_PWR_DIG_1, + RT5682S_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682S_PWR_DIG_1, + RT5682S_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682S_CHOP_ADC, + RT5682S_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682S_PWR_DIG_2, + RT5682S_PWR_ADC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE_SFT, 1, rt5682s_sto1_adc_l_mix, + ARRAY_SIZE(rt5682s_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682S_STO1_ADC_DIG_VOL, + RT5682S_R_MUTE_SFT, 1, rt5682s_sto1_adc_r_mix, + ARRAY_SIZE(rt5682s_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682S_PWR_DIG_1, RT5682S_PWR_I2S1_BIT, + 0, set_i2s_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682S_PWR_DIG_1, RT5682S_PWR_I2S2_BIT, + 0, set_i2s_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5682s_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5682S_I2S1_SDP, RT5682S_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5682S_I2S2_SDP, RT5682S_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, + SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_dac_l_mix, ARRAY_SIZE(rt5682s_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682s_dac_r_mix, ARRAY_SIZE(rt5682s_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5682s_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5682s_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682S_PWR_DIG_2, + RT5682S_PWR_DAC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_dac_l_mix, ARRAY_SIZE(rt5682s_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_dac_r_mix, ARRAY_SIZE(rt5682s_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682S_PWR_DIG_1, + RT5682S_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682S_PWR_DIG_1, + RT5682S_PWR_DAC_R1_BIT, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682s_hp_amp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, + RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682S_CLK_DET, + RT5682S_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MCLK0 DET PWR", RT5682S_PWR_ANLG_2, + RT5682S_PWR_MCLK0_WD_BIT, 0, NULL, 0), + + /* SAR */ + SND_SOC_DAPM_SUPPLY("SAR", SND_SOC_NOPM, 0, 0, sar_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLLA", is_sys_clk_from_plla}, + {"ADC Stereo1 Filter", NULL, "PLLB", is_sys_clk_from_pllb}, + {"DAC Stereo1 Filter", NULL, "PLLA", is_sys_clk_from_plla}, + {"DAC Stereo1 Filter", NULL, "PLLB", is_sys_clk_from_pllb}, + {"PLLA", NULL, "PLLA_LDO"}, + {"PLLA", NULL, "PLLA_BIAS"}, + {"PLLA", NULL, "PLLA_RST"}, + {"PLLB", NULL, "PLLB_LDO"}, + {"PLLB", NULL, "PLLB_BIAS"}, + {"PLLB", NULL, "PLLB_RST"}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + + {"CLKDET SYS", NULL, "MCLK0 DET PWR"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "SAR", is_headset_type}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + {"RECMIX1R", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1R", NULL, "RECMIX1R Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + {"ADC1 R", NULL, "RECMIX1R"}, + {"ADC1 R", NULL, "ADC1 R Power"}, + {"ADC1 R", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "I2S1"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "SAR", is_headset_type}, + + {"HPOL", NULL, "HP Amp"}, + {"HPOR", NULL, "HP Amp"}, +}; + +static int rt5682s_set_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; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, + RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, RT5682S_TDM_EN); + else + snd_soc_component_update_bits(component, + RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682S_TDM_TX_CH_4; + val |= RT5682S_TDM_RX_CH_4; + break; + case 6: + val |= RT5682S_TDM_TX_CH_6; + val |= RT5682S_TDM_RX_CH_6; + break; + case 8: + val |= RT5682S_TDM_TX_CH_8; + val |= RT5682S_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_TDM_CTRL, + RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682S_I2S1_TX_CHL_8 | RT5682S_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682S_TDM_CL_16; + cl = RT5682S_I2S1_TX_CHL_16 | RT5682S_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682S_TDM_CL_20; + cl = RT5682S_I2S1_TX_CHL_20 | RT5682S_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682S_TDM_CL_24; + cl = RT5682S_I2S1_TX_CHL_24 | RT5682S_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682S_TDM_CL_32; + cl = RT5682S_I2S1_TX_CHL_32 | RT5682S_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_TX_CHL_MASK | RT5682S_I2S1_RX_CHL_MASK, cl); + + return 0; +} + +static int rt5682s_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 rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int frame_size; + + rt5682s->lrck[dai->id] = params_rate(params); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682S_I2S1_DL_20; + len_2 |= RT5682S_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682S_I2S1_DL_24; + len_2 |= RT5682S_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682S_I2S1_DL_32; + len_2 |= RT5682S_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682S_I2S2_DL_8; + len_2 |= RT5682S_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682S_AIF1: + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_DL_MASK, len_1); + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_MONO_MASK, RT5682S_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S1_MONO_MASK, RT5682S_I2S1_MONO_DIS); + break; + case RT5682S_AIF2: + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_DL_MASK, len_2); + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MONO_MASK, RT5682S_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MONO_MASK, RT5682S_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682s->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682s->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682S_I2S_BP_INV; + tdm_ctrl |= RT5682S_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682S_AIF1) + tdm_ctrl |= RT5682S_TDM_S_LP_INV | RT5682S_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682S_AIF1) + tdm_ctrl |= RT5682S_TDM_S_BP_INV | RT5682S_TDM_S_LP_INV | + RT5682S_TDM_M_BP_INV | RT5682S_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682S_I2S_DF_LEFT; + tdm_ctrl |= RT5682S_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682S_I2S_DF_PCM_A; + tdm_ctrl |= RT5682S_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682S_I2S_DF_PCM_B; + tdm_ctrl |= RT5682S_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682S_AIF1: + snd_soc_component_update_bits(component, RT5682S_I2S1_SDP, + RT5682S_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_MS_MASK | RT5682S_TDM_S_BP_MASK | + RT5682S_TDM_DF_MASK | RT5682S_TDM_M_BP_MASK | + RT5682S_TDM_M_LP_MASK | RT5682S_TDM_S_LP_MASK, + tdm_ctrl | rt5682s->master[dai->id]); + break; + case RT5682S_AIF2: + if (rt5682s->master[dai->id] == 0) + reg_val |= RT5682S_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682S_I2S2_SDP, + RT5682S_I2S2_MS_MASK | RT5682S_I2S_BP_MASK | + RT5682S_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682s_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int src = 0; + + if (freq == rt5682s->sysclk && clk_id == rt5682s->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682S_SCLK_S_MCLK: + src = RT5682S_CLK_SRC_MCLK; + break; + case RT5682S_SCLK_S_PLL1: + src = RT5682S_CLK_SRC_PLL1; + break; + case RT5682S_SCLK_S_PLL2: + src = RT5682S_CLK_SRC_PLL2; + break; + case RT5682S_SCLK_S_RCCLK: + src = RT5682S_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_SCLK_SRC_MASK, src << RT5682S_SCLK_SRC_SFT); + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_1, + RT5682S_I2S_M_CLK_SRC_MASK, src << RT5682S_I2S_M_CLK_SRC_SFT); + snd_soc_component_update_bits(component, RT5682S_I2S2_M_CLK_CTRL_1, + RT5682S_I2S2_M_CLK_SRC_MASK, src << RT5682S_I2S2_M_CLK_SRC_SFT); + + rt5682s->sysclk = freq; + rt5682s->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static const struct pll_calc_map plla_table[] = { + {2048000, 24576000, 0, 46, 2, true, false, false, false}, + {256000, 24576000, 0, 382, 2, true, false, false, false}, + {512000, 24576000, 0, 190, 2, true, false, false, false}, + {4096000, 24576000, 0, 22, 2, true, false, false, false}, + {1024000, 24576000, 0, 94, 2, true, false, false, false}, + {11289600, 22579200, 1, 22, 2, false, false, false, false}, + {1411200, 22579200, 0, 62, 2, true, false, false, false}, + {2822400, 22579200, 0, 30, 2, true, false, false, false}, + {12288000, 24576000, 1, 22, 2, false, false, false, false}, + {1536000, 24576000, 0, 62, 2, true, false, false, false}, + {3072000, 24576000, 0, 30, 2, true, false, false, false}, + {24576000, 49152000, 4, 22, 0, false, false, false, false}, + {3072000, 49152000, 0, 30, 0, true, false, false, false}, + {6144000, 49152000, 0, 30, 0, false, false, false, false}, + {49152000, 98304000, 10, 22, 0, false, true, false, false}, + {6144000, 98304000, 0, 30, 0, false, true, false, false}, + {12288000, 98304000, 1, 22, 0, false, true, false, false}, + {48000000, 3840000, 10, 22, 23, false, false, false, false}, + {24000000, 3840000, 4, 22, 23, false, false, false, false}, + {19200000, 3840000, 3, 23, 23, false, false, false, false}, + {38400000, 3840000, 8, 23, 23, false, false, false, false}, +}; + +static const struct pll_calc_map pllb_table[] = { + {48000000, 24576000, 8, 6, 3, false, false, false, false}, + {48000000, 22579200, 23, 12, 3, false, false, false, true}, + {24000000, 24576000, 3, 6, 3, false, false, false, false}, + {24000000, 22579200, 23, 26, 3, false, false, false, true}, + {19200000, 24576000, 2, 6, 3, false, false, false, false}, + {19200000, 22579200, 3, 5, 3, false, false, false, true}, + {38400000, 24576000, 6, 6, 3, false, false, false, false}, + {38400000, 22579200, 8, 5, 3, false, false, false, true}, + {3840000, 49152000, 0, 6, 0, true, false, false, false}, +}; + +static int find_pll_inter_combination(unsigned int f_in, unsigned int f_out, + struct pll_calc_map *a, struct pll_calc_map *b) +{ + int i, j; + + /* Look at PLLA table */ + for (i = 0; i < ARRAY_SIZE(plla_table); i++) { + if (plla_table[i].freq_in == f_in && + plla_table[i].freq_out == f_out) { + memcpy(a, plla_table + i, sizeof(*a)); + return USE_PLLA; + } + } + + /* Look at PLLB table */ + for (i = 0; i < ARRAY_SIZE(pllb_table); i++) { + if (pllb_table[i].freq_in == f_in && + pllb_table[i].freq_out == f_out) { + memcpy(b, pllb_table + i, sizeof(*b)); + return USE_PLLB; + } + } + + /* Find a combination of PLLA & PLLB */ + for (i = ARRAY_SIZE(plla_table) - 1; i >= 0; i--) { + if (plla_table[i].freq_in == f_in && + plla_table[i].freq_out == 3840000) { + for (j = ARRAY_SIZE(pllb_table) - 1; j >= 0; j--) { + if (pllb_table[j].freq_in == 3840000 && + pllb_table[j].freq_out == f_out) { + memcpy(a, plla_table + i, sizeof(*a)); + memcpy(b, pllb_table + j, sizeof(*b)); + return USE_PLLAB; + } + } + } + } + + return -EINVAL; +} + +static int rt5682s_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct pll_calc_map a_map, b_map; + + if (source == rt5682s->pll_src[pll_id] && + freq_in == rt5682s->pll_in[pll_id] && + freq_out == rt5682s->pll_out[pll_id]) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + rt5682s->pll_in[pll_id] = 0; + rt5682s->pll_out[pll_id] = 0; + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_SCLK_SRC_MASK, RT5682S_CLK_SRC_MCLK << RT5682S_SCLK_SRC_SFT); + return 0; + } + + switch (source) { + case RT5682S_PLL_S_MCLK: + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_PLL_SRC_MASK, RT5682S_PLL_SRC_MCLK); + break; + case RT5682S_PLL_S_BCLK1: + snd_soc_component_update_bits(component, RT5682S_GLB_CLK, + RT5682S_PLL_SRC_MASK, RT5682S_PLL_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + rt5682s->pll_comb = find_pll_inter_combination(freq_in, freq_out, + &a_map, &b_map); + + if ((pll_id == RT5682S_PLL1 && rt5682s->pll_comb == USE_PLLA) || + (pll_id == RT5682S_PLL2 && (rt5682s->pll_comb == USE_PLLB || + rt5682s->pll_comb == USE_PLLAB))) { + dev_dbg(component->dev, + "Supported freq conversion for PLL%d:(%d->%d): %d\n", + pll_id + 1, freq_in, freq_out, rt5682s->pll_comb); + } else { + dev_err(component->dev, + "Unsupported freq conversion for PLL%d:(%d->%d): %d\n", + pll_id + 1, freq_in, freq_out, rt5682s->pll_comb); + return -EINVAL; + } + + if (rt5682s->pll_comb == USE_PLLA || rt5682s->pll_comb == USE_PLLAB) { + dev_dbg(component->dev, + "PLLA: fin=%d fout=%d m_bp=%d k_bp=%d m=%d n=%d k=%d\n", + a_map.freq_in, a_map.freq_out, + a_map.m_bp, a_map.k_bp, + (a_map.m_bp ? 0 : a_map.m), + a_map.n, (a_map.k_bp ? 0 : a_map.k)); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_1, + RT5682S_PLLA_N_MASK, a_map.n); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_2, + RT5682S_PLLA_M_MASK | RT5682S_PLLA_K_MASK, + a_map.m << RT5682S_PLLA_M_SFT | a_map.k); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_6, + RT5682S_PLLA_M_BP_MASK | RT5682S_PLLA_K_BP_MASK, + a_map.m_bp << RT5682S_PLLA_M_BP_SFT | + a_map.k_bp << RT5682S_PLLA_K_BP_SFT); + } + + if (rt5682s->pll_comb == USE_PLLB || rt5682s->pll_comb == USE_PLLAB) { + dev_dbg(component->dev, + "PLLB: fin=%d fout=%d m_bp=%d k_bp=%d m=%d n=%d k=%d byp_ps=%d sel_ps=%d\n", + b_map.freq_in, b_map.freq_out, + b_map.m_bp, b_map.k_bp, + (b_map.m_bp ? 0 : b_map.m), + b_map.n, (b_map.k_bp ? 0 : b_map.k), + b_map.byp_ps, b_map.sel_ps); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_3, + RT5682S_PLLB_N_MASK, b_map.n); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_4, + RT5682S_PLLB_M_MASK | RT5682S_PLLB_K_MASK, + b_map.m << RT5682S_PLLB_M_SFT | b_map.k); + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_6, + RT5682S_PLLB_SEL_PS_MASK | RT5682S_PLLB_BYP_PS_MASK | + RT5682S_PLLB_M_BP_MASK | RT5682S_PLLB_K_BP_MASK, + b_map.sel_ps << RT5682S_PLLB_SEL_PS_SFT | + b_map.byp_ps << RT5682S_PLLB_BYP_PS_SFT | + b_map.m_bp << RT5682S_PLLB_M_BP_SFT | + b_map.k_bp << RT5682S_PLLB_K_BP_SFT); + } + + if (rt5682s->pll_comb == USE_PLLB) + snd_soc_component_update_bits(component, RT5682S_PLL_CTRL_7, + RT5682S_PLLB_SRC_MASK, RT5682S_PLLB_SRC_DFIN); + + rt5682s->pll_in[pll_id] = freq_in; + rt5682s->pll_out[pll_id] = freq_out; + rt5682s->pll_src[pll_id] = source; + + return 0; +} + +static int rt5682s_set_bclk1_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s->bclk[dai->id] = ratio; + + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682S_TDM_TCON_CTRL_1, + RT5682S_TDM_BCLK_MS1_MASK, RT5682S_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_2, + RT5682S_I2S2_BCLK_MS2_MASK, RT5682S_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682S_ADDA_CLK_2, + RT5682S_I2S2_BCLK_MS2_MASK, RT5682S_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682s_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_PWR_LDO, RT5682S_PWR_LDO); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_DIG_GATE_CTRL, RT5682S_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_1, + RT5682S_DIG_GATE_CTRL | RT5682S_PWR_LDO, 0); + break; + case SND_SOC_BIAS_ON: + break; + } + + return 0; +} + +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_48 48000 +#define CLK_44 44100 + +static bool rt5682s_clk_check(struct rt5682s_priv *rt5682s) +{ + if (!rt5682s->master[RT5682S_AIF1]) { + dev_dbg(rt5682s->component->dev, "dai clk fmt not set correctly\n"); + return false; + } + return true; +} + +static int rt5682s_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_MB, RT5682S_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_FV2, RT5682S_PWR_VREF2); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_FV2, RT5682S_PWR_FV2); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + /* Only need to power PLLB due to the rate set restriction */ + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLLB"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682s_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682s_clk_check(rt5682s)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); + if (!rt5682s->jack_type) + snd_soc_component_update_bits(component, RT5682S_PWR_ANLG_1, + RT5682S_PWR_VREF2 | RT5682S_PWR_FV2 | RT5682S_PWR_MB, 0); + + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLLB"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682s_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + const char * const clk_name = clk_hw_get_name(hw); + + if (!rt5682s_clk_check(rt5682s)) + return 0; + /* + * Only accept to set wclk rate to 44.1k or 48kHz. + */ + if (rt5682s->lrck[RT5682S_AIF1] != CLK_48 && + rt5682s->lrck[RT5682S_AIF1] != CLK_44) { + dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + __func__, clk_name, CLK_44, CLK_48); + return 0; + } + + return rt5682s->lrck[RT5682S_AIF1]; +} + +static long rt5682s_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + const char * const clk_name = clk_hw_get_name(hw); + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + /* + * Only accept to set wclk rate to 44.1k or 48kHz. + * It will force to 48kHz if not both. + */ + if (rate != CLK_48 && rate != CLK_44) { + dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + __func__, clk_name, CLK_44, CLK_48); + rate = CLK_48; + } + + return rate; +} + +static int rt5682s_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct clk *parent_clk; + const char * const clk_name = clk_hw_get_name(hw); + unsigned int clk_pll2_fout; + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + /* + * Whether the wclk's parent clk (mclk) exists or not, please ensure + * it is fixed or set to 48MHz before setting wclk rate. It's a + * temporary limitation. Only accept 48MHz clk as the clk provider. + * + * It will set the codec anyway by assuming mclk is 48MHz. + */ + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + /* + * To achieve the rate conversion from 48MHz to 44.1k or 48kHz, + * PLL2 is needed. + */ + clk_pll2_fout = rate * 512; + rt5682s_set_component_pll(component, RT5682S_PLL2, RT5682S_PLL_S_MCLK, + CLK_PLL2_FIN, clk_pll2_fout); + + rt5682s_set_component_sysclk(component, RT5682S_SCLK_S_PLL2, 0, + clk_pll2_fout, SND_SOC_CLOCK_IN); + + rt5682s->lrck[RT5682S_AIF1] = rate; + + return 0; +} + +static unsigned long rt5682s_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + unsigned int bclks_per_wclk; + + bclks_per_wclk = snd_soc_component_read(component, RT5682S_TDM_TCON_CTRL_1); + + switch (bclks_per_wclk & RT5682S_TDM_BCLK_MS1_MASK) { + case RT5682S_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682S_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682S_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682S_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682s_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682s_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682s_clk_check(rt5682s)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682s_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682s_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682s_priv *rt5682s = + container_of(hw, struct rt5682s_priv, + dai_clks_hw[RT5682S_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682s->component; + struct snd_soc_dai *dai; + unsigned long factor; + + if (!rt5682s_clk_check(rt5682s)) + return -EINVAL; + + factor = rt5682s_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682S_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682S_AIF1); + return -ENODEV; + } + + return rt5682s_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682s_dai_clk_ops[RT5682S_DAI_NUM_CLKS] = { + [RT5682S_DAI_WCLK_IDX] = { + .prepare = rt5682s_wclk_prepare, + .unprepare = rt5682s_wclk_unprepare, + .recalc_rate = rt5682s_wclk_recalc_rate, + .round_rate = rt5682s_wclk_round_rate, + .set_rate = rt5682s_wclk_set_rate, + }, + [RT5682S_DAI_BCLK_IDX] = { + .recalc_rate = rt5682s_bclk_recalc_rate, + .round_rate = rt5682s_bclk_round_rate, + .set_rate = rt5682s_bclk_set_rate, + }, +}; + +static int rt5682s_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct rt5682s_platform_data *pdata = &rt5682s->pdata; + struct clk_hw *dai_clk_hw; + int i, ret; + + for (i = 0; i < RT5682S_DAI_NUM_CLKS; ++i) { + struct clk_init_data init = { }; + + dai_clk_hw = &rt5682s->dai_clks_hw[i]; + + switch (i) { + case RT5682S_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682s->mclk) { + init.parent_data = &(struct clk_parent_data){ + .fw_name = "mclk", + }; + init.num_parents = 1; + } + break; + case RT5682S_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + init.parent_hws = &(const struct clk_hw *){ + &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX] + }; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + return -EINVAL; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682s_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + ret = devm_clk_hw_register(dev, dai_clk_hw); + if (ret) { + dev_warn(dev, "Failed to register %s: %d\n", + init.name, ret); + return ret; + } + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + dai_clk_hw); + } else { + ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, + init.name, + dev_name(dev)); + if (ret) + return ret; + } + } + + return 0; +} +#endif /* CONFIG_COMMON_CLK */ + +static int rt5682s_probe(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; + +#ifdef CONFIG_COMMON_CLK + int ret; +#endif + rt5682s->component = component; + +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682s->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682s->mclk)) { + if (PTR_ERR(rt5682s->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682s->mclk); + return ret; + } + rt5682s->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682s_register_dai_clks(component); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682s->lrck[RT5682S_AIF1] = CLK_48; +#endif + + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "Vref2"); + snd_soc_dapm_sync(dapm); + return 0; +} + +static void rt5682s_remove(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + rt5682s_reset(rt5682s); +} + +#ifdef CONFIG_PM +static int rt5682s_suspend(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + if (rt5682s->hs_jack && rt5682s->jack_type == SND_JACK_HEADSET) + snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, + RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + + regcache_cache_only(rt5682s->regmap, true); + regcache_mark_dirty(rt5682s->regmap); + + return 0; +} + +static int rt5682s_resume(struct snd_soc_component *component) +{ + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682s->regmap, false); + regcache_sync(rt5682s->regmap); + + if (rt5682s->hs_jack) { + rt5682s->jack_type = 0; + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} +#else +#define rt5682s_suspend NULL +#define rt5682s_resume NULL +#endif + +const struct snd_soc_dai_ops rt5682s_aif1_dai_ops = { + .hw_params = rt5682s_hw_params, + .set_fmt = rt5682s_set_dai_fmt, + .set_tdm_slot = rt5682s_set_tdm_slot, + .set_bclk_ratio = rt5682s_set_bclk1_ratio, +}; + +const struct snd_soc_dai_ops rt5682s_aif2_dai_ops = { + .hw_params = rt5682s_hw_params, + .set_fmt = rt5682s_set_dai_fmt, + .set_bclk_ratio = rt5682s_set_bclk2_ratio, +}; + +const struct snd_soc_component_driver rt5682s_soc_component_dev = { + .probe = rt5682s_probe, + .remove = rt5682s_remove, + .suspend = rt5682s_suspend, + .resume = rt5682s_resume, + .set_bias_level = rt5682s_set_bias_level, + .controls = rt5682s_snd_controls, + .num_controls = ARRAY_SIZE(rt5682s_snd_controls), + .dapm_widgets = rt5682s_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682s_dapm_widgets), + .dapm_routes = rt5682s_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682s_dapm_routes), + .set_sysclk = rt5682s_set_component_sysclk, + .set_pll = rt5682s_set_component_pll, + .set_jack = rt5682s_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682s->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682s->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682s->pdata.jd_src); + device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", + &rt5682s->pdata.dmic_clk_rate); + device_property_read_u32(dev, "realtek,dmic-delay-ms", + &rt5682s->pdata.dmic_delay); + + rt5682s->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + if (device_property_read_string_array(dev, "clock-output-names", + rt5682s->pdata.dai_clk_names, + RT5682S_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682s->pdata.dai_clk_names[RT5682S_DAI_WCLK_IDX], + rt5682s->pdata.dai_clk_names[RT5682S_DAI_BCLK_IDX]); + + rt5682s->pdata.dmic_clk_driving_high = device_property_read_bool(dev, + "realtek,dmic-clk-driving-high"); + + return 0; +} + +void rt5682s_calibrate(struct rt5682s_priv *rt5682s) +{ + unsigned int count, value; + + mutex_lock(&rt5682s->calibrate_mutex); + + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0xaa80); + usleep_range(15000, 20000); + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0xfa80); + regmap_write(rt5682s->regmap, RT5682S_PWR_DIG_1, 0x01c0); + regmap_write(rt5682s->regmap, RT5682S_MICBIAS_2, 0x0380); + regmap_write(rt5682s->regmap, RT5682S_GLB_CLK, 0x8000); + regmap_write(rt5682s->regmap, RT5682S_ADDA_CLK_1, 0x1001); + regmap_write(rt5682s->regmap, RT5682S_CHOP_DAC_2, 0x3030); + regmap_write(rt5682s->regmap, RT5682S_CHOP_ADC, 0xb000); + regmap_write(rt5682s->regmap, RT5682S_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682s->regmap, RT5682S_CAL_REC, 0x5151); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682s->regmap, RT5682S_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682s->regmap, RT5682S_HP_CALIB_ST_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + dev_err(rt5682s->component->dev, "HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682s->regmap, RT5682S_MICBIAS_2, 0x0180); + regmap_write(rt5682s->regmap, RT5682S_CAL_REC, 0x5858); + regmap_write(rt5682s->regmap, RT5682S_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682s->regmap, RT5682S_HP_CALIB_CTRL_2, 0x0320); + regmap_write(rt5682s->regmap, RT5682S_PWR_DIG_1, 0x00c0); + regmap_write(rt5682s->regmap, RT5682S_PWR_ANLG_1, 0x0800); + regmap_write(rt5682s->regmap, RT5682S_GLB_CLK, 0x0000); + + mutex_unlock(&rt5682s->calibrate_mutex); +} + +static const struct regmap_config rt5682s_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682S_MAX_REG, + .volatile_reg = rt5682s_volatile_register, + .readable_reg = rt5682s_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682s_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682s_reg), + .use_single_read = true, + .use_single_write = true, +}; + +static struct snd_soc_dai_driver rt5682s_dai[] = { + { + .name = "rt5682s-aif1", + .id = RT5682S_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .ops = &rt5682s_aif1_dai_ops, + }, + { + .name = "rt5682s-aif2", + .id = RT5682S_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682S_STEREO_RATES, + .formats = RT5682S_FORMATS, + }, + .ops = &rt5682s_aif2_dai_ops, + }, +}; + +static void rt5682s_i2c_disable_regulators(void *data) +{ + struct rt5682s_priv *rt5682s = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); +} + +static int rt5682s_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682s_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682s_priv *rt5682s; + int i, ret; + unsigned int val; + + rt5682s = devm_kzalloc(&i2c->dev, sizeof(struct rt5682s_priv), GFP_KERNEL); + if (!rt5682s) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682s); + + rt5682s->pdata = i2s_default_platform_data; + + if (pdata) + rt5682s->pdata = *pdata; + else + rt5682s_parse_dt(rt5682s, &i2c->dev); + + rt5682s->regmap = devm_regmap_init_i2c(i2c, &rt5682s_regmap); + if (IS_ERR(rt5682s->regmap)) { + ret = PTR_ERR(rt5682s->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682s->supplies); i++) + rt5682s->supplies[i].supply = rt5682s_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = devm_add_action_or_reset(&i2c->dev, rt5682s_i2c_disable_regulators, + rt5682s); + if (ret) + return ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682s->supplies), rt5682s->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682s->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682s->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682s")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 50 ms minimum */ + usleep_range(50000, 55000); + + regmap_read(rt5682s->regmap, RT5682S_DEVICE_ID, &val); + if (val != DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %x is not rt5682s\n", val); + return -ENODEV; + } + + rt5682s_reset(rt5682s); + rt5682s_apply_patch_list(rt5682s, &i2c->dev); + + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_DIG_2, + RT5682S_DLDO_I_LIMIT_MASK, RT5682S_DLDO_I_LIMIT_DIS); + usleep_range(20000, 25000); + + mutex_init(&rt5682s->calibrate_mutex); + mutex_init(&rt5682s->sar_mutex); + rt5682s_calibrate(rt5682s); + + regmap_update_bits(rt5682s->regmap, RT5682S_MICBIAS_2, + RT5682S_PWR_CLK25M_MASK | RT5682S_PWR_CLK1M_MASK, + RT5682S_PWR_CLK25M_PD | RT5682S_PWR_CLK1M_PU); + regmap_update_bits(rt5682s->regmap, RT5682S_PWR_ANLG_1, + RT5682S_PWR_BG, RT5682S_PWR_BG); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_LOGIC_CTRL_2, + RT5682S_HP_SIG_SRC_MASK, RT5682S_HP_SIG_SRC_1BIT_CTL); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_CHARGE_PUMP_2, + RT5682S_PM_HP_MASK, RT5682S_PM_HP_HV); + + /* DMIC pin*/ + if (rt5682s->pdata.dmic1_data_pin != RT5682S_DMIC1_NULL) { + switch (rt5682s->pdata.dmic1_data_pin) { + case RT5682S_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682s->regmap, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_DP_MASK, RT5682S_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP2_PIN_MASK, RT5682S_GP2_PIN_DMIC_SDA); + break; + + case RT5682S_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682s->regmap, RT5682S_DMIC_CTRL_1, + RT5682S_DMIC_1_DP_MASK, RT5682S_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP5_PIN_MASK, RT5682S_GP5_PIN_DMIC_SDA); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5682s->pdata.dmic1_clk_pin) { + case RT5682S_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP1_PIN_MASK, RT5682S_GP1_PIN_DMIC_CLK); + break; + + case RT5682S_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682s->regmap, RT5682S_GPIO_CTRL_1, + RT5682S_GP3_PIN_MASK, RT5682S_GP3_PIN_DMIC_CLK); + if (rt5682s->pdata.dmic_clk_driving_high) + regmap_update_bits(rt5682s->regmap, + RT5682S_PAD_DRIVING_CTRL, + RT5682S_PAD_DRV_GP3_MASK, + RT5682S_PAD_DRV_GP3_HIGH); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + INIT_DELAYED_WORK(&rt5682s->jack_detect_work, rt5682s_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682s->jd_check_work, rt5682s_jd_check_handler); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682s_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5682s", rt5682s); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5682s_soc_component_dev, + rt5682s_dai, ARRAY_SIZE(rt5682s_dai)); +} + +static void rt5682s_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682s_priv *rt5682s = i2c_get_clientdata(client); + + disable_irq(client->irq); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); + cancel_delayed_work_sync(&rt5682s->jd_check_work); + + rt5682s_reset(rt5682s); +} + +static int rt5682s_i2c_remove(struct i2c_client *client) +{ + rt5682s_i2c_shutdown(client); + + return 0; +} + +static const struct of_device_id rt5682s_of_match[] = { + {.compatible = "realtek,rt5682s"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682s_of_match); + +static const struct acpi_device_id rt5682s_acpi_match[] = { + {"RTL5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682s_acpi_match); + +static const struct i2c_device_id rt5682s_i2c_id[] = { + {"rt5682s", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682s_i2c_id); + +static struct i2c_driver rt5682s_i2c_driver = { + .driver = { + .name = "rt5682s", + .of_match_table = rt5682s_of_match, + .acpi_match_table = rt5682s_acpi_match, + .probe_type = PROBE_PREFER_ASYNCHRONOUS, + }, + .probe = rt5682s_i2c_probe, + .remove = rt5682s_i2c_remove, + .shutdown = rt5682s_i2c_shutdown, + .id_table = rt5682s_i2c_id, +}; +module_i2c_driver(rt5682s_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682I-VS driver"); +MODULE_AUTHOR("Derek Fang <derek.fang(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h new file mode 100644 index 000000000000..7c755e5efb81 --- /dev/null +++ b/sound/soc/codecs/rt5682s.h @@ -0,0 +1,1455 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5682s.h -- RT5682I-VS ALSA SoC audio driver + * + * Copyright 2021 Realtek Microelectronics + * Author: Derek Fang <derek.fang(a)realtek.com> + */ + +#ifndef __RT5682S_H__ +#define __RT5682S_H__ + +#include <sound/rt5682s.h> +#include <linux/regulator/consumer.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> + + +/* Info */ +#define RT5682S_RESET 0x0000 +#define RT5682S_VERSION_ID 0x00fd +#define RT5682S_VENDOR_ID 0x00fe +#define RT5682S_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682S_HP_CTRL_1 0x0002 +#define RT5682S_HP_CTRL_2 0x0003 +#define RT5682S_HPL_GAIN 0x0005 +#define RT5682S_HPR_GAIN 0x0006 + +#define RT5682S_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682S_CBJ_BST_CTRL 0x000b +#define RT5682S_CBJ_DET_CTRL 0x000f +#define RT5682S_CBJ_CTRL_1 0x0010 +#define RT5682S_CBJ_CTRL_2 0x0011 +#define RT5682S_CBJ_CTRL_3 0x0012 +#define RT5682S_CBJ_CTRL_4 0x0013 +#define RT5682S_CBJ_CTRL_5 0x0014 +#define RT5682S_CBJ_CTRL_6 0x0015 +#define RT5682S_CBJ_CTRL_7 0x0016 +#define RT5682S_CBJ_CTRL_8 0x0017 +/* I/O - ADC/DAC/DMIC */ +#define RT5682S_DAC1_DIG_VOL 0x0019 +#define RT5682S_STO1_ADC_DIG_VOL 0x001c +#define RT5682S_STO1_ADC_BOOST 0x001f +#define RT5682S_HP_IMP_GAIN_1 0x0022 +#define RT5682S_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682S_SIDETONE_CTRL 0x0024 +#define RT5682S_STO1_ADC_MIXER 0x0026 +#define RT5682S_AD_DA_MIXER 0x0029 +#define RT5682S_STO1_DAC_MIXER 0x002a +#define RT5682S_A_DAC1_MUX 0x002b +#define RT5682S_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682S_REC_MIXER 0x003c +#define RT5682S_CAL_REC 0x0044 +/* HP Analog Offset Control */ +#define RT5682S_HP_ANA_OST_CTRL_1 0x004b +#define RT5682S_HP_ANA_OST_CTRL_2 0x004c +#define RT5682S_HP_ANA_OST_CTRL_3 0x004d +/* Power */ +#define RT5682S_PWR_DIG_1 0x0061 +#define RT5682S_PWR_DIG_2 0x0062 +#define RT5682S_PWR_ANLG_1 0x0063 +#define RT5682S_PWR_ANLG_2 0x0064 +#define RT5682S_PWR_ANLG_3 0x0065 +#define RT5682S_PWR_MIXER 0x0066 + +#define RT5682S_MB_CTRL 0x0067 +#define RT5682S_CLK_GATE_TCON_1 0x0068 +#define RT5682S_CLK_GATE_TCON_2 0x0069 +#define RT5682S_CLK_GATE_TCON_3 0x006a +/* Clock Detect */ +#define RT5682S_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682S_RESET_LPF_CTRL 0x006c +#define RT5682S_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682S_DMIC_CTRL_1 0x006e +#define RT5682S_LPF_AD_DMIC 0x006f +/* Format - ADC/DAC */ +#define RT5682S_I2S1_SDP 0x0070 +#define RT5682S_I2S2_SDP 0x0071 +#define RT5682S_ADDA_CLK_1 0x0073 +#define RT5682S_ADDA_CLK_2 0x0074 +#define RT5682S_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682S_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682S_TDM_CTRL 0x0079 +#define RT5682S_TDM_ADDA_CTRL_1 0x007a +#define RT5682S_TDM_ADDA_CTRL_2 0x007b +#define RT5682S_DATA_SEL_CTRL_1 0x007c +#define RT5682S_TDM_TCON_CTRL_1 0x007e +#define RT5682S_TDM_TCON_CTRL_2 0x007f +/* Function - Analog */ +#define RT5682S_GLB_CLK 0x0080 +#define RT5682S_PLL_TRACK_1 0x0083 +#define RT5682S_PLL_TRACK_2 0x0084 +#define RT5682S_PLL_TRACK_3 0x0085 +#define RT5682S_PLL_TRACK_4 0x0086 +#define RT5682S_PLL_TRACK_5 0x0087 +#define RT5682S_PLL_TRACK_6 0x0088 +#define RT5682S_PLL_TRACK_11 0x008c +#define RT5682S_DEPOP_1 0x008e +#define RT5682S_HP_CHARGE_PUMP_1 0x008f +#define RT5682S_HP_CHARGE_PUMP_2 0x0091 +#define RT5682S_HP_CHARGE_PUMP_3 0x0092 +#define RT5682S_MICBIAS_1 0x0093 +#define RT5682S_MICBIAS_2 0x0094 +#define RT5682S_MICBIAS_3 0x0095 + +#define RT5682S_PLL_TRACK_12 0x0096 +#define RT5682S_PLL_TRACK_14 0x0097 +#define RT5682S_PLL_CTRL_1 0x0098 +#define RT5682S_PLL_CTRL_2 0x0099 +#define RT5682S_PLL_CTRL_3 0x009a +#define RT5682S_PLL_CTRL_4 0x009b +#define RT5682S_PLL_CTRL_5 0x009c +#define RT5682S_PLL_CTRL_6 0x009d +#define RT5682S_PLL_CTRL_7 0x009e + +#define RT5682S_RC_CLK_CTRL 0x009f +#define RT5682S_I2S2_M_CLK_CTRL_1 0x00a0 +#define RT5682S_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682S_I2S2_F_DIV_CTRL_2 0x00a4 + +#define RT5682S_IRQ_CTRL_1 0x00b6 +#define RT5682S_IRQ_CTRL_2 0x00b7 +#define RT5682S_IRQ_CTRL_3 0x00b8 +#define RT5682S_IRQ_CTRL_4 0x00b9 +#define RT5682S_INT_ST_1 0x00be +#define RT5682S_GPIO_CTRL_1 0x00c0 +#define RT5682S_GPIO_CTRL_2 0x00c1 +#define RT5682S_GPIO_ST 0x00c2 +#define RT5682S_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682S_MID_HP_AMP_DET 0x00d2 +#define RT5682S_LOW_HP_AMP_DET 0x00d3 +#define RT5682S_DELAY_BUF_CTRL 0x00d4 +#define RT5682S_SV_ZCD_1 0x00d9 +#define RT5682S_SV_ZCD_2 0x00da +#define RT5682S_IL_CMD_1 0x00db +#define RT5682S_IL_CMD_2 0x00dc +#define RT5682S_IL_CMD_3 0x00dd +#define RT5682S_IL_CMD_4 0x00de +#define RT5682S_IL_CMD_5 0x00df +#define RT5682S_IL_CMD_6 0x00e0 +#define RT5682S_4BTN_IL_CMD_1 0x00e2 +#define RT5682S_4BTN_IL_CMD_2 0x00e3 +#define RT5682S_4BTN_IL_CMD_3 0x00e4 +#define RT5682S_4BTN_IL_CMD_4 0x00e5 +#define RT5682S_4BTN_IL_CMD_5 0x00e6 +#define RT5682S_4BTN_IL_CMD_6 0x00e7 +#define RT5682S_4BTN_IL_CMD_7 0x00e8 + +#define RT5682S_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682S_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682S_AJD1_CTRL 0x00f0 +#define RT5682S_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682S_DUMMY_1 0x00fa +#define RT5682S_DUMMY_2 0x00fb +#define RT5682S_DUMMY_3 0x00fc + +#define RT5682S_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682S_BIAS_CUR_CTRL_2 0x010b +#define RT5682S_BIAS_CUR_CTRL_3 0x010c +#define RT5682S_BIAS_CUR_CTRL_4 0x010d +#define RT5682S_BIAS_CUR_CTRL_5 0x010e +#define RT5682S_BIAS_CUR_CTRL_6 0x010f +#define RT5682S_BIAS_CUR_CTRL_7 0x0110 +#define RT5682S_BIAS_CUR_CTRL_8 0x0111 +#define RT5682S_BIAS_CUR_CTRL_9 0x0112 +#define RT5682S_BIAS_CUR_CTRL_10 0x0113 +#define RT5682S_VREF_REC_OP_FB_CAP_CTRL_1 0x0117 +#define RT5682S_VREF_REC_OP_FB_CAP_CTRL_2 0x0118 +#define RT5682S_CHARGE_PUMP_1 0x0125 +#define RT5682S_DIG_IN_CTRL_1 0x0132 +#define RT5682S_PAD_DRIVING_CTRL 0x0136 +#define RT5682S_CHOP_DAC_1 0x0139 +#define RT5682S_CHOP_DAC_2 0x013a +#define RT5682S_CHOP_ADC 0x013b +#define RT5682S_CALIB_ADC_CTRL 0x013c +#define RT5682S_VOL_TEST 0x013f +#define RT5682S_SPKVDD_DET_ST 0x0142 +#define RT5682S_TEST_MODE_CTRL_1 0x0145 +#define RT5682S_TEST_MODE_CTRL_2 0x0146 +#define RT5682S_TEST_MODE_CTRL_3 0x0147 +#define RT5682S_TEST_MODE_CTRL_4 0x0148 +#define RT5682S_PLL_INTERNAL_1 0x0156 +#define RT5682S_PLL_INTERNAL_2 0x0157 +#define RT5682S_PLL_INTERNAL_3 0x0158 +#define RT5682S_PLL_INTERNAL_4 0x0159 +#define RT5682S_STO_NG2_CTRL_1 0x0160 +#define RT5682S_STO_NG2_CTRL_2 0x0161 +#define RT5682S_STO_NG2_CTRL_3 0x0162 +#define RT5682S_STO_NG2_CTRL_4 0x0163 +#define RT5682S_STO_NG2_CTRL_5 0x0164 +#define RT5682S_STO_NG2_CTRL_6 0x0165 +#define RT5682S_STO_NG2_CTRL_7 0x0166 +#define RT5682S_STO_NG2_CTRL_8 0x0167 +#define RT5682S_STO_NG2_CTRL_9 0x0168 +#define RT5682S_STO_NG2_CTRL_10 0x0169 +#define RT5682S_STO1_DAC_SIL_DET 0x0190 +#define RT5682S_SIL_PSV_CTRL1 0x0194 +#define RT5682S_SIL_PSV_CTRL2 0x0195 +#define RT5682S_SIL_PSV_CTRL3 0x0197 +#define RT5682S_SIL_PSV_CTRL4 0x0198 +#define RT5682S_SIL_PSV_CTRL5 0x0199 +#define RT5682S_HP_IMP_SENS_CTRL_1 0x01ac +#define RT5682S_HP_IMP_SENS_CTRL_2 0x01ad +#define RT5682S_HP_IMP_SENS_CTRL_3 0x01ae +#define RT5682S_HP_IMP_SENS_CTRL_4 0x01af +#define RT5682S_HP_IMP_SENS_CTRL_5 0x01b0 +#define RT5682S_HP_IMP_SENS_CTRL_6 0x01b1 +#define RT5682S_HP_IMP_SENS_CTRL_7 0x01b2 +#define RT5682S_HP_IMP_SENS_CTRL_8 0x01b3 +#define RT5682S_HP_IMP_SENS_CTRL_9 0x01b4 +#define RT5682S_HP_IMP_SENS_CTRL_10 0x01b5 +#define RT5682S_HP_IMP_SENS_CTRL_11 0x01b6 +#define RT5682S_HP_IMP_SENS_CTRL_12 0x01b7 +#define RT5682S_HP_IMP_SENS_CTRL_13 0x01b8 +#define RT5682S_HP_IMP_SENS_CTRL_14 0x01b9 +#define RT5682S_HP_IMP_SENS_CTRL_15 0x01ba +#define RT5682S_HP_IMP_SENS_CTRL_16 0x01bb +#define RT5682S_HP_IMP_SENS_CTRL_17 0x01bc +#define RT5682S_HP_IMP_SENS_CTRL_18 0x01bd +#define RT5682S_HP_IMP_SENS_CTRL_19 0x01be +#define RT5682S_HP_IMP_SENS_CTRL_20 0x01bf +#define RT5682S_HP_IMP_SENS_CTRL_21 0x01c0 +#define RT5682S_HP_IMP_SENS_CTRL_22 0x01c1 +#define RT5682S_HP_IMP_SENS_CTRL_23 0x01c2 +#define RT5682S_HP_IMP_SENS_CTRL_24 0x01c3 +#define RT5682S_HP_IMP_SENS_CTRL_25 0x01c4 +#define RT5682S_HP_IMP_SENS_CTRL_26 0x01c5 +#define RT5682S_HP_IMP_SENS_CTRL_27 0x01c6 +#define RT5682S_HP_IMP_SENS_CTRL_28 0x01c7 +#define RT5682S_HP_IMP_SENS_CTRL_29 0x01c8 +#define RT5682S_HP_IMP_SENS_CTRL_30 0x01c9 +#define RT5682S_HP_IMP_SENS_CTRL_31 0x01ca +#define RT5682S_HP_IMP_SENS_CTRL_32 0x01cb +#define RT5682S_HP_IMP_SENS_CTRL_33 0x01cc +#define RT5682S_HP_IMP_SENS_CTRL_34 0x01cd +#define RT5682S_HP_IMP_SENS_CTRL_35 0x01ce +#define RT5682S_HP_IMP_SENS_CTRL_36 0x01cf +#define RT5682S_HP_IMP_SENS_CTRL_37 0x01d0 +#define RT5682S_HP_IMP_SENS_CTRL_38 0x01d1 +#define RT5682S_HP_IMP_SENS_CTRL_39 0x01d2 +#define RT5682S_HP_IMP_SENS_CTRL_40 0x01d3 +#define RT5682S_HP_IMP_SENS_CTRL_41 0x01d4 +#define RT5682S_HP_IMP_SENS_CTRL_42 0x01d5 +#define RT5682S_HP_IMP_SENS_CTRL_43 0x01d6 +#define RT5682S_HP_IMP_SENS_CTRL_44 0x01d7 +#define RT5682S_HP_IMP_SENS_CTRL_45 0x01d8 +#define RT5682S_HP_IMP_SENS_CTRL_46 0x01d9 +#define RT5682S_HP_LOGIC_CTRL_1 0x01da +#define RT5682S_HP_LOGIC_CTRL_2 0x01db +#define RT5682S_HP_LOGIC_CTRL_3 0x01dc +#define RT5682S_HP_CALIB_CTRL_1 0x01de +#define RT5682S_HP_CALIB_CTRL_2 0x01df +#define RT5682S_HP_CALIB_CTRL_3 0x01e0 +#define RT5682S_HP_CALIB_CTRL_4 0x01e1 +#define RT5682S_HP_CALIB_CTRL_5 0x01e2 +#define RT5682S_HP_CALIB_CTRL_6 0x01e3 +#define RT5682S_HP_CALIB_CTRL_7 0x01e4 +#define RT5682S_HP_CALIB_CTRL_8 0x01e5 +#define RT5682S_HP_CALIB_CTRL_9 0x01e6 +#define RT5682S_HP_CALIB_CTRL_10 0x01e7 +#define RT5682S_HP_CALIB_CTRL_11 0x01e8 +#define RT5682S_HP_CALIB_ST_1 0x01ea +#define RT5682S_HP_CALIB_ST_2 0x01eb +#define RT5682S_HP_CALIB_ST_3 0x01ec +#define RT5682S_HP_CALIB_ST_4 0x01ed +#define RT5682S_HP_CALIB_ST_5 0x01ee +#define RT5682S_HP_CALIB_ST_6 0x01ef +#define RT5682S_HP_CALIB_ST_7 0x01f0 +#define RT5682S_HP_CALIB_ST_8 0x01f1 +#define RT5682S_HP_CALIB_ST_9 0x01f2 +#define RT5682S_HP_CALIB_ST_10 0x01f3 +#define RT5682S_HP_CALIB_ST_11 0x01f4 +#define RT5682S_SAR_IL_CMD_1 0x0210 +#define RT5682S_SAR_IL_CMD_2 0x0211 +#define RT5682S_SAR_IL_CMD_3 0x0212 +#define RT5682S_SAR_IL_CMD_4 0x0213 +#define RT5682S_SAR_IL_CMD_5 0x0214 +#define RT5682S_SAR_IL_CMD_6 0x0215 +#define RT5682S_SAR_IL_CMD_7 0x0216 +#define RT5682S_SAR_IL_CMD_8 0x0217 +#define RT5682S_SAR_IL_CMD_9 0x0218 +#define RT5682S_SAR_IL_CMD_10 0x0219 +#define RT5682S_SAR_IL_CMD_11 0x021a +#define RT5682S_SAR_IL_CMD_12 0x021b +#define RT5682S_SAR_IL_CMD_13 0x021c +#define RT5682S_SAR_IL_CMD_14 0x021d +#define RT5682S_DUMMY_4 0x02fa +#define RT5682S_DUMMY_5 0x02fb +#define RT5682S_DUMMY_6 0x02fc +#define RT5682S_VERSION_ID_HIDE 0x03fe +#define RT5682S_VERSION_ID_CUS 0x03ff +#define RT5682S_SCAN_CTL 0x0500 +#define RT5682S_HP_AMP_DET 0x0600 +#define RT5682S_BIAS_CUR_CTRL_11 0x0610 +#define RT5682S_BIAS_CUR_CTRL_12 0x0611 +#define RT5682S_BIAS_CUR_CTRL_13 0x0620 +#define RT5682S_BIAS_CUR_CTRL_14 0x0621 +#define RT5682S_BIAS_CUR_CTRL_15 0x0630 +#define RT5682S_BIAS_CUR_CTRL_16 0x0631 +#define RT5682S_BIAS_CUR_CTRL_17 0x0640 +#define RT5682S_BIAS_CUR_CTRL_18 0x0641 +#define RT5682S_I2C_TRANS_CTRL 0x07fa +#define RT5682S_DUMMY_7 0x08fa +#define RT5682S_DUMMY_8 0x08fb +#define RT5682S_DMIC_FLOAT_DET 0x0d00 +#define RT5682S_HA_CMP_OP_1 0x1100 +#define RT5682S_HA_CMP_OP_2 0x1101 +#define RT5682S_HA_CMP_OP_3 0x1102 +#define RT5682S_HA_CMP_OP_4 0x1103 +#define RT5682S_HA_CMP_OP_5 0x1104 +#define RT5682S_HA_CMP_OP_6 0x1105 +#define RT5682S_HA_CMP_OP_7 0x1106 +#define RT5682S_HA_CMP_OP_8 0x1107 +#define RT5682S_HA_CMP_OP_9 0x1108 +#define RT5682S_HA_CMP_OP_10 0x1109 +#define RT5682S_HA_CMP_OP_11 0x110a +#define RT5682S_HA_CMP_OP_12 0x110b +#define RT5682S_HA_CMP_OP_13 0x110c +#define RT5682S_HA_CMP_OP_14 0x1111 +#define RT5682S_HA_CMP_OP_15 0x1112 +#define RT5682S_HA_CMP_OP_16 0x1113 +#define RT5682S_HA_CMP_OP_17 0x1114 +#define RT5682S_HA_CMP_OP_18 0x1115 +#define RT5682S_HA_CMP_OP_19 0x1116 +#define RT5682S_HA_CMP_OP_20 0x1117 +#define RT5682S_HA_CMP_OP_21 0x1118 +#define RT5682S_HA_CMP_OP_22 0x1119 +#define RT5682S_HA_CMP_OP_23 0x111a +#define RT5682S_HA_CMP_OP_24 0x111b +#define RT5682S_HA_CMP_OP_25 0x111c +#define RT5682S_NEW_CBJ_DET_CTL_1 0x1401 +#define RT5682S_NEW_CBJ_DET_CTL_2 0x1402 +#define RT5682S_NEW_CBJ_DET_CTL_3 0x1403 +#define RT5682S_NEW_CBJ_DET_CTL_4 0x1404 +#define RT5682S_NEW_CBJ_DET_CTL_5 0x1406 +#define RT5682S_NEW_CBJ_DET_CTL_6 0x1407 +#define RT5682S_NEW_CBJ_DET_CTL_7 0x1408 +#define RT5682S_NEW_CBJ_DET_CTL_8 0x1409 +#define RT5682S_NEW_CBJ_DET_CTL_9 0x140a +#define RT5682S_NEW_CBJ_DET_CTL_10 0x140b +#define RT5682S_NEW_CBJ_DET_CTL_11 0x140c +#define RT5682S_NEW_CBJ_DET_CTL_12 0x140d +#define RT5682S_NEW_CBJ_DET_CTL_13 0x140e +#define RT5682S_NEW_CBJ_DET_CTL_14 0x140f +#define RT5682S_NEW_CBJ_DET_CTL_15 0x1410 +#define RT5682S_NEW_CBJ_DET_CTL_16 0x1411 +#define RT5682S_DA_FILTER_1 0x1801 +#define RT5682S_DA_FILTER_2 0x1802 +#define RT5682S_DA_FILTER_3 0x1803 +#define RT5682S_DA_FILTER_4 0x1804 +#define RT5682S_DA_FILTER_5 0x1805 +#define RT5682S_CLK_SW_TEST_1 0x2c00 +#define RT5682S_CLK_SW_TEST_2 0x3400 +#define RT5682S_CLK_SW_TEST_3 0x3404 +#define RT5682S_CLK_SW_TEST_4 0x3405 +#define RT5682S_CLK_SW_TEST_5 0x3406 +#define RT5682S_CLK_SW_TEST_6 0x3407 +#define RT5682S_CLK_SW_TEST_7 0x3408 +#define RT5682S_CLK_SW_TEST_8 0x3409 +#define RT5682S_CLK_SW_TEST_9 0x340a +#define RT5682S_CLK_SW_TEST_10 0x340b +#define RT5682S_CLK_SW_TEST_11 0x340c +#define RT5682S_CLK_SW_TEST_12 0x340d +#define RT5682S_CLK_SW_TEST_13 0x340e +#define RT5682S_CLK_SW_TEST_14 0x340f +#define RT5682S_EFUSE_MANU_WRITE_1 0x3410 +#define RT5682S_EFUSE_MANU_WRITE_2 0x3411 +#define RT5682S_EFUSE_MANU_WRITE_3 0x3412 +#define RT5682S_EFUSE_MANU_WRITE_4 0x3413 +#define RT5682S_EFUSE_MANU_WRITE_5 0x3414 +#define RT5682S_EFUSE_MANU_WRITE_6 0x3415 +#define RT5682S_EFUSE_READ_1 0x3424 +#define RT5682S_EFUSE_READ_2 0x3425 +#define RT5682S_EFUSE_READ_3 0x3426 +#define RT5682S_EFUSE_READ_4 0x3427 +#define RT5682S_EFUSE_READ_5 0x3428 +#define RT5682S_EFUSE_READ_6 0x3429 +#define RT5682S_EFUSE_READ_7 0x342a +#define RT5682S_EFUSE_READ_8 0x342b +#define RT5682S_EFUSE_READ_9 0x342c +#define RT5682S_EFUSE_READ_10 0x342d +#define RT5682S_EFUSE_READ_11 0x342e +#define RT5682S_EFUSE_READ_12 0x342f +#define RT5682S_EFUSE_READ_13 0x3430 +#define RT5682S_EFUSE_READ_14 0x3431 +#define RT5682S_EFUSE_READ_15 0x3432 +#define RT5682S_EFUSE_READ_16 0x3433 +#define RT5682S_EFUSE_READ_17 0x3434 +#define RT5682S_EFUSE_READ_18 0x3435 +#define RT5682S_EFUSE_TIMING_CTL_1 0x3440 +#define RT5682S_EFUSE_TIMING_CTL_2 0x3441 +#define RT5682S_PILOT_DIG_CTL_1 0x3500 +#define RT5682S_PILOT_DIG_CTL_2 0x3501 +#define RT5682S_HP_AMP_DET_CTL_1 0x3b00 +#define RT5682S_HP_AMP_DET_CTL_2 0x3b01 +#define RT5682S_HP_AMP_DET_CTL_3 0x3b02 +#define RT5682S_HP_AMP_DET_CTL_4 0x3b03 + +#define RT5682S_MAX_REG (RT5682S_HP_AMP_DET_CTL_4) + +/* global definition */ +#define RT5682S_L_MUTE (0x1 << 15) +#define RT5682S_L_MUTE_SFT 15 +#define RT5682S_R_MUTE (0x1 << 7) +#define RT5682S_R_MUTE_SFT 7 +#define RT5682S_L_VOL_SFT 8 +#define RT5682S_R_VOL_SFT 0 +#define RT5682S_CLK_SRC_MCLK (0x0) +#define RT5682S_CLK_SRC_PLL1 (0x1) +#define RT5682S_CLK_SRC_PLL2 (0x2) +#define RT5682S_CLK_SRC_RCCLK (0x4) /* 25M */ + + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5682S_HPO_L_PATH_MASK (0x1 << 14) +#define RT5682S_HPO_L_PATH_EN (0x1 << 14) +#define RT5682S_HPO_L_PATH_DIS (0x0 << 14) +#define RT5682S_HPO_R_PATH_MASK (0x1 << 13) +#define RT5682S_HPO_R_PATH_EN (0x1 << 13) +#define RT5682S_HPO_R_PATH_DIS (0x0 << 13) +#define RT5682S_HPO_SEL_IP_EN_SW (0x1) +#define RT5682S_HPO_IP_EN_GATING (0x1) +#define RT5682S_HPO_IP_NO_GATING (0x0) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682S_G_HP (0xf << 8) +#define RT5682S_G_HP_SFT 8 +#define RT5682S_G_STO_DA_DMIX (0xf) +#define RT5682S_G_STO_DA_SFT 0 + +/* Embeeded Jack and Type Detection Control 2 (0x0010) */ +#define RT5682S_EMB_JD_MASK (0x1 << 15) +#define RT5682S_EMB_JD_EN (0x1 << 15) +#define RT5682S_EMB_JD_EN_SFT 15 +#define RT5682S_EMB_JD_RST (0x1 << 14) +#define RT5682S_JD_MODE (0x1 << 13) +#define RT5682S_JD_MODE_SFT 13 +#define RT5682S_DET_TYPE (0x1 << 12) +#define RT5682S_DET_TYPE_SFT 12 +#define RT5682S_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682S_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682S_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682S_SEL_FAST_OFF_MASK (0x3 << 9) +#define RT5682S_SEL_FAST_OFF_SFT 9 +#define RT5682S_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682S_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682S_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682S_FAST_OFF_MASK (0x1 << 7) +#define RT5682S_FAST_OFF_EN (0x1 << 7) +#define RT5682S_FAST_OFF_DIS (0x0 << 7) +#define RT5682S_VREF_POW_MASK (0x1 << 6) +#define RT5682S_VREF_POW_FSM (0x0 << 6) +#define RT5682S_VREF_POW_REG (0x1 << 6) +#define RT5682S_MB1_PATH_BIT 5 +#define RT5682S_MB1_PATH_MASK (0x1 << 5) +#define RT5682S_CTRL_MB1_REG (0x1 << 5) +#define RT5682S_CTRL_MB1_FSM (0x0 << 5) +#define RT5682S_MB2_PATH_BIT 4 +#define RT5682S_MB2_PATH_MASK (0x1 << 4) +#define RT5682S_CTRL_MB2_REG (0x1 << 4) +#define RT5682S_CTRL_MB2_FSM (0x0 << 4) +#define RT5682S_TRIG_JD_MASK (0x1 << 3) +#define RT5682S_TRIG_JD_HIGH (0x1 << 3) +#define RT5682S_TRIG_JD_LOW (0x0 << 3) +#define RT5682S_MIC_CAP_MASK (0x1 << 1) +#define RT5682S_MIC_CAP_HS (0x1 << 1) +#define RT5682S_MIC_CAP_HP (0x0 << 1) +#define RT5682S_MIC_CAP_SRC_MASK (0x1) +#define RT5682S_MIC_CAP_SRC_REG (0x1) +#define RT5682S_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 3 (0x0011) */ +#define RT5682S_SEL_CBJ_TYPE_SLOW (0x1 << 15) +#define RT5682S_SEL_CBJ_TYPE_NORM (0x0 << 15) +#define RT5682S_SEL_CBJ_TYPE_MASK (0x1 << 15) +#define RT5682S_POW_BG_MB1_MASK (0x1 << 13) +#define RT5682S_POW_BG_MB1_REG (0x1 << 13) +#define RT5682S_POW_BG_MB1_FSM (0x0 << 13) +#define RT5682S_POW_BG_MB2_MASK (0x1 << 12) +#define RT5682S_POW_BG_MB2_REG (0x1 << 12) +#define RT5682S_POW_BG_MB2_FSM (0x0 << 12) +#define RT5682S_EXT_JD_SRC (0x7 << 4) +#define RT5682S_EXT_JD_SRC_SFT 4 +#define RT5682S_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682S_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682S_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682S_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682S_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682S_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 4 (0x0012) */ +#define RT5682S_CBJ_IN_BUF_MASK (0x1 << 7) +#define RT5682S_CBJ_IN_BUF_EN (0x1 << 7) +#define RT5682S_CBJ_IN_BUF_DIS (0x0 << 7) +#define RT5682S_CBJ_IN_BUF_BIT 7 + +/* Combo Jack and Type Detection Control 5 (0x0013) */ +#define RT5682S_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682S_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682S_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682S_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682S_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682S_CBJ_JD_TEST_MODE (0x1 << 6) + +/* Combo Jack and Type Detection Control 6 (0x0014) */ +#define RT5682S_JD_FAST_OFF_SRC_MASK (0x7 << 8) +#define RT5682S_JD_FAST_OFF_SRC_JDH (0x6 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO6 (0x5 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO5 (0x4 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO4 (0x3 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO3 (0x2 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO2 (0x1 << 8) +#define RT5682S_JD_FAST_OFF_SRC_GPIO1 (0x0 << 8) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682S_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682S_DAC_L1_VOL_SFT 8 +#define RT5682S_DAC_R1_VOL_MASK (0xff) +#define RT5682S_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682S_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682S_ADC_L_VOL_SFT 8 +#define RT5682S_ADC_R_VOL_MASK (0x7f) +#define RT5682S_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682S_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682S_STO1_ADC_L_BST_SFT 14 +#define RT5682S_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682S_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682S_ST_SRC_SEL (0x1 << 8) +#define RT5682S_ST_SRC_SFT 8 +#define RT5682S_ST_EN_MASK (0x1 << 6) +#define RT5682S_ST_DIS (0x0 << 6) +#define RT5682S_ST_EN (0x1 << 6) +#define RT5682S_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682S_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682S_M_STO1_ADC_L1_SFT 15 +#define RT5682S_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682S_M_STO1_ADC_L2_SFT 14 +#define RT5682S_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682S_STO1_ADC1L_SRC_SFT 13 +#define RT5682S_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682S_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682S_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682S_STO1_ADC2L_SRC_SFT 12 +#define RT5682S_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682S_STO1_ADCL_SRC_SFT 10 +#define RT5682S_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682S_M_STO1_ADC_R1_SFT 7 +#define RT5682S_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682S_M_STO1_ADC_R2_SFT 6 +#define RT5682S_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682S_STO1_ADC1R_SRC_SFT 5 +#define RT5682S_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682S_STO1_ADC2R_SRC_SFT 4 +#define RT5682S_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682S_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682S_M_ADCMIX_L (0x1 << 15) +#define RT5682S_M_ADCMIX_L_SFT 15 +#define RT5682S_M_DAC1_L (0x1 << 14) +#define RT5682S_M_DAC1_L_SFT 14 +#define RT5682S_M_ADCMIX_R (0x1 << 7) +#define RT5682S_M_ADCMIX_R_SFT 7 +#define RT5682S_M_DAC1_R (0x1 << 6) +#define RT5682S_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682S_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682S_M_DAC_L1_STO_L_SFT 15 +#define RT5682S_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682S_G_DAC_L1_STO_L_SFT 14 +#define RT5682S_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682S_M_DAC_R1_STO_L_SFT 13 +#define RT5682S_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682S_G_DAC_R1_STO_L_SFT 12 +#define RT5682S_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682S_M_DAC_L1_STO_R_SFT 7 +#define RT5682S_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682S_G_DAC_L1_STO_R_SFT 6 +#define RT5682S_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682S_M_DAC_R1_STO_R_SFT 5 +#define RT5682S_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682S_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682S_M_ST_STO_L (0x1 << 9) +#define RT5682S_M_ST_STO_L_SFT 9 +#define RT5682S_M_ST_STO_R (0x1 << 8) +#define RT5682S_M_ST_STO_R_SFT 8 +#define RT5682S_DAC_L1_SRC_MASK (0x1 << 4) +#define RT5682S_A_DACL1_SFT 4 +#define RT5682S_DAC_R1_SRC_MASK (0x1) +#define RT5682S_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682S_IF2_DAC_SEL_MASK (0x3 << 2) +#define RT5682S_IF2_DAC_SEL_SFT 2 +#define RT5682S_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682S_IF2_ADC_SEL_SFT 0 + +/* REC Left/Right Mixer Control 2 (0x003c) */ +#define RT5682S_BST_CBJ_MASK (0x3f << 8) +#define RT5682S_BST_CBJ_SFT 8 +#define RT5682S_M_CBJ_RM1_L (0x1 << 7) +#define RT5682S_M_CBJ_RM1_L_SFT 7 +#define RT5682S_M_CBJ_RM1_R (0x1 << 6) +#define RT5682S_M_CBJ_RM1_R_SFT 6 + +/* REC Left/Right Mixer Calibration Control(0x0044) */ +#define RT5682S_PWR_RM1_R_BIT 8 +#define RT5682S_PWR_RM1_L_BIT 0 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682S_PWR_I2S1 (0x1 << 15) +#define RT5682S_PWR_I2S1_BIT 15 +#define RT5682S_PWR_I2S2 (0x1 << 14) +#define RT5682S_PWR_I2S2_BIT 14 +#define RT5682S_PRE_CHR_DAC_L1 (0x1 << 13) +#define RT5682S_PRE_CHR_DAC_L1_BIT 13 +#define RT5682S_PRE_CHR_DAC_R1 (0x1 << 12) +#define RT5682S_PRE_CHR_DAC_R1_BIT 12 +#define RT5682S_PWR_DAC_L1 (0x1 << 11) +#define RT5682S_PWR_DAC_L1_BIT 11 +#define RT5682S_PWR_DAC_R1 (0x1 << 10) +#define RT5682S_PWR_DAC_R1_BIT 10 +#define RT5682S_PWR_LDO (0x1 << 8) +#define RT5682S_PWR_LDO_BIT 8 +#define RT5682S_PWR_D2S_L (0x1 << 7) +#define RT5682S_PWR_D2S_L_BIT 7 +#define RT5682S_PWR_D2S_R (0x1 << 6) +#define RT5682S_PWR_D2S_R_BIT 6 +#define RT5682S_PWR_ADC_L1 (0x1 << 4) +#define RT5682S_PWR_ADC_L1_BIT 4 +#define RT5682S_PWR_ADC_R1 (0x1 << 3) +#define RT5682S_PWR_ADC_R1_BIT 3 +#define RT5682S_EFUSE_SW_EN (0x1 << 2) +#define RT5682S_EFUSE_SW_DIS (0x0 << 2) +#define RT5682S_PWR_EFUSE (0x1 << 1) +#define RT5682S_PWR_EFUSE_BIT 1 +#define RT5682S_DIG_GATE_CTRL (0x1 << 0) +#define RT5682S_DIG_GATE_CTRL_SFT 0 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682S_PWR_ADC_S1F (0x1 << 15) +#define RT5682S_PWR_ADC_S1F_BIT 15 +#define RT5682S_PWR_DAC_S1F (0x1 << 10) +#define RT5682S_PWR_DAC_S1F_BIT 10 +#define RT5682S_DLDO_I_LIMIT_MASK (0x1 << 7) +#define RT5682S_DLDO_I_LIMIT_EN (0x1 << 7) +#define RT5682S_DLDO_I_LIMIT_DIS (0x0 << 7) +#define RT5682S_DLDO_I_BIAS_SEL_4 (0x1 << 6) +#define RT5682S_DLDO_I_BIAS_SEL_0 (0x0 << 6) +#define RT5682S_DLDO_REG_TEST_1 (0x1 << 5) +#define RT5682S_DLDO_REG_TEST_0 (0x0 << 5) +#define RT5682S_DLDO_SRC_REG (0x1 << 4) +#define RT5682S_DLDO_SRC_EFUSE (0x0 << 4) + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682S_PWR_VREF1 (0x1 << 15) +#define RT5682S_PWR_VREF1_BIT 15 +#define RT5682S_PWR_FV1 (0x1 << 14) +#define RT5682S_PWR_FV1_BIT 14 +#define RT5682S_PWR_VREF2 (0x1 << 13) +#define RT5682S_PWR_VREF2_BIT 13 +#define RT5682S_PWR_FV2 (0x1 << 12) +#define RT5682S_PWR_FV2_BIT 12 +#define RT5682S_LDO1_DBG_MASK (0x3 << 10) +#define RT5682S_PWR_MB (0x1 << 9) +#define RT5682S_PWR_MB_BIT 9 +#define RT5682S_PWR_BG (0x1 << 7) +#define RT5682S_PWR_BG_BIT 7 +#define RT5682S_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682S_LDO1_BYPASS (0x1 << 6) +#define RT5682S_LDO1_NOT_BYPASS (0x0 << 6) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682S_PWR_MCLK0_WD (0x1 << 15) +#define RT5682S_PWR_MCLK0_WD_BIT 15 +#define RT5682S_PWR_MCLK1_WD (0x1 << 14) +#define RT5682S_PWR_MCLK1_WD_BIT 14 +#define RT5682S_RST_MCLK0 (0x1 << 13) +#define RT5682S_RST_MCLK0_BIT 13 +#define RT5682S_RST_MCLK1 (0x1 << 12) +#define RT5682S_RST_MCLK1_BIT 12 +#define RT5682S_PWR_MB1 (0x1 << 11) +#define RT5682S_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682S_PWR_MB1_BIT 11 +#define RT5682S_PWR_MB2 (0x1 << 10) +#define RT5682S_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682S_PWR_MB2_BIT 10 +#define RT5682S_PWR_JD_MASK (0x1 << 0) +#define RT5682S_PWR_JD_ENABLE (0x1 << 0) +#define RT5682S_PWR_JD_DISABLE (0x0 << 0) + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682S_PWR_LDO_PLLA (0x1 << 15) +#define RT5682S_PWR_LDO_PLLA_BIT 15 +#define RT5682S_PWR_LDO_PLLB (0x1 << 14) +#define RT5682S_PWR_LDO_PLLB_BIT 14 +#define RT5682S_PWR_BIAS_PLLA (0x1 << 13) +#define RT5682S_PWR_BIAS_PLLA_BIT 13 +#define RT5682S_PWR_BIAS_PLLB (0x1 << 12) +#define RT5682S_PWR_BIAS_PLLB_BIT 12 +#define RT5682S_PWR_CBJ (0x1 << 9) +#define RT5682S_PWR_CBJ_BIT 9 +#define RT5682S_RSTB_PLLB (0x1 << 7) +#define RT5682S_RSTB_PLLB_BIT 7 +#define RT5682S_RSTB_PLLA (0x1 << 6) +#define RT5682S_RSTB_PLLA_BIT 6 +#define RT5682S_PWR_PLLB (0x1 << 5) +#define RT5682S_PWR_PLLB_BIT 5 +#define RT5682S_PWR_PLLA (0x1 << 4) +#define RT5682S_PWR_PLLA_BIT 4 +#define RT5682S_PWR_LDO_MB2 (0x1 << 2) +#define RT5682S_PWR_LDO_MB2_BIT 2 +#define RT5682S_PWR_LDO_MB1 (0x1 << 1) +#define RT5682S_PWR_LDO_MB1_BIT 1 +#define RT5682S_PWR_BGLDO (0x1 << 0) +#define RT5682S_PWR_BGLDO_BIT 0 + +/* Power Management for Mixer (0x0066) */ +#define RT5682S_PWR_CLK_COMP_8FS (0x1 << 15) +#define RT5682S_PWR_CLK_COMP_8FS_BIT 15 +#define RT5682S_DBG_BGLDO_MASK (0x3 << 12) +#define RT5682S_DBG_BGLDO_SFT 12 +#define RT5682S_DBG_BGLDO_MB1_MASK (0x3 << 10) +#define RT5682S_DBG_BGLDO_MB1_SFT 10 +#define RT5682S_DBG_BGLDO_MB2_MASK (0x3 << 8) +#define RT5682S_DBG_BGLDO_MB2_SFT 8 +#define RT5682S_DLDO_BGLDO_MASK (0x3 << 6) +#define RT5682S_DLDO_BGLDO_MB2_SFT 6 +#define RT5682S_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682S_PWR_STO1_DAC_L_BIT 5 +#define RT5682S_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682S_PWR_STO1_DAC_R_BIT 4 +#define RT5682S_DVO_BGLDO_MB1_MASK (0x3 << 2) +#define RT5682S_DVO_BGLDO_MB1_SFT 2 +#define RT5682S_DVO_BGLDO_MB2_MASK (0x3 << 0) + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682S_SYS_CLK_DET (0x1 << 15) +#define RT5682S_SYS_CLK_DET_SFT 15 +#define RT5682S_PLL1_CLK_DET (0x1 << 14) +#define RT5682S_PLL1_CLK_DET_SFT 14 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682S_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682S_DMIC_1_EN_SFT 15 +#define RT5682S_DMIC_1_DIS (0x0 << 15) +#define RT5682S_DMIC_1_EN (0x1 << 15) +#define RT5682S_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682S_FIFO_CLK_DIV_2 (0x1 << 12) +#define RT5682S_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682S_DMIC_1_DP_SFT 4 +#define RT5682S_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682S_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682S_DMIC_CLK_MASK (0xf << 0) +#define RT5682S_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682S_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682S_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682S_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682S_SEL_ADCDAT_SFT 15 +#define RT5682S_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682S_I2S1_TX_CHL_SFT 12 +#define RT5682S_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682S_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682S_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682S_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682S_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682S_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682S_I2S1_RX_CHL_SFT 8 +#define RT5682S_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682S_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682S_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682S_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682S_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682S_I2S1_MONO_MASK (0x1 << 7) +#define RT5682S_I2S1_MONO_EN (0x1 << 7) +#define RT5682S_I2S1_MONO_DIS (0x0 << 7) +#define RT5682S_I2S1_DL_MASK (0x7 << 4) +#define RT5682S_I2S1_DL_SFT 4 +#define RT5682S_I2S1_DL_16 (0x0 << 4) +#define RT5682S_I2S1_DL_20 (0x1 << 4) +#define RT5682S_I2S1_DL_24 (0x2 << 4) +#define RT5682S_I2S1_DL_32 (0x3 << 4) +#define RT5682S_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0071) */ +#define RT5682S_I2S2_MS_MASK (0x1 << 15) +#define RT5682S_I2S2_MS_SFT 15 +#define RT5682S_I2S2_MS_M (0x0 << 15) +#define RT5682S_I2S2_MS_S (0x1 << 15) +#define RT5682S_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682S_I2S2_PIN_CFG_SFT 14 +#define RT5682S_I2S2_OUT_MASK (0x1 << 9) +#define RT5682S_I2S2_OUT_SFT 9 +#define RT5682S_I2S2_OUT_UM (0x0 << 9) +#define RT5682S_I2S2_OUT_M (0x1 << 9) +#define RT5682S_I2S_BP_MASK (0x1 << 8) +#define RT5682S_I2S_BP_SFT 8 +#define RT5682S_I2S_BP_NOR (0x0 << 8) +#define RT5682S_I2S_BP_INV (0x1 << 8) +#define RT5682S_I2S2_MONO_MASK (0x1 << 7) +#define RT5682S_I2S2_MONO_EN (0x1 << 7) +#define RT5682S_I2S2_MONO_DIS (0x0 << 7) +#define RT5682S_I2S2_DL_MASK (0x7 << 4) +#define RT5682S_I2S2_DL_SFT 4 +#define RT5682S_I2S2_DL_8 (0x0 << 4) +#define RT5682S_I2S2_DL_16 (0x1 << 4) +#define RT5682S_I2S2_DL_20 (0x2 << 4) +#define RT5682S_I2S2_DL_24 (0x3 << 4) +#define RT5682S_I2S2_DL_32 (0x4 << 4) +#define RT5682S_I2S_DF_MASK (0x7) +#define RT5682S_I2S_DF_SFT 0 +#define RT5682S_I2S_DF_I2S (0x0) +#define RT5682S_I2S_DF_LEFT (0x1) +#define RT5682S_I2S_DF_PCM_A (0x2) +#define RT5682S_I2S_DF_PCM_B (0x3) +#define RT5682S_I2S_DF_PCM_A_N (0x6) +#define RT5682S_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682S_ADC_OSR_MASK (0xf << 12) +#define RT5682S_ADC_OSR_SFT 12 +#define RT5682S_ADC_OSR_D_1 (0x0 << 12) +#define RT5682S_ADC_OSR_D_2 (0x1 << 12) +#define RT5682S_ADC_OSR_D_4 (0x2 << 12) +#define RT5682S_ADC_OSR_D_6 (0x3 << 12) +#define RT5682S_ADC_OSR_D_8 (0x4 << 12) +#define RT5682S_ADC_OSR_D_12 (0x5 << 12) +#define RT5682S_ADC_OSR_D_16 (0x6 << 12) +#define RT5682S_ADC_OSR_D_24 (0x7 << 12) +#define RT5682S_ADC_OSR_D_32 (0x8 << 12) +#define RT5682S_ADC_OSR_D_48 (0x9 << 12) +#define RT5682S_I2S_M_D_MASK (0xf << 8) +#define RT5682S_I2S_M_D_SFT 8 +#define RT5682S_I2S_M_D_1 (0x0 << 8) +#define RT5682S_I2S_M_D_2 (0x1 << 8) +#define RT5682S_I2S_M_D_3 (0x2 << 8) +#define RT5682S_I2S_M_D_4 (0x3 << 8) +#define RT5682S_I2S_M_D_6 (0x4 << 8) +#define RT5682S_I2S_M_D_8 (0x5 << 8) +#define RT5682S_I2S_M_D_12 (0x6 << 8) +#define RT5682S_I2S_M_D_16 (0x7 << 8) +#define RT5682S_I2S_M_D_24 (0x8 << 8) +#define RT5682S_I2S_M_D_32 (0x9 << 8) +#define RT5682S_I2S_M_D_48 (0x10 << 8) +#define RT5682S_I2S_M_CLK_SRC_MASK (0x7 << 4) +#define RT5682S_I2S_M_CLK_SRC_SFT 4 +#define RT5682S_DAC_OSR_MASK (0xf << 0) +#define RT5682S_DAC_OSR_SFT 0 +#define RT5682S_DAC_OSR_D_1 (0x0 << 0) +#define RT5682S_DAC_OSR_D_2 (0x1 << 0) +#define RT5682S_DAC_OSR_D_4 (0x2 << 0) +#define RT5682S_DAC_OSR_D_6 (0x3 << 0) +#define RT5682S_DAC_OSR_D_8 (0x4 << 0) +#define RT5682S_DAC_OSR_D_12 (0x5 << 0) +#define RT5682S_DAC_OSR_D_16 (0x6 << 0) +#define RT5682S_DAC_OSR_D_24 (0x7 << 0) +#define RT5682S_DAC_OSR_D_32 (0x8 << 0) +#define RT5682S_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682S_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682S_I2S2_BCLK_MS2_SFT 11 +#define RT5682S_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682S_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682S_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682S_TDM_TX_CH_2 (0x0 << 12) +#define RT5682S_TDM_TX_CH_4 (0x1 << 12) +#define RT5682S_TDM_TX_CH_6 (0x2 << 12) +#define RT5682S_TDM_TX_CH_8 (0x3 << 12) +#define RT5682S_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682S_TDM_RX_CH_2 (0x0 << 8) +#define RT5682S_TDM_RX_CH_4 (0x1 << 8) +#define RT5682S_TDM_RX_CH_6 (0x2 << 8) +#define RT5682S_TDM_RX_CH_8 (0x3 << 8) +#define RT5682S_TDM_ADC_LCA_MASK (0x7 << 4) +#define RT5682S_TDM_ADC_LCA_SFT 4 +#define RT5682S_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682S_IF1_ADC1_SEL_SFT 14 +#define RT5682S_IF1_ADC2_SEL_SFT 12 +#define RT5682S_IF1_ADC3_SEL_SFT 10 +#define RT5682S_IF1_ADC4_SEL_SFT 8 +#define RT5682S_TDM_ADC_SEL_SFT 3 + +/* TDM control 3 (0x007b) */ +#define RT5682S_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682S_TDM_S_BP_MASK (0x1 << 15) +#define RT5682S_TDM_S_BP_SFT 15 +#define RT5682S_TDM_S_BP_NOR (0x0 << 15) +#define RT5682S_TDM_S_BP_INV (0x1 << 15) +#define RT5682S_TDM_S_LP_MASK (0x1 << 14) +#define RT5682S_TDM_S_LP_SFT 14 +#define RT5682S_TDM_S_LP_NOR (0x0 << 14) +#define RT5682S_TDM_S_LP_INV (0x1 << 14) +#define RT5682S_TDM_DF_MASK (0x7 << 11) +#define RT5682S_TDM_DF_SFT 11 +#define RT5682S_TDM_DF_I2S (0x0 << 11) +#define RT5682S_TDM_DF_LEFT (0x1 << 11) +#define RT5682S_TDM_DF_PCM_A (0x2 << 11) +#define RT5682S_TDM_DF_PCM_B (0x3 << 11) +#define RT5682S_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682S_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682S_TDM_BCLK_MS1_MASK (0x3 << 8) +#define RT5682S_TDM_BCLK_MS1_SFT 8 +#define RT5682S_TDM_BCLK_MS1_32 (0x0 << 8) +#define RT5682S_TDM_BCLK_MS1_64 (0x1 << 8) +#define RT5682S_TDM_BCLK_MS1_128 (0x2 << 8) +#define RT5682S_TDM_BCLK_MS1_256 (0x3 << 8) +#define RT5682S_TDM_BCLK_MS1_16 (0x4 << 8) +#define RT5682S_TDM_CL_MASK (0x3 << 4) +#define RT5682S_TDM_CL_16 (0x0 << 4) +#define RT5682S_TDM_CL_20 (0x1 << 4) +#define RT5682S_TDM_CL_24 (0x2 << 4) +#define RT5682S_TDM_CL_32 (0x3 << 4) +#define RT5682S_TDM_M_BP_MASK (0x1 << 2) +#define RT5682S_TDM_M_BP_SFT 2 +#define RT5682S_TDM_M_BP_NOR (0x0 << 2) +#define RT5682S_TDM_M_BP_INV (0x1 << 2) +#define RT5682S_TDM_M_LP_MASK (0x1 << 1) +#define RT5682S_TDM_M_LP_SFT 1 +#define RT5682S_TDM_M_LP_NOR (0x0 << 1) +#define RT5682S_TDM_M_LP_INV (0x1 << 1) +#define RT5682S_TDM_MS_MASK (0x1 << 0) +#define RT5682S_TDM_MS_SFT 0 +#define RT5682S_TDM_MS_S (0x0 << 0) +#define RT5682S_TDM_MS_M (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682S_SCLK_SRC_MASK (0x7 << 13) +#define RT5682S_SCLK_SRC_SFT 13 +#define RT5682S_PLL_SRC_MASK (0x3 << 8) +#define RT5682S_PLL_SRC_SFT 8 +#define RT5682S_PLL_SRC_MCLK (0x0 << 8) +#define RT5682S_PLL_SRC_BCLK1 (0x1 << 8) +#define RT5682S_PLL_SRC_RC (0x3 << 8) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682S_DA_ASRC_MASK (0x1 << 13) +#define RT5682S_DA_ASRC_SFT 13 +#define RT5682S_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682S_DAC_STO1_ASRC_SFT 12 +#define RT5682S_AD_ASRC_MASK (0x1 << 8) +#define RT5682S_AD_ASRC_SFT 8 +#define RT5682S_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682S_AD_ASRC_SEL_SFT 4 +#define RT5682S_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682S_DMIC_ASRC_SFT 3 +#define RT5682S_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682S_ADC_STO1_ASRC_SFT 2 +#define RT5682S_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682S_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682S_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682S_FILTER_CLK_SEL_SFT 12 +#define RT5682S_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682S_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682S_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682S_ASRCIN_FTK_N1_SFT 14 +#define RT5682S_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682S_ASRCIN_FTK_N2_SFT 12 +#define RT5682S_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682S_ASRCIN_FTK_M1_SFT 8 +#define RT5682S_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682S_ASRCIN_FTK_M2_SFT 4 + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682S_OUT_HP_L_EN (0x1 << 6) +#define RT5682S_OUT_HP_R_EN (0x1 << 5) +#define RT5682S_LDO_PUMP_EN (0x1 << 4) +#define RT5682S_LDO_PUMP_EN_SFT 4 +#define RT5682S_PUMP_EN (0x1 << 3) +#define RT5682S_PUMP_EN_SFT 3 +#define RT5682S_CAPLESS_L_EN (0x1 << 1) +#define RT5682S_CAPLESS_L_EN_SFT 1 +#define RT5682S_CAPLESS_R_EN (0x1 << 0) +#define RT5682S_CAPLESS_R_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682S_RAMP_MASK (0x1 << 12) +#define RT5682S_RAMP_SFT 12 +#define RT5682S_RAMP_DIS (0x0 << 12) +#define RT5682S_RAMP_EN (0x1 << 12) +#define RT5682S_BPS_MASK (0x1 << 11) +#define RT5682S_BPS_SFT 11 +#define RT5682S_BPS_DIS (0x0 << 11) +#define RT5682S_BPS_EN (0x1 << 11) +#define RT5682S_FAST_UPDN_MASK (0x1 << 10) +#define RT5682S_FAST_UPDN_SFT 10 +#define RT5682S_FAST_UPDN_DIS (0x0 << 10) +#define RT5682S_FAST_UPDN_EN (0x1 << 10) +#define RT5682S_VLO_MASK (0x1 << 7) +#define RT5682S_VLO_SFT 7 +#define RT5682S_VLO_3V (0x0 << 7) +#define RT5682S_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682S_OSW_L_MASK (0x1 << 11) +#define RT5682S_OSW_L_SFT 11 +#define RT5682S_OSW_L_DIS (0x0 << 11) +#define RT5682S_OSW_L_EN (0x1 << 11) +#define RT5682S_OSW_R_MASK (0x1 << 10) +#define RT5682S_OSW_R_SFT 10 +#define RT5682S_OSW_R_DIS (0x0 << 10) +#define RT5682S_OSW_R_EN (0x1 << 10) +#define RT5682S_PM_HP_MASK (0x3 << 8) +#define RT5682S_PM_HP_SFT 8 +#define RT5682S_PM_HP_LV (0x0 << 8) +#define RT5682S_PM_HP_MV (0x1 << 8) +#define RT5682S_PM_HP_HV (0x2 << 8) + +/* Micbias Control1 (0x93) */ +#define RT5682S_MIC1_OV_MASK (0x3 << 14) +#define RT5682S_MIC1_OV_SFT 14 +#define RT5682S_MIC1_OV_2V7 (0x0 << 14) +#define RT5682S_MIC1_OV_2V4 (0x1 << 14) +#define RT5682S_MIC1_OV_2V25 (0x3 << 14) +#define RT5682S_MIC1_OV_1V8 (0x4 << 14) +#define RT5682S_MIC2_OV_MASK (0x3 << 8) +#define RT5682S_MIC2_OV_SFT 8 +#define RT5682S_MIC2_OV_2V7 (0x0 << 8) +#define RT5682S_MIC2_OV_2V4 (0x1 << 8) +#define RT5682S_MIC2_OV_2V25 (0x3 << 8) +#define RT5682S_MIC2_OV_1V8 (0x4 << 8) + +/* Micbias Control2 (0x0094) */ +#define RT5682S_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682S_PWR_CLK25M_SFT 9 +#define RT5682S_PWR_CLK25M_PD (0x0 << 9) +#define RT5682S_PWR_CLK25M_PU (0x1 << 9) +#define RT5682S_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682S_PWR_CLK1M_SFT 8 +#define RT5682S_PWR_CLK1M_PD (0x0 << 8) +#define RT5682S_PWR_CLK1M_PU (0x1 << 8) + +/* PLL M/N/K Code Control 1 (0x0098) */ +#define RT5682S_PLLA_N_MASK (0x1ff << 0) + +/* PLL M/N/K Code Control 2 (0x0099) */ +#define RT5682S_PLLA_M_MASK (0x1f << 8) +#define RT5682S_PLLA_M_SFT 8 +#define RT5682S_PLLA_K_MASK (0x1f << 0) + +/* PLL M/N/K Code Control 3 (0x009a) */ +#define RT5682S_PLLB_N_MASK (0x3ff << 0) + +/* PLL M/N/K Code Control 4 (0x009b) */ +#define RT5682S_PLLB_M_MASK (0x1f << 8) +#define RT5682S_PLLB_M_SFT 8 +#define RT5682S_PLLB_K_MASK (0x1f << 0) + +/* PLL M/N/K Code Control 6 (0x009d) */ +#define RT5682S_PLLB_SEL_PS_MASK (0x1 << 13) +#define RT5682S_PLLB_SEL_PS_SFT 13 +#define RT5682S_PLLB_BYP_PS_MASK (0x1 << 12) +#define RT5682S_PLLB_BYP_PS_SFT 12 +#define RT5682S_PLLB_M_BP_MASK (0x1 << 11) +#define RT5682S_PLLB_M_BP_SFT 11 +#define RT5682S_PLLB_K_BP_MASK (0x1 << 10) +#define RT5682S_PLLB_K_BP_SFT 10 +#define RT5682S_PLLA_M_BP_MASK (0x1 << 7) +#define RT5682S_PLLA_M_BP_SFT 7 +#define RT5682S_PLLA_K_BP_MASK (0x1 << 6) +#define RT5682S_PLLA_K_BP_SFT 6 + +/* PLL M/N/K Code Control 7 (0x009e) */ +#define RT5682S_PLLB_SRC_MASK (0x3 << 0) +#define RT5682S_PLLB_SRC_DFIN (0x1) +#define RT5682S_PLLB_SRC_PLLA (0x0) + +/* RC Clock Control (0x009f) */ +#define RT5682S_POW_IRQ (0x1 << 15) +#define RT5682S_POW_JDH (0x1 << 14) + +/* I2S2 Master Mode Clock Control 1 (0x00a0) */ +#define RT5682S_I2S2_M_CLK_SRC_MASK (0x7 << 4) +#define RT5682S_I2S2_M_CLK_SRC_SFT 4 +#define RT5682S_I2S2_M_D_MASK (0xf << 0) +#define RT5682S_I2S2_M_D_1 (0x0) +#define RT5682S_I2S2_M_D_2 (0x1) +#define RT5682S_I2S2_M_D_3 (0x2) +#define RT5682S_I2S2_M_D_4 (0x3) +#define RT5682S_I2S2_M_D_6 (0x4) +#define RT5682S_I2S2_M_D_8 (0x5) +#define RT5682S_I2S2_M_D_12 (0x6) +#define RT5682S_I2S2_M_D_16 (0x7) +#define RT5682S_I2S2_M_D_24 (0x8) +#define RT5682S_I2S2_M_D_32 (0x9) +#define RT5682S_I2S2_M_D_48 (0xa) +#define RT5682S_I2S2_M_D_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682S_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682S_JD1_PULSE_EN_SFT 10 +#define RT5682S_JD1_PULSE_DIS (0x0 << 10) +#define RT5682S_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682S_JD1_EN_MASK (0x1 << 15) +#define RT5682S_JD1_EN_SFT 15 +#define RT5682S_JD1_DIS (0x0 << 15) +#define RT5682S_JD1_EN (0x1 << 15) +#define RT5682S_JD1_POL_MASK (0x1 << 13) +#define RT5682S_JD1_POL_NOR (0x0 << 13) +#define RT5682S_JD1_POL_INV (0x1 << 13) +#define RT5682S_JD1_IRQ_MASK (0x1 << 10) +#define RT5682S_JD1_IRQ_LEV (0x0 << 10) +#define RT5682S_JD1_IRQ_PUL (0x1 << 10) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682S_IL_IRQ_MASK (0x1 << 7) +#define RT5682S_IL_IRQ_DIS (0x0 << 7) +#define RT5682S_IL_IRQ_EN (0x1 << 7) +#define RT5682S_IL_IRQ_TYPE_MASK (0x1 << 4) +#define RT5682S_IL_IRQ_LEV (0x0 << 4) +#define RT5682S_IL_IRQ_PUL (0x1 << 4) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682S_GP1_PIN_MASK (0x3 << 14) +#define RT5682S_GP1_PIN_SFT 14 +#define RT5682S_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682S_GP1_PIN_IRQ (0x1 << 14) +#define RT5682S_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682S_GP2_PIN_MASK (0x3 << 12) +#define RT5682S_GP2_PIN_SFT 12 +#define RT5682S_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682S_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682S_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682S_GP3_PIN_MASK (0x3 << 10) +#define RT5682S_GP3_PIN_SFT 10 +#define RT5682S_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682S_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682S_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682S_GP4_PIN_MASK (0x3 << 8) +#define RT5682S_GP4_PIN_SFT 8 +#define RT5682S_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682S_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682S_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682S_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682S_GP5_PIN_MASK (0x3 << 6) +#define RT5682S_GP5_PIN_SFT 6 +#define RT5682S_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682S_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682S_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682S_GP6_PIN_MASK (0x1 << 5) +#define RT5682S_GP6_PIN_SFT 5 +#define RT5682S_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682S_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682S_GP1_PF_MASK (0x1 << 15) +#define RT5682S_GP1_PF_IN (0x0 << 15) +#define RT5682S_GP1_PF_OUT (0x1 << 15) +#define RT5682S_GP1_OUT_MASK (0x1 << 14) +#define RT5682S_GP1_OUT_L (0x0 << 14) +#define RT5682S_GP1_OUT_H (0x1 << 14) +#define RT5682S_GP2_PF_MASK (0x1 << 13) +#define RT5682S_GP2_PF_IN (0x0 << 13) +#define RT5682S_GP2_PF_OUT (0x1 << 13) +#define RT5682S_GP2_OUT_MASK (0x1 << 12) +#define RT5682S_GP2_OUT_L (0x0 << 12) +#define RT5682S_GP2_OUT_H (0x1 << 12) +#define RT5682S_GP3_PF_MASK (0x1 << 11) +#define RT5682S_GP3_PF_IN (0x0 << 11) +#define RT5682S_GP3_PF_OUT (0x1 << 11) +#define RT5682S_GP3_OUT_MASK (0x1 << 10) +#define RT5682S_GP3_OUT_L (0x0 << 10) +#define RT5682S_GP3_OUT_H (0x1 << 10) +#define RT5682S_GP4_PF_MASK (0x1 << 9) +#define RT5682S_GP4_PF_IN (0x0 << 9) +#define RT5682S_GP4_PF_OUT (0x1 << 9) +#define RT5682S_GP4_OUT_MASK (0x1 << 8) +#define RT5682S_GP4_OUT_L (0x0 << 8) +#define RT5682S_GP4_OUT_H (0x1 << 8) +#define RT5682S_GP5_PF_MASK (0x1 << 7) +#define RT5682S_GP5_PF_IN (0x0 << 7) +#define RT5682S_GP5_PF_OUT (0x1 << 7) +#define RT5682S_GP5_OUT_MASK (0x1 << 6) +#define RT5682S_GP5_OUT_L (0x0 << 6) +#define RT5682S_GP5_OUT_H (0x1 << 6) +#define RT5682S_GP6_PF_MASK (0x1 << 5) +#define RT5682S_GP6_PF_IN (0x0 << 5) +#define RT5682S_GP6_PF_OUT (0x1 << 5) +#define RT5682S_GP6_OUT_MASK (0x1 << 4) +#define RT5682S_GP6_OUT_L (0x0 << 4) +#define RT5682S_GP6_OUT_H (0x1 << 4) + +/* GPIO Status (0x00c2) */ +#define RT5682S_GP6_ST (0x1 << 6) +#define RT5682S_GP5_ST (0x1 << 5) +#define RT5682S_GP4_ST (0x1 << 4) +#define RT5682S_GP3_ST (0x1 << 3) +#define RT5682S_GP2_ST (0x1 << 2) +#define RT5682S_GP1_ST (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682S_ZCD_MASK (0x1 << 10) +#define RT5682S_ZCD_SFT 10 +#define RT5682S_ZCD_PD (0x0 << 10) +#define RT5682S_ZCD_PU (0x1 << 10) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682S_4BTN_IL_MASK (0x1 << 15) +#define RT5682S_4BTN_IL_EN (0x1 << 15) +#define RT5682S_4BTN_IL_DIS (0x0 << 15) +#define RT5682S_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682S_4BTN_IL_NOR (0x1 << 14) +#define RT5682S_4BTN_IL_RST (0x0 << 14) + +/* 4 Button Inline Command Control 3~6 (0x00e5~0x00e8) */ +#define RT5682S_4BTN_IL_HOLD_WIN_MASK (0x7f << 8) +#define RT5682S_4BTN_IL_HOLD_WIN_SFT 8 +#define RT5682S_4BTN_IL_CLICK_WIN_MASK (0x7f) +#define RT5682S_4BTN_IL_CLICK_WIN_SFT 0 + +/* Analog JD Control (0x00f0) */ +#define RT5682S_JDH_RS_MASK (0x1 << 4) +#define RT5682S_JDH_NO_PLUG (0x1 << 4) +#define RT5682S_JDH_PLUG (0x0 << 4) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682S_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682S_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682S_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682S_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682S_CP_CLK_HP_600KHZ (0x3 << 4) + +/* Pad Driving Control (0x0136) */ +#define RT5682S_PAD_DRV_GP1_MASK (0x1 << 14) +#define RT5682S_PAD_DRV_GP1_HIGH (0x1 << 14) +#define RT5682S_PAD_DRV_GP1_LOW (0x0 << 14) +#define RT5682S_PAD_DRV_GP2_MASK (0x1 << 12) +#define RT5682S_PAD_DRV_GP2_HIGH (0x1 << 12) +#define RT5682S_PAD_DRV_GP2_LOW (0x0 << 12) +#define RT5682S_PAD_DRV_GP3_MASK (0x1 << 10) +#define RT5682S_PAD_DRV_GP3_HIGH (0x1 << 10) +#define RT5682S_PAD_DRV_GP3_LOW (0x0 << 10) +#define RT5682S_PAD_DRV_GP4_MASK (0x1 << 8) +#define RT5682S_PAD_DRV_GP4_HIGH (0x1 << 8) +#define RT5682S_PAD_DRV_GP4_LOW (0x0 << 8) +#define RT5682S_PAD_DRV_GP5_MASK (0x1 << 6) +#define RT5682S_PAD_DRV_GP5_HIGH (0x1 << 6) +#define RT5682S_PAD_DRV_GP5_LOW (0x0 << 6) +#define RT5682S_PAD_DRV_GP6_MASK (0x1 << 4) +#define RT5682S_PAD_DRV_GP6_HIGH (0x1 << 4) +#define RT5682S_PAD_DRV_GP6_LOW (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682S_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682S_CKXEN_DAC1_SFT 13 +#define RT5682S_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682S_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682S_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682S_CKXEN_ADC1_SFT 13 +#define RT5682S_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682S_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682S_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682S_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682S_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682S_AD2DA_LB_MASK (0x1 << 10) +#define RT5682S_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682S_NG2_EN_MASK (0x1 << 15) +#define RT5682S_NG2_EN (0x1 << 15) +#define RT5682S_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682S_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682S_DEB_80_MS (0x0 << 4) + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5682S_HP_SIG_SRC_MASK (0x3) +#define RT5682S_HP_SIG_SRC_1BIT_CTL (0x3) +#define RT5682S_HP_SIG_SRC_REG (0x2) +#define RT5682S_HP_SIG_SRC_IMPE_REG (0x1) +#define RT5682S_HP_SIG_SRC_DC_CALI (0x0) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682S_SAR_BUTDET_MASK (0x1 << 15) +#define RT5682S_SAR_BUTDET_EN (0x1 << 15) +#define RT5682S_SAR_BUTDET_DIS (0x0 << 15) +#define RT5682S_SAR_BUTDET_POW_MASK (0x1 << 14) +#define RT5682S_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682S_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682S_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682S_SAR_BUTDET_RST_NORM (0x1 << 13) +#define RT5682S_SAR_BUTDET_RST (0x0 << 13) +#define RT5682S_SAR_POW_MASK (0x1 << 12) +#define RT5682S_SAR_POW_EN (0x1 << 12) +#define RT5682S_SAR_POW_DIS (0x0 << 12) +#define RT5682S_SAR_RST_MASK (0x1 << 11) +#define RT5682S_SAR_RST_NORMAL (0x1 << 11) +#define RT5682S_SAR_RST (0x0 << 11) +#define RT5682S_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682S_SAR_BYPASS_EN (0x1 << 10) +#define RT5682S_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682S_SAR_SEL_MB1_2_MASK (0x3 << 8) +#define RT5682S_SAR_SEL_MB1_2_SFT 8 +#define RT5682S_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682S_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682S_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682S_SAR_SEL_MB1_2_CTL_MASK (0x1 << 5) +#define RT5682S_SAR_SEL_MB1_2_AUTO (0x1 << 5) +#define RT5682S_SAR_SEL_MB1_2_MANU (0x0 << 5) +#define RT5682S_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682S_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682S_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 2 (0x0211) */ +#define RT5682S_SAR_ADC_PSV_MASK (0x1 << 4) +#define RT5682S_SAR_ADC_PSV_ENTRY (0x1 << 4) + + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682S_SAR_SOUR_MASK (0x3f) +#define RT5682S_SAR_SOUR_BTN (0x3f) +#define RT5682S_SAR_SOUR_TYPE (0x0) + + +#define RT5682S_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +/* System Clock Source */ +enum { + RT5682S_SCLK_S_MCLK, + RT5682S_SCLK_S_PLL1, + RT5682S_SCLK_S_PLL2, + RT5682S_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682S_PLL_S_MCLK, + RT5682S_PLL_S_BCLK1, + RT5682S_PLL_S_BCLK2, + RT5682S_PLL_S_RCCLK, +}; + +enum { + RT5682S_PLL1, + RT5682S_PLL2, + RT5682S_PLLS, +}; + +enum { + RT5682S_AIF1, + RT5682S_AIF2, + RT5682S_AIFS +}; + +/* filter mask */ +enum { + RT5682S_DA_STEREO1_FILTER = 0x1, + RT5682S_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682S_CLK_SEL_SYS, + RT5682S_CLK_SEL_I2S1_ASRC, + RT5682S_CLK_SEL_I2S2_ASRC, +}; + +enum { + USE_PLLA, + USE_PLLB, + USE_PLLAB, +}; + +struct pll_calc_map { + unsigned int freq_in; + unsigned int freq_out; + int m; + int n; + int k; + bool m_bp; + bool k_bp; + bool byp_ps; + bool sel_ps; +}; + +#define RT5682S_NUM_SUPPLIES 2 + +struct rt5682s_priv { + struct snd_soc_component *component; + struct rt5682s_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682S_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + struct mutex sar_mutex; + +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682S_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + + int sysclk; + int sysclk_src; + int lrck[RT5682S_AIFS]; + int bclk[RT5682S_AIFS]; + int master[RT5682S_AIFS]; + + int pll_src[RT5682S_PLLS]; + int pll_in[RT5682S_PLLS]; + int pll_out[RT5682S_PLLS]; + int pll_comb; + + int jack_type; + int irq_work_delay_time; +}; + +int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682S_H__ */ -- 2.17.1

5 7

[PATCH v5 00/11] ASoC: mediatek: Add support for MT8195 SoC
by Trevor Wu 31 Aug '21

31 Aug '21

This series of patches adds support for Mediatek AFE of MT8195 SoC. Patches are based on broonie tree "for-next" branch. Changes since v4: - removed sof related code Changes since v3: - fixed warnings found by kernel test robot - removed unused critical section - corrected the lock protected sections on etdm driver - added DPTX and HDMITX audio support Changes since v2: - added audio clock gate control - added 'mediatek' prefix to private dts properties - added consumed clocks to dt-bindins and adopted suggestions from Rob - refined clock usage and remove unused clock and control code - fixed typos Changes since v1: - fixed some problems related to dt-bindings - added some missing properties to dt-bindings - added depency declaration on dt-bindings - fixed some warnings found by kernel test robot Trevor Wu (11): ASoC: mediatek: mt8195: update mediatek common driver ASoC: mediatek: mt8195: support audsys clock control ASoC: mediatek: mt8195: support etdm in platform driver ASoC: mediatek: mt8195: support adda in platform driver ASoC: mediatek: mt8195: support pcm in platform driver ASoC: mediatek: mt8195: add platform driver dt-bindings: mediatek: mt8195: add audio afe document ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 ASoC: mediatek: mt8195: add DPTX audio support ASoC: mediatek: mt8195: add HDMITX audio support dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682 document .../bindings/sound/mt8195-afe-pcm.yaml | 184 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 24 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 22 +- sound/soc/mediatek/common/mtk-base-afe.h | 10 +- sound/soc/mediatek/mt8195/Makefile | 15 + sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 441 +++ sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 109 + sound/soc/mediatek/mt8195/mt8195-afe-common.h | 158 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3281 +++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 214 ++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 15 + .../soc/mediatek/mt8195/mt8195-audsys-clkid.h | 93 + sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 830 +++++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2639 +++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 389 ++ .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 1087 ++++++ sound/soc/mediatek/mt8195/mt8195-reg.h | 2796 ++++++++++++++ 19 files changed, 12350 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1019-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/Makefile create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-adda.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h -- 2.18.0

3 14

[sound:test/usb-audio-lowlatency 198/198] sound/usb/pcm.c:1322:12: sparse: sparse: context imbalance in 'prepare_playback_urb' - different lock contexts for basic block
by kernel test robot 31 Aug '21

31 Aug '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/usb-audio-lowlatency head: 0dbf51c865c64e95c711e4adf89a89140077269e commit: 0dbf51c865c64e95c711e4adf89a89140077269e [198/198] ALSA: usb-audio: Improved lowlatency playback support :::::: branch date: 17 hours ago :::::: commit date: 17 hours ago config: arc-randconfig-s032-20210831 (attached as .config) compiler: arc-elf-gcc (GCC) 11.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # apt-get install sparse # sparse version: v0.6.3-348-gf0e6938b-dirty # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound test/usb-audio-lowlatency git checkout 0dbf51c865c64e95c711e4adf89a89140077269e # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-11.2.0 make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' O=build_dir ARCH=arc SHELL=/bin/bash sound/usb/ If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/usb/pcm.c:1322:12: sparse: sparse: context imbalance in 'prepare_playback_urb' - different lock contexts for basic block vim +/prepare_playback_urb +1322 sound/usb/pcm.c e0570446773623 Ricard Wanderlof 2015-10-19 1321 0dbf51c865c64e Takashi Iwai 2021-08-30 @1322 static int prepare_playback_urb(struct snd_usb_substream *subs, edcd3633e72a15 Daniel Mack 2012-04-12 1323 struct urb *urb) edcd3633e72a15 Daniel Mack 2012-04-12 1324 { edcd3633e72a15 Daniel Mack 2012-04-12 1325 struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; 245baf983cc395 Daniel Mack 2012-08-30 1326 struct snd_usb_endpoint *ep = subs->data_endpoint; edcd3633e72a15 Daniel Mack 2012-04-12 1327 struct snd_urb_ctx *ctx = urb->context; 0dbf51c865c64e Takashi Iwai 2021-08-30 1328 unsigned int counts, frames, bytes, transfer_done; edcd3633e72a15 Daniel Mack 2012-04-12 1329 int i, stride, period_elapsed = 0; edcd3633e72a15 Daniel Mack 2012-04-12 1330 unsigned long flags; edcd3633e72a15 Daniel Mack 2012-04-12 1331 d303c5d38b37ee Takashi Iwai 2021-06-01 1332 stride = ep->stride; edcd3633e72a15 Daniel Mack 2012-04-12 1333 edcd3633e72a15 Daniel Mack 2012-04-12 1334 frames = 0; e8a8f09cb0b3b8 Takashi Iwai 2021-06-01 1335 ctx->queued = 0; edcd3633e72a15 Daniel Mack 2012-04-12 1336 urb->number_of_packets = 0; edcd3633e72a15 Daniel Mack 2012-04-12 1337 spin_lock_irqsave(&subs->lock, flags); 976b6c064a9574 Alan Stern 2013-09-24 1338 subs->frame_limit += ep->max_urb_frames; 0dbf51c865c64e Takashi Iwai 2021-08-30 1339 transfer_done = subs->transfer_done; edcd3633e72a15 Daniel Mack 2012-04-12 1340 for (i = 0; i < ctx->packets; i++) { 3d58760f4d0015 Takashi Iwai 2020-11-23 1341 counts = snd_usb_endpoint_next_packet_size(ep, ctx, i); edcd3633e72a15 Daniel Mack 2012-04-12 1342 /* set up descriptor */ d303c5d38b37ee Takashi Iwai 2021-06-01 1343 urb->iso_frame_desc[i].offset = frames * stride; d303c5d38b37ee Takashi Iwai 2021-06-01 1344 urb->iso_frame_desc[i].length = counts * stride; edcd3633e72a15 Daniel Mack 2012-04-12 1345 frames += counts; edcd3633e72a15 Daniel Mack 2012-04-12 1346 urb->number_of_packets++; 0dbf51c865c64e Takashi Iwai 2021-08-30 1347 transfer_done += counts; 0dbf51c865c64e Takashi Iwai 2021-08-30 1348 if (transfer_done >= runtime->period_size) { 0dbf51c865c64e Takashi Iwai 2021-08-30 1349 transfer_done -= runtime->period_size; 976b6c064a9574 Alan Stern 2013-09-24 1350 subs->frame_limit = 0; edcd3633e72a15 Daniel Mack 2012-04-12 1351 period_elapsed = 1; edcd3633e72a15 Daniel Mack 2012-04-12 1352 if (subs->fmt_type == UAC_FORMAT_TYPE_II) { 0dbf51c865c64e Takashi Iwai 2021-08-30 1353 if (transfer_done > 0) { edcd3633e72a15 Daniel Mack 2012-04-12 1354 /* FIXME: fill-max mode is not edcd3633e72a15 Daniel Mack 2012-04-12 1355 * supported yet */ 0dbf51c865c64e Takashi Iwai 2021-08-30 1356 frames -= transfer_done; 0dbf51c865c64e Takashi Iwai 2021-08-30 1357 counts -= transfer_done; edcd3633e72a15 Daniel Mack 2012-04-12 1358 urb->iso_frame_desc[i].length = d303c5d38b37ee Takashi Iwai 2021-06-01 1359 counts * stride; 0dbf51c865c64e Takashi Iwai 2021-08-30 1360 transfer_done = 0; edcd3633e72a15 Daniel Mack 2012-04-12 1361 } edcd3633e72a15 Daniel Mack 2012-04-12 1362 i++; edcd3633e72a15 Daniel Mack 2012-04-12 1363 if (i < ctx->packets) { edcd3633e72a15 Daniel Mack 2012-04-12 1364 /* add a transfer delimiter */ edcd3633e72a15 Daniel Mack 2012-04-12 1365 urb->iso_frame_desc[i].offset = d303c5d38b37ee Takashi Iwai 2021-06-01 1366 frames * stride; edcd3633e72a15 Daniel Mack 2012-04-12 1367 urb->iso_frame_desc[i].length = 0; edcd3633e72a15 Daniel Mack 2012-04-12 1368 urb->number_of_packets++; edcd3633e72a15 Daniel Mack 2012-04-12 1369 } edcd3633e72a15 Daniel Mack 2012-04-12 1370 break; edcd3633e72a15 Daniel Mack 2012-04-12 1371 } edcd3633e72a15 Daniel Mack 2012-04-12 1372 } 976b6c064a9574 Alan Stern 2013-09-24 1373 /* finish at the period boundary or after enough frames */ 0dbf51c865c64e Takashi Iwai 2021-08-30 1374 if ((period_elapsed || transfer_done >= subs->frame_limit) && 976b6c064a9574 Alan Stern 2013-09-24 1375 !snd_usb_endpoint_implicit_feedback_sink(ep)) edcd3633e72a15 Daniel Mack 2012-04-12 1376 break; edcd3633e72a15 Daniel Mack 2012-04-12 1377 } d24f5061ee7b9b Daniel Mack 2013-04-17 1378 0dbf51c865c64e Takashi Iwai 2021-08-30 1379 /* check the appl_ptr in low latency mode */ 0dbf51c865c64e Takashi Iwai 2021-08-30 1380 if (!subs->early_playback_start && 0dbf51c865c64e Takashi Iwai 2021-08-30 1381 frames > snd_pcm_playback_hw_avail(runtime)) 0dbf51c865c64e Takashi Iwai 2021-08-30 @1382 return -EAGAIN; 0dbf51c865c64e Takashi Iwai 2021-08-30 1383 0dbf51c865c64e Takashi Iwai 2021-08-30 1384 bytes = frames * stride; 0dbf51c865c64e Takashi Iwai 2021-08-30 1385 subs->transfer_done = transfer_done; 6aa719d15a1903 Takashi Iwai 2020-11-23 1386 if (unlikely(ep->cur_format == SNDRV_PCM_FORMAT_DSD_U16_LE && d24f5061ee7b9b Daniel Mack 2013-04-17 1387 subs->cur_audiofmt->dsd_dop)) { d24f5061ee7b9b Daniel Mack 2013-04-17 1388 fill_playback_urb_dsd_dop(subs, urb, bytes); 6aa719d15a1903 Takashi Iwai 2020-11-23 1389 } else if (unlikely(ep->cur_format == SNDRV_PCM_FORMAT_DSD_U8 && 44dcbbb1cd6156 Daniel Mack 2013-04-17 1390 subs->cur_audiofmt->dsd_bitrev)) { 4f083917994fdd Takashi Iwai 2021-06-01 1391 fill_playback_urb_dsd_bitrev(subs, urb, bytes); d24f5061ee7b9b Daniel Mack 2013-04-17 1392 } else { d24f5061ee7b9b Daniel Mack 2013-04-17 1393 /* usual PCM */ e0570446773623 Ricard Wanderlof 2015-10-19 1394 if (!subs->tx_length_quirk) b97a936910c8d6 Ricard Wanderlof 2015-10-19 1395 copy_to_urb(subs, urb, 0, stride, bytes); e0570446773623 Ricard Wanderlof 2015-10-19 1396 else e0570446773623 Ricard Wanderlof 2015-10-19 1397 bytes = copy_to_urb_quirk(subs, urb, stride, bytes); e0570446773623 Ricard Wanderlof 2015-10-19 1398 /* bytes is now amount of outgoing data */ d24f5061ee7b9b Daniel Mack 2013-04-17 1399 } d24f5061ee7b9b Daniel Mack 2013-04-17 1400 fbcfbf5f673847 Daniel Mack 2012-08-30 1401 subs->last_frame_number = usb_get_current_frame_number(subs->dev); fbcfbf5f673847 Daniel Mack 2012-08-30 1402 ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1403 if (subs->trigger_tstamp_pending_update) { ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1404 /* this is the first actual URB submitted, ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1405 * update trigger timestamp to reflect actual start time ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1406 */ ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1407 snd_pcm_gettime(runtime, &runtime->trigger_tstamp); ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1408 subs->trigger_tstamp_pending_update = false; ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1409 } ea33d359c4421f Pierre-Louis Bossart 2015-02-06 1410 4267c5a8f3133d Takashi Iwai 2021-08-27 1411 if (period_elapsed && !subs->running && !subs->early_playback_start) { 307cc9baac5c07 Takashi Iwai 2021-07-07 1412 subs->period_elapsed_pending = 1; 307cc9baac5c07 Takashi Iwai 2021-07-07 1413 period_elapsed = 0; 307cc9baac5c07 Takashi Iwai 2021-07-07 1414 } edcd3633e72a15 Daniel Mack 2012-04-12 1415 spin_unlock_irqrestore(&subs->lock, flags); edcd3633e72a15 Daniel Mack 2012-04-12 1416 urb->transfer_buffer_length = bytes; edcd3633e72a15 Daniel Mack 2012-04-12 1417 if (period_elapsed) edcd3633e72a15 Daniel Mack 2012-04-12 1418 snd_pcm_period_elapsed(subs->pcm_substream); 0dbf51c865c64e Takashi Iwai 2021-08-30 1419 return 0; edcd3633e72a15 Daniel Mack 2012-04-12 1420 } edcd3633e72a15 Daniel Mack 2012-04-12 1421 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

Typo in alsamixer(1) manpage
by GitHub issues - opened 31 Aug '21

31 Aug '21

alsa-project/alsa-utils issue #110 was opened from climbTheStairs: In the second paragraph of the "MIXER VIEWS" section, it says... > The currently selected item is drawn in red and/of emphasized. ...when it should say... > The currently selected item is drawn in red **and/or** emphasized. Issue URL : https://github.com/alsa-project/alsa-utils/issues/110 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL] ASoC updates for v5.15
by Mark Brown 30 Aug '21

30 Aug '21

The following changes since commit e22ce8eb631bdc47a4a4ea7ecf4e4ba499db4f93: Linux 5.14-rc7 (2021-08-22 14:24:56 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.15 for you to fetch changes up to 38b7673000949ca784fcb8a9feb70d2a802befa6: Merge remote-tracking branch 'asoc/for-5.15' into asoc-linus (2021-08-30 12:30:33 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.15 Quite a quiet release this time, mostly a combination of cleanups and a good set of new drivers. - Lots of cleanups and improvements to the Intel drivers, including some new systems support. - New support for AMD Vangoh, CUI CMM-4030D-261, Mediatek Mt8195, Renesas RZ/G2L Mediatek Mt8195, RealTek RT101P, Renesas RZ/G2L,, Rockchip RK3568 S/PDIF. ---------------------------------------------------------------- Aakash Hemadri (4): ASoC: tegra30: ahub: Use of_device_get_match_data ASoC: tegra30: i2s: Use of_device_get_match_data ASoC: tegra30: ahub: Fix incorrect usage of of_device_get_match_data ASoC: tegra30: i2s: Fix incorrect usage of of_device_get_match_data Angelo Dureghello (3): ASoC: ics43432: add CMM-4030D-261 support dt-bindings: add compatible vendor prefix for CUI Devices ASoC: ics43432: add compatible for CUI Devices Bard Liao (8): ASoC: Intel: sof_sdw: extends SOF_RT711_JDSRC to 4 bits ASoC: Intel: sof_sdw: include rt711.h for RT711 JD mode soundwire: move intel sdw register definitions to sdw_intel.h ASoC: SOF: intel: add sdw_shim/alh_base to sof_intel_dsp_desc ASoC: SOF: intel: hda: remove HDA_DSP_REG_SNDW_WAKE_STS definition ASoC: SOF: intel: move sof_intel_dsp_desc() forward ASoC: SOF: intel: add snd_sof_dsp_check_sdw_irq ops soundwire: intel: introduce shim and alh base Biju Das (8): ASoC: dt-bindings: Document RZ/G2L bindings ASoC: dt-bindings: sound: renesas,rz-ssi: Document DMA support ASoC: sh: Add RZ/G2L SSIF-2 driver ASoC: dt-bindings: renesas,rz-ssi: Update slave dma channel configuration parameter ASoC: sh: rz-ssi: Add SSI DMAC support ASoC: sh: rz-ssi: Fix dereference of noderef expression warning ASoC: sh: rz-ssi: Fix wrong operator used issue ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function Brent Lu (6): ASoC: SOF: add a helper to get topology configured bclk ASoC: Intel: sof_cs42l42: use helper function to get bclk frequency ASoC: Intel: sof_cs42l42: support arbitrary DAI link sequence ASoC: Intel: maxim-common: support max98360a ASoC: Intel: sof_cs42l42: add support for jsl_cs4242_mx98360a ASoC: Intel: sof_rt5682: code refactor for max98360a Cezary Rojewski (5): ASoC: Intel: kbl_da7219_max98927: Fix format selection for max98373 ASoC: Intel: Skylake: Leave data as is when invoking TLV IPCs ASoC: Intel: Skylake: Fix module resource and format selection ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER ASoC: Intel: Skylake: Select first entry for singular pipe config arrays Changcheng Deng (1): ASoC: SOF: intel: remove duplicate include Charles Keepax (2): ASoC: wm_adsp: Remove pointless string comparison ASoC: wm_adsp: Put debugfs_remove_recursive back in Christophe JAILLET (3): ASoC: wcd9335: Fix a double irq free in the remove function ASoC: wcd9335: Fix a memory leak in the error handling path of the probe function ASoC: wcd9335: Disable irq on slave ports in the remove function Codrin Ciubotariu (1): ASoC: codecs: ad193x: add support for 96kHz and 192kHz playback rates Colin Ian King (7): ASoC: codecs: lpass-rx-macro: clean up for-loop indentation in switch statement ASoC: Intel: Fix spelling contraction "cant" -> "can't" ASoC: rsnd: make some arrays static const, makes object smaller ASoC: tlv320aic32x4: make array clocks static, makes object smaller ASoC: rt5514: make array div static const, makes object smaller ASoC: uniphier: make arrays mul and div static const, makes object smaller ASoC: mediatek: mt8195: Fix spelling mistake "bitwiedh" -> "bitwidth" Curtis Malainey (1): ASoC: Intel: Fix platform ID matching Dan Carpenter (2): ASoC: ti: delete some dead code in omap_abe_probe() ASoC: amd: fix an IS_ERR() vs NULL bug in probe Derek Fang (2): ASoC: rt5682: Adjust headset volume button threshold again ASoC: rt5682: Fix the vol+ button detection issue Dmitry Osipenko (1): ASoC: rt5640: Silence warning message about missing interrupt Douglas Anderson (2): ASoC: rt5682: Properly turn off regulators if wrong device ID ASoC: rt5682: Remove unused variable in rt5682_i2c_remove() Gongjun Song (1): ASoC: Intel: soc-acpi: add support for SoundWire of TGL-H-RVP Gu Shengxian (2): ASoC: atmel: fix spelling mistakes ASoC: amd: fix spelling mistakes Gustaw Lewandowski (2): ASoC: Intel: Skylake: Fix passing loadable flag for module ASoC: Intel: Skylake: Simplify m_state for loadable modules Hans de Goede (14): ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output ASoC: Intel: bytcr_rt5640: Add line-out support ASoC: Intel: bytcr_rt5640: Add a byt_rt5640_get_codec_dai() helper ASoC: Intel: bytcr_rt5640: Add support for a second headphones output ASoC: Intel: bytcr_rt5640: Add support for a second headset mic input ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 quirk ASoC: Intel: bytcr_rt5640: Use cfg-lineout:2 in the components string ASoC: rt5640: Move rt5640_disable_jack_detect() up in the rt5640.c file ASoC: rt5640: Delay requesting IRQ until the machine-drv calls set_jack ASoC: rt5640: Add optional hp_det_gpio parameter to rt5640_detect_headset() ASoC: rt5640: Add rt5640_set_ovcd_params() helper ASoC: Intel: bytct_rt5640: Add a separate "Headset Mic 2" DAPM pin for the mic on the 2nd jack ASoC: Intel: bytcr_rt5640: Add support for HP Elite Pad 1000G2 jack-detect ASoC: Intel: bytcr_rt5640: Mark hp_elitepad_1000g2_jack?_check functions static Heikki Krogerus (1): ASoC: Intel: boards: use software node API in Atom boards Jack Yu (4): ASoC: rt1015: Remove unnecessary flush work on rt1015 driver ASoC: rt1015p: add new acpi id and comapatible id ASoC: dt-bindings: rt1015p: add new compatible id ASoC: dt-bindings: rt1015p: fix syntax error in dts-binding document Jiri Slaby (2): ASoC: cx20442: tty_ldisc_ops::write_wakeup is optional ASoC: v253_init: eliminate pointer to string Kareem Shaik (1): ASoC: Intel: Skylake: Support multiple format configs Krzysztof Kozlowski (1): ASoC: rt1015p: correct indentation Kuninori Morimoto (44): ASoC: soc-pcm: cleanup cppcheck warning at soc_pcm_apply_msb() ASoC: soc-pcm: cleanup cppcheck warning at soc_pcm_components_close() ASoC: soc-pcm: cleanup cppcheck warning at soc_get_playback_capture() ASoC: soc-pcm: cleanup cppcheck warning at dpcm_be_is_active() ASoC: soc-pcm: cleanup cppcheck warning at dpcm_runtime_setup_be_chan() ASoC: soc-core: cleanup cppcheck warning at snd_soc_set_dmi_name() ASoC: soc-core: cleanup cppcheck warning at snd_soc_get_dai_name() ASoC: soc-core: cleanup cppcheck warning at snd_soc_daifmt_parse_format() ASoC: soc-core: cleanup cppcheck warning at snd_soc_unregister_component() ASoC: soc-core: cleanup cppcheck warning at snd_soc_add_controls() ASoC: soc-core: cleanup cppcheck warning at snd_soc_of_parse_audio_simple_widgets() ASoC: soc-core: cleanup cppcheck warning at snd_soc_of_parse_audio_routing() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_process_headers() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_dai_elems_load() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_dapm_widget_elems_load() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_kcontrol_elems_load() ASoC: soc-topology: cleanup cppcheck warning at snd_soc_find_dai_link() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_put_volsw_sx() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_limit_volume() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_get_xr_sx() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_put_xr_sx() ASoC: soc-dapm: cleanup cppcheck warning at dapm_wcache_lookup() ASoC: soc-dapm: cleanup cppcheck warning at dapm_connect_mux() ASoC: soc-dapm: cleanup cppcheck warning at dapm_set_mixer_path_status() ASoC: soc-dapm: cleanup cppcheck warning at dapm_new_pga() ASoC: soc-dapm: cleanup cppcheck warning at dapm_new_dai_link() ASoC: soc-dapm: cleanup cppcheck warning at dapm_seq_check_event() ASoC: soc-dapm: cleanup cppcheck warning at dapm_seq_run() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_del_route() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_add_routes() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_weak_routes() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_new_controls() ASoC: soc-dapm: cleanup cppcheck warning at soc_dapm_dai_stream_event() ASoC: soc-generic-dmaengine-pcm: cleanup cppcheck warning at dmaengine_pcm_hw_params() ASoC: soc-generic-dmaengine-pcm: cleanup cppcheck warning at dmaengine_pcm_new() ASoC: soc-generic-dmaengine-pcm: cleanup cppcheck warning at dmaengine_copy_user() ASoC: soc-dai: cleanup cppcheck warning at snd_soc_dai_link_set_capabilities() ASoC: soc-dai: cleanup cppcheck warning at snd_soc_pcm_dai_new() ASoC: soc-jack: cleanup cppcheck warning at snd_soc_jack_report() ASoC: soc-jack: cleanup cppcheck warning for CONFIG_GPIOLIB ASoC: soc-component: cleanup cppcheck warning at snd_soc_pcm_component_pm_runtime_get() ASoC: soc-ac97: cleanup cppcheck warning ASoC: rsnd: core: make some arrays static const, makes object smaller ASoC: rsnd: adg: clearly handle clock error / NULL case Marek Vasut (1): ASoC: tlv320aic32x4: Fix TAS2505/TAS2521 channel count Mario Limonciello (2): ASoC: amd: Don't show messages about deferred probing by default ASoC: amd: Use dev_probe_err helper Mark Brown (31): Merge existing fixes from asoc/for-5.14 Merge series "ASoC: Intel: machine driver corrections" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Fix unbalanced pm_runtime_enable in error handle" from Zhang Qilong <zhangqilong3(a)huawei.com>: Merge series "ASoC: ti: davinci-mcasp: Fix the DIT mode and OMAP4 support" from Peter Ujfalusi <peter.ujfalusi(a)gmail.com>: Merge series "soundwire/ASoC: add mockup codec support" from Bard Liao <yung-chuan.liao(a)linux.intel.com>: Merge series "ASoC: soc-pcm: cleanup cppcheck warning" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: soc-core: cleanup cppcheck warning" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "Add RZ/G2L Sound support" from Biju Das <biju.das.jz(a)bp.renesas.com>: Merge series "ASoC: soc-topology: cleanup cppcheck warning" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 audio routing" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "soundwire/ASoC: abstract platform-dependent bases" from Bard Liao <yung-chuan.liao(a)linux.intel.com>: Merge series "ASoC: SOF/Intel: machine driver updates" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: soc-ops: cleanup cppcheck warning" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: soc-dapm: cleanup cppcheck warning" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: codecs: cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge some cs42l42 patches into asoc-5.15 Merge series "ASoC: SOF: Intel: DMI L1 power optimization for HDaudio platforms" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Add RZ/G2L Sound support" from Biju Das <biju.das.jz(a)bp.renesas.com>: Merge series "ASoC: Intel: boards: use software node API" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: soc-xxx: cleanup cppcheck warning" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: tegra30: Fix use of of_device_get_match_data" from Aakash Hemadri <aakashhemadri123(a)gmail.com>: Merge series "ASoC: Intel/rt5640: Add support for HP Elite Pad 1000G2 jack-detect" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: Intel: Skylake: Fix and support complex" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Merge series "ASoC: mediatek: Add support for MT8195 SoC" from Trevor Wu <trevor.wu(a)mediatek.com>: Merge branch 'for-5.14' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-5.15 Merge series "Patches to update for rockchip spdif" from Sugar Zhang <sugar.zhang(a)rock-chips.com>: Merge series "Patches to update for rockchip i2s" from Sugar Zhang <sugar.zhang(a)rock-chips.com>: Merge series "ASoC: wcd9335: Firx some resources leak in the probe and remove function" from Christophe JAILLET <christophe.jaillet(a)wanadoo.fr>: ASoC: Revert PCM trigger changes Merge remote-tracking branch 'asoc/for-5.14' into asoc-linus Merge remote-tracking branch 'asoc/for-5.15' into asoc-linus Nobuhiro Iwamatsu (1): ASoC: mediatek: mt6359: convert to use module_platform_driver Pawel Harlozinski (1): ASoC: Intel: Skylake: Properly configure modules with generic extension Peter Ujfalusi (5): ASoC: ti: davinci-mcasp: Fix DIT mode support ASoC: dt-bindings: davinci-mcasp: Add compatible string for OMAP4 ASoC: ti: davinci-mcasp: Add support for the OMAP4 version of McASP ASoC: ti: j721e-evm: Convert the audio domain IDs to enum ASoC: Intel: bytcr_rt5640: Make rt5640_jack_gpio/rt5640_jack2_gpio static Pierre-Louis Bossart (24): ASoC: codecs: add SoundWire mockup device support ASoC: soc-acpi: cnl: add table for SoundWire mockup devices ASoC: soc-acpi: tgl: add table for SoundWire mockup devices ASoC: Intel: boards: sof_sdw: add SoundWire mockup codecs for tests ASOC: Intel: sof_sdw: add quirk for Intel 'Bishop County' NUC M15 ASoC: Intel: sof_sdw: add quirk for Dell XPS 9710 ASoC: Intel: update sof_pcm512x quirks ASoC: Intel: sof_sdw_max98373: remove useless inits ASoC: max98090: remove duplicate status reads and useless assignmment ASoC: mt6359-accdet.c: remove useless assignments ASoC: wcd938x: simplify return value ASoC: SOF: Intel: Kconfig: clarify DMI L1 option description ASoC: SOF: Intel: hda-stream: remove always true condition ASoC: SOF: Intel: simplify logic for DMI_L1 handling ASoC: SOF: Intel: make DMI L1 selection more robust ASoC: Intel: boards: harden codec property handling ASoC: Intel: boards: handle errors with acpi_dev_get_first_match_dev() ASoC: Intel: boards: get codec device with ACPI instead of bus search ASoC: Intel: sof_sdw: pass card information to init/exit functions ASoC: Intel: sof_sdw_rt711*: keep codec device reference until remove ASoC: Intel: use software node API in SoundWire machines ASoC: Intel: remove device_properties for Atom boards ASoC: soc-pcm: protect BE dailink state changes in trigger ASoC: soc-pcm: test refcount before triggering Piotr Maziarz (1): ASoC: Intel: Skylake: Select proper format for NHLT blob Randy Dunlap (1): ASoC: atmel: ATMEL drivers don't need HAS_DMA Richard Fitzgerald (4): ASoC: cs42l42: Add PLL configuration for 44.1kHz/16-bit ASoC: cs42l42: Validate dai_set_sysclk() frequency ASoC: cs42l42: Assume 24-bit samples are in 32-bit slots ASoC: cs42l42: Update module authors Rikard Falkeborn (1): ASoC: samsung: Constify static snd_soc_ops Robin Murphy (1): ASoC: dwc: Get IRQ optionally Samuel Holland (1): ASoC: simple-card-utils: Avoid over-allocating DLCs Shengjiu Wang (2): ASoC: fsl_rpmsg: Check -EPROBE_DEFER for getting clocks ASoC: imx-rpmsg: change dev_err to dev_err_probe for -EPROBE_DEFER Shuming Fan (1): ASoC: rt5682: enable SAR ADC power saving mode during suspend Srinivas Kandagatla (4): ASoC: codecs: wcd938x: remove unused port-map reference ASoC: codecs: wcd938x: add Multi Button Headset Control support ASoC: qdsp6: q6asm: fix cppcheck warnings for unnecessary initialization ASoC: qdsp6: q6adm: fix cppcheck warnings for unnecessary initialization Steve Lee (1): ASoC: max98390: Add support change dsm param name Sugar Zhang (13): ASoC: rockchip: spdif: Mark SPDIF_SMPDR as volatile ASoC: rockchip: spdif: Fix some coding style ASoC: rockchip: spdif: Add support for rk3568 spdif ASoC: dt-bindings: rockchip: Add compatible for rk3568 spdif ASoC: rockchip: i2s: Add support for set bclk ratio ASoC: rockchip: i2s: Fixup clk div error ASoC: rockchip: i2s: Improve dma data transfer efficiency ASoC: rockchip: i2s: Fix regmap_ops hang ASoC: dt-bindings: rockchip: Document reset property for i2s ASoC: rockchip: i2s: Make playback/capture optional ASoC: rockchip: i2s: Add compatible for more SoCs ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs ASoC: rockchip: i2s: Add support for frame inversion Szymon Mielczarek (1): ASoC: Intel: Skylake: Support modules with generic extension Takashi Iwai (3): ASoC: amd: Drop superfluous mmap callbacks ASoC: intel: skylake: Drop superfluous mmap callback ASoC: amd: vangogh: Drop superfluous mmap callback Tang Bin (2): ASoC: fsl_xcvr: Omit superfluous error message in fsl_xcvr_probe() ASoC: bcm: cygnus-pcm: Fix unused assignment about 'rc' Trevor Wu (11): ASoC: mediatek: mt8195: update mediatek common driver ASoC: mediatek: mt8195: support audsys clock control ASoC: mediatek: mt8195: support etdm in platform driver ASoC: mediatek: mt8195: support adda in platform driver ASoC: mediatek: mt8195: support pcm in platform driver ASoC: mediatek: mt8195: add platform driver dt-bindings: mediatek: mt8195: add audio afe document ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 ASoC: mediatek: mt8195: add DPTX audio support ASoC: mediatek: mt8195: add HDMITX audio support dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682 document Tzung-Bi Shih (1): ASoC: rt1015: remove possible unused variable `bclk_ms' Vijendar Mukunda (12): ASoC: amd: add Vangogh ACP5x IP register header ASoC: amd: add Vangogh ACP PCI driver ASoc: amd: add acp5x init/de-init functions ASoC: amd: create acp5x platform devices ASoC: amd: add ACP5x PCM platform driver ASoC: amd: irq handler changes for ACP5x PCM dma driver ASoC: amd: add ACP5x pcm dma driver ops ASoC: amd: add vangogh i2s controller driver ASoC: amd: add vangogh i2s dai driver ops ASoC: amd: add vangogh pci driver pm ops ASoC: amd: add vangogh i2s dma driver pm ops ASoC: amd: enable vangogh acp5x driver build Vincent Knecht (1): ASoC: qcom: apq8016_sbc: Add SEC_MI2S support Wolfram Sang (1): ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination Xiaotan Luo (1): ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B Zhang Qilong (2): ASoC: mediatek: mt8192:Fix Unbalanced pm_runtime_enable in mt8192_afe_pcm_dev_probe ASoC: mediatek: mt8183: Fix Unbalanced pm_runtime_enable in mt8183_afe_pcm_dev_probe jairaj arava (2): ASoC: Intel: sof_sdw: update quirk for jack detection in ADL RVP ASoC: SOF: Intel: Use DMI string to search for adl_mx98373_rt5682 variant kernel test robot (1): ASoC: codecs: wcd938x: fix returnvar.cocci warnings .../bindings/sound/davinci-mcasp-audio.txt | 1 + .../devicetree/bindings/sound/ics43432.txt | 6 +- .../devicetree/bindings/sound/mt8195-afe-pcm.yaml | 184 ++ .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 47 + .../devicetree/bindings/sound/realtek,rt1015p.yaml | 4 +- .../devicetree/bindings/sound/renesas,rz-ssi.yaml | 118 + .../devicetree/bindings/sound/rockchip-i2s.yaml | 11 + .../devicetree/bindings/sound/rockchip-spdif.yaml | 1 + .../devicetree/bindings/vendor-prefixes.yaml | 2 + drivers/soundwire/intel.c | 74 - drivers/soundwire/intel_init.c | 14 +- include/linux/platform_data/davinci_asp.h | 1 + include/linux/soundwire/sdw_intel.h | 87 + include/sound/sof.h | 1 + include/uapi/sound/snd_sst_tokens.h | 6 +- sound/soc/amd/Kconfig | 9 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp-da7219-max98357a.c | 12 +- sound/soc/amd/acp-pcm-dma.c | 20 +- sound/soc/amd/acp3x-rt5682-max9836.c | 14 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 10 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 - sound/soc/amd/renoir/acp3x-rn.c | 7 +- sound/soc/amd/vangogh/Makefile | 9 + sound/soc/amd/vangogh/acp5x-i2s.c | 427 +++ sound/soc/amd/vangogh/acp5x-pcm-dma.c | 517 +++ sound/soc/amd/vangogh/acp5x.h | 193 ++ sound/soc/amd/vangogh/pci-acp5x.c | 326 ++ sound/soc/amd/vangogh/vg_chip_offset_byte.h | 337 ++ sound/soc/atmel/Kconfig | 1 - sound/soc/atmel/mchp-spdifrx.c | 6 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/Kconfig | 21 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 30 +- sound/soc/codecs/ad193x.h | 4 + sound/soc/codecs/cs42l42.c | 28 +- sound/soc/codecs/cx20442.c | 8 +- sound/soc/codecs/ics43432.c | 4 +- sound/soc/codecs/lpass-rx-macro.c | 65 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98390.c | 26 +- sound/soc/codecs/max98390.h | 1 + sound/soc/codecs/mt6359-accdet.c | 25 +- sound/soc/codecs/rt1015.c | 46 +- sound/soc/codecs/rt1015.h | 2 - sound/soc/codecs/rt1015p.c | 2 + sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5640.c | 133 +- sound/soc/codecs/rt5640.h | 6 + sound/soc/codecs/rt5682-i2c.c | 15 +- sound/soc/codecs/rt5682.c | 57 +- sound/soc/codecs/sdw-mockup.c | 310 ++ sound/soc/codecs/tlv320aic32x4.c | 10 +- sound/soc/codecs/wcd9335.c | 23 +- sound/soc/codecs/wcd938x.c | 854 ++++- sound/soc/codecs/wcd938x.h | 1 - sound/soc/codecs/wm_adsp.c | 9 +- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_rpmsg.c | 20 +- sound/soc/fsl/fsl_xcvr.c | 4 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/intel/atom/sst/sst_ipc.c | 2 +- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bytcht_es8316.c | 31 +- sound/soc/intel/boards/bytcr_rt5640.c | 328 +- sound/soc/intel/boards/bytcr_rt5651.c | 63 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 55 +- sound/soc/intel/boards/sof_cs42l42.c | 361 ++- sound/soc/intel/boards/sof_maxim_common.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 4 +- sound/soc/intel/boards/sof_pcm512x.c | 13 +- sound/soc/intel/boards/sof_rt5682.c | 66 +- sound/soc/intel/boards/sof_sdw.c | 105 +- sound/soc/intel/boards/sof_sdw_common.h | 64 +- sound/soc/intel/boards/sof_sdw_max98373.c | 7 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1316.c | 3 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 3 +- sound/soc/intel/boards/sof_sdw_rt700.c | 3 +- sound/soc/intel/boards/sof_sdw_rt711.c | 51 +- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 52 +- sound/soc/intel/boards/sof_sdw_rt715.c | 3 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 3 +- sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 15 + sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- .../intel/common/soc-acpi-intel-sdw-mockup-match.c | 166 + .../intel/common/soc-acpi-intel-sdw-mockup-match.h | 17 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 38 + sound/soc/intel/skylake/skl-messages.c | 155 +- sound/soc/intel/skylake/skl-pcm.c | 33 +- sound/soc/intel/skylake/skl-topology.c | 155 +- sound/soc/intel/skylake/skl-topology.h | 26 +- sound/soc/mediatek/Kconfig | 24 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 22 +- sound/soc/mediatek/common/mtk-base-afe.h | 10 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 43 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 27 +- sound/soc/mediatek/mt8195/Makefile | 15 + sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 441 +++ sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 109 + sound/soc/mediatek/mt8195/mt8195-afe-common.h | 158 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3281 ++++++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 214 ++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 15 + sound/soc/mediatek/mt8195/mt8195-audsys-clkid.h | 93 + sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 830 +++++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2639 ++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 389 +++ .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 1087 +++++++ sound/soc/mediatek/mt8195/mt8195-reg.h | 2796 +++++++++++++++++ sound/soc/qcom/apq8016_sbc.c | 17 + sound/soc/qcom/qdsp6/q6adm.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 212 +- sound/soc/rockchip/rockchip_i2s.h | 10 +- sound/soc/rockchip/rockchip_spdif.c | 38 +- sound/soc/samsung/aries_wm8994.c | 2 +- sound/soc/samsung/arndale.c | 4 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 6 +- sound/soc/samsung/tobermory.c | 2 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rcar/adg.c | 12 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/ssi.c | 6 +- sound/soc/sh/rz-ssi.c | 1074 +++++++ sound/soc/soc-ac97.c | 14 +- sound/soc/soc-component.c | 4 +- sound/soc/soc-core.c | 36 +- sound/soc/soc-dai.c | 18 +- sound/soc/soc-dapm.c | 65 +- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/soc-jack.c | 15 +- sound/soc/soc-ops.c | 21 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 23 +- sound/soc/sof/intel/Kconfig | 10 - sound/soc/sof/intel/cnl.c | 6 + sound/soc/sof/intel/hda-pcm.c | 16 +- sound/soc/sof/intel/hda-stream.c | 11 +- sound/soc/sof/intel/hda.c | 39 +- sound/soc/sof/intel/hda.h | 8 +- sound/soc/sof/intel/icl.c | 3 + sound/soc/sof/intel/pci-tng.c | 1 - sound/soc/sof/intel/shim.h | 3 + sound/soc/sof/intel/tgl.c | 12 + sound/soc/sof/sof-audio.c | 42 +- sound/soc/sof/sof-pci-dev.c | 9 + sound/soc/tegra/tegra30_ahub.c | 6 +- sound/soc/tegra/tegra30_i2s.c | 8 +- sound/soc/ti/Kconfig | 1 + sound/soc/ti/davinci-mcasp.c | 176 +- sound/soc/ti/j721e-evm.c | 30 +- sound/soc/ti/omap-abe-twl6040.c | 5 - sound/soc/uniphier/aio-cpu.c | 4 +- 173 files changed, 18989 insertions(+), 1285 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1019-rt5682.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml create mode 100644 sound/soc/amd/vangogh/Makefile create mode 100644 sound/soc/amd/vangogh/acp5x-i2s.c create mode 100644 sound/soc/amd/vangogh/acp5x-pcm-dma.c create mode 100644 sound/soc/amd/vangogh/acp5x.h create mode 100644 sound/soc/amd/vangogh/pci-acp5x.c create mode 100644 sound/soc/amd/vangogh/vg_chip_offset_byte.h create mode 100644 sound/soc/codecs/sdw-mockup.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.h create mode 100644 sound/soc/mediatek/mt8195/Makefile create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-adda.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h create mode 100644 sound/soc/sh/rz-ssi.c

2 1

[PATCH AUTOSEL 5.13 01/12] ASoC: rt5682: Adjust headset volume button threshold
by Sasha Levin 30 Aug '21

30 Aug '21

From: Derek Fang <derek.fang(a)realtek.com> [ Upstream commit 6d20bf7c020f417fdef1810a22da17c126603472 ] Adjust the threshold of headset button volume+ to fix the wrong button detection issue with some brand headsets. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> Link: https://lore.kernel.org/r/20210721133121.12333-1-derek.fang@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 abcd6f483788..51ecaa2abcd1 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.30.2

3 5

[PATCH 1/2] dt-bindings: panel: ili9341: correct indentation
by Krzysztof Kozlowski 30 Aug '21

30 Aug '21

Correct indentation warning: ilitek,ili9341.yaml:25:9: [warning] wrong indentation: expected 10 but found 8 (indentation) Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> --- .../devicetree/bindings/display/panel/ilitek,ili9341.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/display/panel/ilitek,ili9341.yaml b/Documentation/devicetree/bindings/display/panel/ilitek,ili9341.yaml index 2ed010f91e2d..20ce88ab4b3a 100644 --- a/Documentation/devicetree/bindings/display/panel/ilitek,ili9341.yaml +++ b/Documentation/devicetree/bindings/display/panel/ilitek,ili9341.yaml @@ -22,7 +22,7 @@ properties: items: - enum: # ili9341 240*320 Color on stm32f429-disco board - - st,sf-tc240t-9370-t + - st,sf-tc240t-9370-t - const: ilitek,ili9341 reg: true -- 2.30.2

5 9

[RFC PATCH 0/2] ASoC: soc-pcm: fix trigger race conditions with shared BE
by Pierre-Louis Bossart 30 Aug '21

30 Aug '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 using the same spinlock used in other parts of soc-dpcm.c, and the use of a refcount to decide when to trigger the BE. With the two patches I am able to run our entire validation suite without any issues with this new 'deep buffer' topology. 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. 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) it's not clear if we need the full-blown spin_lock_irq_save() version or just the regular spin_lock()? 3) is this universal? this might break platforms with other types of 'POST' or 'BESPOKE' trigger. 4) is this ok to proceed with such an incremental fix, or do we have to revisit (possibly rewrite) the entire FE-BE interaction? I chose to send this patchset as an RFC to gather 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. Note that these DPCM issues will add more complexity to our SOF distribution. We've never hit a case where even recent kernels do not support a minor topology change that isn't related to an ABI change. We might have to introduce the deep buffer PCM device in a new topology file, add a kernel patch to make use of this new file and keep the old (w/o deep buffer) and new file (w/ deep buffer) for a very long time (not sure when we can assume that all users of SOF would transition to 5.15+...) Pierre-Louis Bossart (2): ASoC: soc-pcm: protect BE dailink state changes in trigger ASoC: soc-pcm: test refcount before triggering include/sound/soc-dpcm.h | 2 + sound/soc/soc-pcm.c | 151 +++++++++++++++++++++++++++++++++------ 2 files changed, 130 insertions(+), 23 deletions(-) -- 2.25.1

2 5

[bug report] ASoC: mediatek: implement mediatek common structure
by Dan Carpenter 30 Aug '21

30 Aug '21

Hello Garlic Tseng, The patch 283b612429a2: "ASoC: mediatek: implement mediatek common structure" from Jun 17, 2016, leads to the following Smatch static checker warning: sound/soc/mediatek/common/mtk-afe-fe-dai.c:364 mtk_afe_resume() error: we previously assumed 'afe->reg_back_up' could be null (see line 359) sound/soc/mediatek/common/mtk-afe-fe-dai.c 354 if (pm_runtime_status_suspended(dev) || !afe->suspended) 355 return 0; 356 357 afe->runtime_resume(dev); 358 359 if (!afe->reg_back_up) 360 dev_dbg(dev, "%s no reg_backup\n", __func__); Should this return? Also when ->reg_back_up is allocated there is no check for NULL so it seems like a very possible condition. 361 362 for (i = 0; i < afe->reg_back_up_list_num; i++) 363 mtk_regmap_write(regmap, afe->reg_back_up_list[i], --> 364 afe->reg_back_up[i]); 365 366 afe->suspended = false; 367 return 0; 368 } regards, dan carpenter

2 1

[PATCH] ASoC: rockchip: i2s: Fix concurrency between tx/rx
by Sugar Zhang 30 Aug '21

30 Aug '21

This patch adds lock to fix comcurrency between tx/rx to fix 'rockchip-i2s ff070000.i2s; fail to clear' Considering the situation; tx stream rx stream | | | disable enable | | reset After this patch: lock | tx stream | enable | unlock -------- --------- lock | rx stream | disable | reset | unlock Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- --- sound/soc/rockchip/rockchip_i2s.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index c7dc350..e4782ba 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -15,6 +15,7 @@ #include <linux/clk.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/spinlock.h> #include <sound/pcm_params.h> #include <sound/dmaengine_pcm.h> @@ -49,6 +50,7 @@ struct rk_i2s_dev { bool rx_start; bool is_master_mode; const struct rk_i2s_pins *pins; + spinlock_t lock; /* tx/rx lock */ }; static int i2s_runtime_suspend(struct device *dev) @@ -92,6 +94,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) unsigned int val = 0; int retry = 10; + spin_lock(&lock); if (on) { regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); @@ -132,6 +135,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } + spin_unlock(&lock); } static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) @@ -139,6 +143,7 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) unsigned int val = 0; int retry = 10; + spin_lock(&lock); if (on) { regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); @@ -179,6 +184,7 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } + spin_unlock(&lock); } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -589,6 +595,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; + spin_lock_init(&i2s->lock); i2s->dev = &pdev->dev; i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); -- 2.7.4

2 2

Re: linux-next: Tree for Aug 27 (sound/soc/mediatek: Kconfig)
by Randy Dunlap 30 Aug '21

30 Aug '21

On 8/27/21 4:24 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20210826: > > The kbuild tree still had its build failure for which I reverted a commit. > > The uml tree gained a conflict against the asm-generic tree. > > The net-next tree still had its build failure for which I applied a > patch and it also gained a semantic conflict against Linus' tree. > > The bpf-next tree lost its build failure. > > The tip tree gained conflicts against the arm64 tree. > > The ftrace tree gained a conflict against the uml tree. > > The vfio tree gained conflicts against the drm tree. > > The kspp tree still had its build failure so I used the version from > next-20210825. > > Non-merge commits (relative to Linus' tree): 9867 > 9528 files changed, 552507 insertions(+), 227054 deletions(-) > > ---------------------------------------------------------------------------- > > I have created today's linux-next tree at > git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git > (patches at http://www.kernel.org/pub/linux/kernel/next/ ). If you > are tracking the linux-next tree using git, you should not use "git pull" > to do so as that will try to merge the new linux-next release with the > old one. You should use "git fetch" and checkout or reset to the new > master. > > You can see which trees have been included by looking in the Next/Trees > file in the source. There are also quilt-import.log and merge.log > files in the Next directory. Between each merge, the tree was built > with a ppc64_defconfig for powerpc, an allmodconfig for x86_64, a > multi_v7_defconfig for arm and a native build of tools/perf. After > the final fixups (if any), I do an x86_64 modules_install followed by > builds for x86_64 allnoconfig, powerpc allnoconfig (32 and 64 bit), > ppc44x_defconfig, allyesconfig and pseries_le_defconfig and i386, sparc > and sparc64 defconfig and htmldocs. And finally, a simple boot test > of the powerpc pseries_le_defconfig kernel in qemu (with and without > kvm enabled). > > Below is a summary of the state of the merge. > > I am currently merging 334 trees (counting Linus' and 90 trees of bug > fix patches pending for the current merge release). > > Stats about the size of the tree over time can be seen at > http://neuling.org/linux-next-size.html . > > Status of my local build tests will be at > http://kisskb.ellerman.id.au/linux-next . If maintainers want to give > advice about cross compilers/configs that work, we are always open to add > more builds. > > Thanks to Randy Dunlap for doing many randconfig builds. And to Paul > Gortmaker for triage and bug fixes. > on i386: WARNING: unmet direct dependencies detected for SND_SOC_MT6359 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MTK_PMIC_WRAP [=n] Selected by [y]: - SND_SOC_MT8195_MT6359_RT1019_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8195 [=y] caused by 40d605df0a7b: ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

2 1

[PATCH 0/2] Add reset-gpios handling for max98927
by Alejandro Tafalla 29 Aug '21

29 Aug '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) require hardware-resetting the codec by driving a reset-gpio. This series add support for it through an optional reset-gpios property. Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 16 ++++++++++++++++ sound/soc/codecs/max98927.h | 1 + 3 files changed, 20 insertions(+) -- 2.32.0

1 2

Question: different install prefixes for libalsa and alsa-plugins?
by GitHub issues - opened 29 Aug '21

29 Aug '21

alsa-project/alsa-lib issue #175 was opened from daravi: Hello, I use Conan to manage my dependencies and wanted to add a recipe for `alsa-plugins`. However it seems that `ALSA_CONFIG_DIR` can only be set to a single path: https://github.com/conan-io/conan-center-index/pull/7098#discussion_r697920… So I was wondering if there is any other way I could have alsa and plugins installed in separate directories? (so that they can each have their own config directory) Issue URL : https://github.com/alsa-project/alsa-lib/issues/175 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: Add lowlatency module option
by Takashi Iwai 29 Aug '21

29 Aug '21

For making user to switch back to the old playback mode, this patch adds a new module option 'lowlatency' to snd-usb-audio driver. When user face a regression due to the recent low-latency playback support, they can test easily by passing lowlatency=0 option without rebuilding the kernel. Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 4 ++++ sound/usb/pcm.c | 3 ++- sound/usb/usbaudio.h | 1 + 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index cf8f3953f78f..fd570a42f043 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -68,6 +68,7 @@ static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; +static bool lowlatency = true; static char *quirk_alias[SNDRV_CARDS]; static char *delayed_register[SNDRV_CARDS]; static bool implicit_fb[SNDRV_CARDS]; @@ -93,6 +94,8 @@ MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); +module_param(lowlatency, bool, 0444); +MODULE_PARM_DESC(lowlatency, "Enable low latency playback (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); module_param_array(delayed_register, charp, NULL, 0444); @@ -623,6 +626,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, chip->setup = device_setup[idx]; chip->generic_implicit_fb = implicit_fb[idx]; chip->autoclock = autoclock; + chip->lowlatency = lowlatency; atomic_set(&chip->active, 1); /* avoid autopm during probing */ atomic_set(&chip->usage_count, 0); atomic_set(&chip->shutdown, 0); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f5cbf61ac366..5dc9266180e3 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -617,7 +617,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* check whether early start is needed for playback stream */ subs->early_playback_start = subs->direction == SNDRV_PCM_STREAM_PLAYBACK && - subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes; + (!chip->lowlatency || + (subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes)); if (subs->early_playback_start) ret = start_endpoints(subs); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 94261d19cceb..167834133b9b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -55,6 +55,7 @@ struct snd_usb_audio { bool generic_implicit_fb; /* from the 'implicit_fb' module param */ bool autoclock; /* from the 'autoclock' module param */ + bool lowlatency; /* from the 'lowlatency' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ struct media_device *media_dev; struct media_intf_devnode *ctl_intf_media_devnode; -- 2.26.2

1 0

Static build generates broken symlinks
by GitHub issues - opened 28 Aug '21

28 Aug '21

alsa-project/alsa-plugins issue #34 was opened from daravi: Hello, This snippet seems to generate some symlinks: ``` install-exec-hook: rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_speexrate_*.so $(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_speexrate_best.so $(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_speexrate_medium.so ``` But it assumes we are doing a shared build. Which results in broken symlinks in a static build. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/34 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/2] ctl/rawmidi/hwdep: fix trivial bugs
by GitHub pull_request - opened 28 Aug '21

28 Aug '21

alsa-project/alsa-gobject pull request #62 was opened from takaswie: Current implementation of libalsactl0, libalsarawmidi0, and libalsahwdep0 includes trivial bugs such as usage of uninitialized local variable. This patchset fixes the bugs. ``` Takashi Sakamoto (2): ctl/rawmidi/hwdep: fix compiler warning due to uninitialized variables seq: use safer way to copy strings src/ctl/query.c | 4 ++-- src/hwdep/query.c | 4 ++-- src/rawmidi/query.c | 4 ++-- src/seq/client-info.c | 2 +- src/seq/port-info.c | 2 +- src/seq/query.c | 2 +- src/seq/queue-info.c | 2 +- 7 files changed, 10 insertions(+), 10 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/62 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 1/2] ALSA: hda/cs8409: Ensure Type Detection is only run on startup when necessary
by Vitaly Rodionov 27 Aug '21

27 Aug '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Type Detection should only be run after init and when the controls have been built. There is no need to run it multiple times. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 34 ++++++++++++++++++++++++++++------ sound/pci/hda/patch_cs8409.h | 2 ++ 2 files changed, 30 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 9db16b6292f4..805441b7bf86 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -784,6 +784,8 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) cs8409_i2c_write(cs42l42, 0x1102, 0x9C); cs42l42->suspended = 1; cs42l42->last_page = 0; + cs42l42->hp_jack_in = 0; + cs42l42->mic_jack_in = 0; /* Put CS42L42 into Reset */ gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0); @@ -852,6 +854,8 @@ static int cs8409_cs42l42_suspend(struct hda_codec *codec) struct cs8409_spec *spec = codec->spec; int i; + spec->init_done = 0; + cs8409_enable_ur(codec, 0); for (i = 0; i < spec->num_scodecs; i++) @@ -1025,14 +1029,21 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, break; case HDA_FIXUP_ACT_INIT: cs8409_cs42l42_hw_init(codec); - fallthrough; + spec->init_done = 1; + if (spec->init_done && spec->build_ctrl_done + && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + break; case HDA_FIXUP_ACT_BUILD: + spec->build_ctrl_done = 1; /* Run jack auto detect first time on boot * after controls have been added, to check if jack has * been already plugged in. * Run immediately after init. */ - cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + if (spec->init_done && spec->build_ctrl_done + && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); break; default: break; @@ -1223,16 +1234,27 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac break; case HDA_FIXUP_ACT_INIT: dolphin_hw_init(codec); - fallthrough; + spec->init_done = 1; + if (spec->init_done && spec->build_ctrl_done) { + for (i = 0; i < spec->num_scodecs; i++) { + if (!spec->scodecs[i]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[i]); + } + } + break; case HDA_FIXUP_ACT_BUILD: + spec->build_ctrl_done = 1; /* Run jack auto detect first time on boot * after controls have been added, to check if jack has * been already plugged in. * Run immediately after init. */ - for (i = 0; i < spec->num_scodecs; i++) - cs42l42_run_jack_detect(spec->scodecs[i]); - + if (spec->init_done && spec->build_ctrl_done) { + for (i = 0; i < spec->num_scodecs; i++) { + if (!spec->scodecs[i]->hp_jack_in) + cs42l42_run_jack_detect(spec->scodecs[i]); + } + } break; default: break; diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index 207315ad5bf6..ade2b838590c 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -333,6 +333,8 @@ struct cs8409_spec { unsigned int playback_started:1; unsigned int capture_started:1; + unsigned int init_done:1; + unsigned int build_ctrl_done:1; /* verb exec op override */ int (*exec_verb)(struct hdac_device *dev, unsigned int cmd, unsigned int flags, -- 2.25.1

2 3

[PATCH v2] ALSA: pcm: fix divide error in snd_pcm_lib_ioctl
by Zubin Mithra 27 Aug '21

27 Aug '21

Syzkaller reported a divide error in snd_pcm_lib_ioctl. fifo_size is of type snd_pcm_uframes_t(unsigned long). If frame_size is 0x100000000, the error occurs. Fixes: a9960e6a293e ("ALSA: pcm: fix fifo_size frame calculation") Signed-off-by: Zubin Mithra <zsm(a)chromium.org> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d5883432085..a144a3f68e9e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1746,7 +1746,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, channels = params_channels(params); frame_size = snd_pcm_format_size(format, channels); if (frame_size > 0) - params->fifo_size /= (unsigned)frame_size; + params->fifo_size /= frame_size; } return 0; } -- 2.33.0.259.gc128427fd7-goog

3 2

[PATCH] ALSA: usb-audio: Work around for XRUN with low latency playback
by Takashi Iwai 27 Aug '21

27 Aug '21

The recent change for low latency playback works in most of test cases but it turned out still to hit errors on some use cases, most notably with JACK with small buffer sizes. This is because USB-audio driver fills up and submits full URBs at the beginning, while the URBs would return immediately and try to fill more -- that can easily trigger XRUN. It was more or less expected, but in the small buffer size, the problem became pretty obvious. Fixing this behavior properly would require the change of the fundamental driver design, so it's no trivial task, unfortunately. Instead, here we work around the problem just by switching back to the old method when the given configuration is too fragile with the low latency stream handling. As a threshold, we calculate the total buffer bytes in all plus one URBs, and check whether it's beyond the PCM buffer bytes. The one extra URB is needed because XRUN happens at the next submission after the first round. Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 4 ++++ sound/usb/pcm.c | 13 +++++++++++-- 3 files changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 6c0a052a28f9..5b19901f305a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,7 @@ struct snd_usb_endpoint { struct list_head ready_playback_urbs; /* playback URB FIFO for implicit fb */ unsigned int nurbs; /* # urbs */ + unsigned int nominal_queue_size; /* total buffer sizes in URBs */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ char *syncbuf; /* sync buffer for all sync URBs */ @@ -187,6 +188,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + bool early_playback_start; /* early start needed for playback? */ struct media_ctl *media_ctl; }; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8b1bec51c806..bf26c04cf471 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1126,6 +1126,10 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep) INIT_LIST_HEAD(&u->ready_list); } + /* total buffer bytes of all URBs plus the next queue; + * referred in pcm.c + */ + ep->nominal_queue_size = maxsize * urb_packs * (ep->nurbs + 1); return 0; out_of_memory: diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4e5031a68064..f5cbf61ac366 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -614,6 +614,14 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->period_elapsed_pending = 0; runtime->delay = 0; + /* check whether early start is needed for playback stream */ + subs->early_playback_start = + subs->direction == SNDRV_PCM_STREAM_PLAYBACK && + subs->data_endpoint->nominal_queue_size >= subs->buffer_bytes; + + if (subs->early_playback_start) + ret = start_endpoints(subs); + unlock: snd_usb_unlock_shutdown(chip); return ret; @@ -1394,7 +1402,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->trigger_tstamp_pending_update = false; } - if (period_elapsed && !subs->running) { + if (period_elapsed && !subs->running && !subs->early_playback_start) { subs->period_elapsed_pending = 1; period_elapsed = 0; } @@ -1448,7 +1456,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea prepare_playback_urb, retire_playback_urb, subs); - if (cmd == SNDRV_PCM_TRIGGER_START) { + if (!subs->early_playback_start && + cmd == SNDRV_PCM_TRIGGER_START) { err = start_endpoints(subs); if (err < 0) { snd_usb_endpoint_set_callback(subs->data_endpoint, -- 2.26.2

1 0

Adjust appl_ptr for multi plugin on rewind/forward operations
by GitHub pull_request - edited 27 Aug '21

27 Aug '21

alsa-project/alsa-lib pull request #161 was edited from sava-j: * When doing rewind or forward operations, application pointer needs to be properly adjusted, in the same way when doing commit/write operations. * Introduce helper functions for manipulating the application pointer. Request URL : https://github.com/alsa-project/alsa-lib/pull/161 Patch URL : https://github.com/alsa-project/alsa-lib/pull/161.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: pcm: fix divide error in snd_pcm_lib_ioctl
by Zubin Mithra 27 Aug '21

27 Aug '21

Syzkaller reported a divide error in snd_pcm_lib_ioctl. fifo_size is of type snd_pcm_uframes_t(unsigned long). If frame_size is 0x100000000, the error occurs. Fixes: a9960e6a293e ("ALSA: pcm: fix fifo_size frame calculation") Signed-off-by: Zubin Mithra <zsm(a)chromium.org> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d5883432085..e41b4e01aa37 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1746,7 +1746,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, channels = params_channels(params); frame_size = snd_pcm_format_size(format, channels); if (frame_size > 0) - params->fifo_size /= (unsigned)frame_size; + params->fifo_size /= (unsigned long)frame_size; } return 0; } -- 2.33.0.259.gc128427fd7-goog

2 2

[PATCH 0/6] ASoC: audio-graph: cleanups
by Kuninori Morimoto 27 Aug '21

27 Aug '21

Hi Mark These patches cleanups audio-graph. This is part of prepare for new audio-graph-card2. Kuninori Morimoto (6): ASoC: audio-graph: move audio_graph_card_probe() to simple-card-utils.c ASoC: audio-graph: move audio_graph_remove() to simple-card-utils.c ASoC: audio-graph: check ports if exists ASoC: audio-graph: remove "audio-graph-card," preix support ASoC: audio-graph: remove unused "node" from graph_parse_mclk_fs() ASoC: audio-graph: remove Platform support include/sound/graph_card.h | 4 -- include/sound/simple_card_utils.h | 3 ++ sound/soc/generic/audio-graph-card.c | 52 ++++-------------------- sound/soc/generic/simple-card-utils.c | 25 ++++++++++++ sound/soc/generic/simple-card.c | 7 ---- sound/soc/tegra/tegra_audio_graph_card.c | 4 +- 6 files changed, 38 insertions(+), 57 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

3 10

[PATCH] seq: event_data_ctl: fix getter API for value parameter
by GitHub pull_request - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-gobject pull request #61 was opened from takaswie: ALSASeq.EventDataCtl.get_value() is programmed carelessly to return the value in parameter field instead of value field. This commit fixes the bug. Request URL : https://github.com/alsa-project/alsa-gobject/pull/61 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/61.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: dwc: Get IRQ optionally
by Robin Murphy 27 Aug '21

27 Aug '21

The IRQ is explicitly optional, so use platform_get_irq_optional() and avoid platform_get_irq() logging a spurious error when trying to use the thing in DMA mode. Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 8ebf76e04702..33ce257ae198 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -642,7 +642,7 @@ static int dw_i2s_probe(struct platform_device *pdev) dev->dev = &pdev->dev; - irq = platform_get_irq(pdev, 0); + irq = platform_get_irq_optional(pdev, 0); if (irq >= 0) { ret = devm_request_irq(&pdev->dev, irq, i2s_irq_handler, 0, pdev->name, dev); -- 2.25.1

3 2

[PATCH 1/2] ASoC: fsl_rpmsg: add soc specific data structure
by Shengjiu Wang 27 Aug '21

27 Aug '21

Each platform has different supported rates and formats, so add soc specific data for each platform. This soc specific data is attached with compatible string. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_rpmsg.c | 47 +++++++++++++++++++++++++++++++++++---- sound/soc/fsl/fsl_rpmsg.h | 12 ++++++++++ 2 files changed, 55 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index d60f4dac6c1b..bda6cc96071d 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -138,11 +138,42 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", }; +static const struct fsl_rpmsg_soc_data imx7ulp_data = { + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mm_data = { + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U8 | + SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mn_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mp_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + static const struct of_device_id fsl_rpmsg_ids[] = { - { .compatible = "fsl,imx7ulp-rpmsg-audio"}, - { .compatible = "fsl,imx8mm-rpmsg-audio"}, - { .compatible = "fsl,imx8mn-rpmsg-audio"}, - { .compatible = "fsl,imx8mp-rpmsg-audio"}, + { .compatible = "fsl,imx7ulp-rpmsg-audio", .data = &imx7ulp_data}, + { .compatible = "fsl,imx8mm-rpmsg-audio", .data = &imx8mm_data}, + { .compatible = "fsl,imx8mn-rpmsg-audio", .data = &imx8mn_data}, + { .compatible = "fsl,imx8mp-rpmsg-audio", .data = &imx8mp_data}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); @@ -157,6 +188,14 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) if (!rpmsg) return -ENOMEM; + rpmsg->soc_data = of_device_get_match_data(&pdev->dev); + if (rpmsg->soc_data) { + fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; + fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + } + if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; rpmsg->buffer_size = LPA_LARGE_BUFFER_SIZE; diff --git a/sound/soc/fsl/fsl_rpmsg.h b/sound/soc/fsl/fsl_rpmsg.h index 4f5b49eb18d8..b04086fbf828 100644 --- a/sound/soc/fsl/fsl_rpmsg.h +++ b/sound/soc/fsl/fsl_rpmsg.h @@ -6,6 +6,16 @@ #ifndef __FSL_RPMSG_H #define __FSL_RPMSG_H +/* + * struct fsl_rpmsg_soc_data + * @rates: supported rates + * @formats: supported formats + */ +struct fsl_rpmsg_soc_data { + int rates; + u64 formats; +}; + /* * struct fsl_rpmsg - rpmsg private data * @@ -15,6 +25,7 @@ * @pll8k: parent clock for multiple of 8kHz frequency * @pll11k: parent clock for multiple of 11kHz frequency * @card_pdev: Platform_device pointer to register a sound card + * @soc_data: soc specific data * @mclk_streams: Active streams that are using baudclk * @force_lpa: force enable low power audio routine if condition satisfy * @enable_lpa: enable low power audio routine according to dts setting @@ -27,6 +38,7 @@ struct fsl_rpmsg { struct clk *pll8k; struct clk *pll11k; struct platform_device *card_pdev; + const struct fsl_rpmsg_soc_data *soc_data; unsigned int mclk_streams; int force_lpa; int enable_lpa; -- 2.17.1

3 2

No cards found in 1.2.5
by GitHub issues - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-ucm-conf issue #111 was opened from Tooniis: On 1.2.4: ``` $ alsaucm listcards 0: HDA-Intel 1: SOF 2: bdw-rt5677 3: broadwell-rt286 4: broxton-rt298 5: bytcht-cx2072x 6: bytcht-es8316 7: bytcr-rt5640 8: bytcr-rt5651 9: cht-bsw-rt5672 10: chtmax98090 11: chtnau8824 chtnau8824 internal card 12: chtrt5645 Intel SoC Audio Device 13: chtrt5650 Intel SoC Audio Device 14: hda-dsp 15: kblrt5660 16: skylake-rt286 17: sof-hda-dsp 18: sof-soundwire ``` On 1.2.5: ``` $ alsaucm listcards list is empty ``` My card's config (snd-soc-apq8096, using DB820c config) isn't being loaded on 1.2.5 as a result and I get no sound. Note that in both cases above, alsa-ucm-conf and alsa-lib are of the same version; as mixing versions causes some other errors (lack of conf.virt.d on alsa-lib 1.2.5 alsa-ucm-conf 1.2.4, and invalid syntax 4 on alsa-lib 1.2.4 with alsa-ucm-conf 1.2.5) Using Arch Linux ARM. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/111 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 0/14] Patches to update for rockchip i2s
by Sugar Zhang 27 Aug '21

27 Aug '21

These patches fixup or update for rockchip i2s. Changes in v3: - Drop property 'rockchip,playback-only', 'rockchip,capture-only'. Implement it by 'dma-names' of DT instead. Changes in v2: - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - drop change-id Sugar Zhang (12): ASoC: rockchip: i2s: Add support for set bclk ratio ASoC: rockchip: i2s: Fixup clk div error ASoC: rockchip: i2s: Improve dma data transfer efficiency ASoC: rockchip: i2s: Fix regmap_ops hang ASoC: rockchip: i2s: Fix concurrency between tx/rx ASoC: rockchip: i2s: Reset the controller if soft reset failed ASoC: dt-bindings: rockchip: Document reset property for i2s ASoC: rockchip: i2s: Make playback/capture optional ASoC: rockchip: i2s: Add compatible for more SoCs ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs ASoC: rockchip: i2s: Add support for frame inversion ASoC: dt-bindings: rockchip: i2s: Document property TRCM Xiaotan Luo (1): ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B Xing Zheng (1): ASoC: rockchip: i2s: Add support for TRCM property .../devicetree/bindings/sound/rockchip-i2s.yaml | 19 ++ sound/soc/rockchip/rockchip_i2s.c | 278 +++++++++++++++------ sound/soc/rockchip/rockchip_i2s.h | 10 +- 3 files changed, 224 insertions(+), 83 deletions(-) -- 2.7.4

2 13

[PATCH v1 0/7] Patches to update for rockchip pdm
by Sugar Zhang 27 Aug '21

27 Aug '21

These patches fixup or update for rockchip pdm. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 --------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 114 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 ++++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 226 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

2 10

Error in recording and playing at the same time on raspberry pi
by GitHub issues - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-lib issue #174 was opened from amirush786: I am try to record and play at the same time. I have a speaker connected to pi via aux cable. I have to play a sound from the speaker and also re ord it at the same time from the tool which is seeed microphone tool which is connected to pi. I am able to do that but the recorded file has some hissssshhhhhhhh sound in it don't know why it's coming out to that way. But when I do a individual record I can hear a bit proper noise and less of hisshy noise in it. And also how can I check the capture sound is high or not. I followed the approach below. I did Alsamixer command and it's shows me this. So after that I typed in F6 to change my card to seeed card. Then is scroll down to seed card and hit enter. So I see this. Then I press f4 and I see this so I don't know where should I see that capture is high. This is the last thing that messing with me. ![20210822_200619.jpg](https://user-images.githubusercontent.com/31735000/131046816-b77895b8-f6c8-4faa-b624-283100ad00f8.jpg) ![20210822_200626.jpg](https://user-images.githubusercontent.com/31735000/131046829-fab9edb3-d9b2-4938-be84-e7dcccfb4aaf.jpg) ![20210822_200641.jpg](https://user-images.githubusercontent.com/31735000/131046851-631a8e34-33f3-4e88-b971-726b2fdd7756.jpg) Issue URL : https://github.com/alsa-project/alsa-lib/issues/174 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] [v3] ASoC: Intel: sof_rt5682: Add support for max98390 speaker amp
by Mark Hsieh 26 Aug '21

26 Aug '21

Configure adl_max98390_rt5682 to support the rt5682 headset codec and max98390 speaker. Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 57 +++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 12 ++++ sound/soc/intel/boards/sof_rt5682.c | 19 +++++++ .../intel/common/soc-acpi-intel-adl-match.c | 13 +++++ 5 files changed, 102 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7e29b0d911e2..0017c08c5a62 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -470,6 +470,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1015 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..a6fcf6e08996 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,63 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { /* For Right */ + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { /* For Left */ + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_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; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* DEV0 tdm slot configuration */ + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 2, 16); + /* DEV1 tdm slot configuration */ + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 2, 16); + } + return 0; +} + +const struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..bad7b5303280 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,18 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern const struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..18b6ef311462 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,7 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -781,6 +782,13 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); } @@ -917,6 +925,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682); if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1043,6 +1053,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); 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 a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.17.1

3 2

[PATCH 0/3] ASoC: wcd9335: Firx some resources leak in the probe and remove function
by Christophe JAILLET 26 Aug '21

26 Aug '21

The first 2 patches are sraightforward and look logical to me. However, the 3rd one in purely speculative. It is based on the fact that a comment states that we enable some irqs on some slave ports. That said, it writes 0xFF in some registers. So, I guess that we should disable these irqs when the driver is removed. That said, writing 0x00 at the same place looks logical to me. This cis untested and NOT based on any documentation. Just a blind fix. Review with care. You'll be warned :) Christophe JAILLET (3): ASoC: wcd9335: Fix a double irq free in the remove function ASoC: wcd9335: Fix a memory leak in the error handling path of the probe function ASoC: wcd9335: Disable irq on slave ports in the remove function sound/soc/codecs/wcd9335.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) -- 2.30.2

2 4

[PATCH v2] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 26 Aug '21

26 Aug '21

The document was sent in [1] and merged to ASoC for-next branch as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [2] revealed that the commit 1afc60e00de3 breaks dt_binding_check in linux-next due to dt-bindings/clock/mt8192-clk.h doesn't exist. [3] reverted the commit 1afc60e00de3 and merged as commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document"). dt-bindings/clock/mt8192-clk.h now can be found in linux-next per [4], re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176577.… [2]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… [3]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176897.… [4]: https://git.kernel.org/pub/scm/linux/kernel/git/clk/linux.git/commit/?h=clk… Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v1 (https://patchwork.kernel.org/project/alsa-devel/list/?series=537817): - 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.rc2.250.ged5fa647cd-goog

2 1

[PATCH] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 26 Aug '21

26 Aug '21

The document was removed temporarily by commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") because the dependency was not merged at that time. The dependency has merged. Re-adds back the document. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- More context: - [1] reported `dt_binding_check` was broken. - [2] reverted. - The dependency has merged in [3] and it can be found from linux-next. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/1604390378-23993-8-gi… [2]: https://patchwork.kernel.org/project/alsa-devel/patch/1605081920-11848-1-gi… [3]: https://git.kernel.org/pub/scm/linux/kernel/git/clk/linux.git/commit/?h=clk… .../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.rc2.250.ged5fa647cd-goog

2 3

[PATCH] ALSA: hda: Disable runtime resume at shutdown
by Takashi Iwai 26 Aug '21

26 Aug '21

Although we modified the codec shutdown callback to perform runtime-suspend, it's still not fully effective, as this may be resumed again at any time later. For fixing such an unwanted resume, this patch replaces pm_runtime_suspend() with pm_runtime_force_suspend(), and call pm_runtime_disable() afterward. It assures to keep the device suspended. Also for code simplification, we apply the code unconditionally; when it's been already suspended, nothing would happen by calls of snd_pcm_suspend_all() and pm_runtime_force_suspend(), just proceed to pm_runtime_disable(). Fixes: b98444ed597d ("ALSA: hda: Suspend codec at shutdown") Reported-and-tested-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c91c92c9ab2..a9ebefd60cf6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2986,13 +2986,11 @@ void snd_hda_codec_shutdown(struct hda_codec *codec) { struct hda_pcm *cpcm; - if (pm_runtime_suspended(hda_codec_dev(codec))) - return; - list_for_each_entry(cpcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(cpcm->pcm); - pm_runtime_suspend(hda_codec_dev(codec)); + pm_runtime_force_suspend(hda_codec_dev(codec)); + pm_runtime_disable(hda_codec_dev(codec)); } /* -- 2.31.1

1 0

Re: [PATCH 2/2] ALSA: hda/cs8409: Prevent pops and clicks during reboot
by Takashi Iwai 26 Aug '21

26 Aug '21

On Thu, 26 Aug 2021 13:49:32 +0200, Vitaly Rodionov wrote: > > On 26/08/2021 11:49 am, Takashi Iwai wrote: > > On Thu, 26 Aug 2021 08:03:45 +0200, > > Takashi Iwai wrote: > >> On Wed, 25 Aug 2021 20:04:05 +0200, > >> Vitaly Rodionov wrote: > >>> Actually when codec is suspended and we do reboot from UI, then sometimes we > >>> see suspend() calls in kernel log and no pops, but sometimes > >>> > >>> we still have no suspend() on reboot and we hear pops. But when we do reboot > >>> from command line: > sudo reboot then we always have pops and no suspend() > >>> called. > >>> > >>> Then we have added extra logging and we can see that on reboot codec somehow > >>> getting resume() call and we run jack detect on resume that causing pops. > >> Hm, it's interesting who triggers the runtime resume. > >> > >>> We were thinking about possible solution for that and we would propose some > >>> changes in generic code hda_bind.c: > >>> > >>> static void hda_codec_driver_shutdown(struct device *dev) { + if (codec-> > >>> patch_ops.suspend) + codec->patch_ops.suspend(codec); > >>> snd_hda_codec_shutdown(dev_to_hda_codec(dev)); + hda_codec_driver_remove > >>> (dev_to_hda_codec(dev)); } > >> Sorry, it's no-no. The suspend can't be called unconditionally, and > >> the driver unbind must not be called in the callback itself. > >> > >> Does the patch below work instead? > > Sorry there was a typo. A bit more revised patch is below. > > > > > > Takashi > > > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -1383,14 +1383,17 @@ static void azx_free(struct azx *chip) > > hda->freed = 1; > > } > > -static int azx_dev_disconnect(struct snd_device *device) > > +static void __azx_disconnect(struct azx *chip) > > { > > - struct azx *chip = device->device_data; > > struct hdac_bus *bus = azx_bus(chip); > > chip->bus.shutdown = 1; > > cancel_work_sync(&bus->unsol_work); > > +} > > +static int azx_dev_disconnect(struct snd_device *device) > > +{ > > + __azx_disconnect(device->device_data); > > return 0; > > } > > @@ -2356,8 +2359,10 @@ static void azx_shutdown(struct pci_dev > > *pci) > > if (!card) > > return; > > chip = card->private_data; > > - if (chip && chip->running) > > + if (chip && chip->running) { > > + __azx_disconnect(chip); > > azx_shutdown_chip(chip); > > + } > > } > > /* PCI IDs */ > > Hi Takashi, > > Applied fix and tested on dolphin HW. Issue still there, here is > captured screen on reboot from command line: > > reboot capture > > Reboot from UI works differently, no resume() call in this case. Thanks for quick testing. After reconsideration, I believe we can even take a simpler path. Use pm_runtime_force_suspend(), and keep suspended by pm_runtime_disable() call afterwards. Below is another test patch. Could you check whether this works better? Takashi --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2986,13 +2986,11 @@ void snd_hda_codec_shutdown(struct hda_codec *codec) { struct hda_pcm *cpcm; - if (pm_runtime_suspended(hda_codec_dev(codec))) - return; - list_for_each_entry(cpcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(cpcm->pcm); - pm_runtime_suspend(hda_codec_dev(codec)); + pm_runtime_force_suspend(hda_codec_dev(codec)); + pm_runtime_disable(hda_codec_dev(codec)); } /*

2 3

[PATCH][next] ASoC: mediatek: mt8195: Fix spelling mistake "bitwiedh" -> "bitwidth"
by Colin King 26 Aug '21

26 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_dbg message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c index bd9876cf02c8..7378e42f2766 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c @@ -1909,7 +1909,7 @@ static int mtk_dai_etdm_configure(struct mtk_base_afe *afe, __func__, etdm_data->format, etdm_data->data_mode, etdm_data->lrck_inv, etdm_data->lrck_width, etdm_data->bck_inv, etdm_data->clock_mode, etdm_data->slave_mode); - dev_dbg(afe->dev, "%s rate %u channels %u bitwiedh %u, id %d\n", + dev_dbg(afe->dev, "%s rate %u channels %u bitwidth %u, id %d\n", __func__, rate, channels, bit_width, dai_id); etdm_channels = (etdm_data->data_mode == MTK_DAI_ETDM_DATA_ONE_PIN) ? -- 2.32.0

2 1

[PATCH v1 0/4] Patches to update for rockchip spdif
by Sugar Zhang 26 Aug '21

26 Aug '21

These patches fixup or update for rockchip spdif. Sugar Zhang (4): ASoC: rockchip: spdif: Mark SPDIF_SMPDR as volatile ASoC: rockchip: spdif: Fix some coding style ASoC: rockchip: spdif: Add support for rk3568 spdif ASoC: dt-bindings: rockchip: Add compatible for rk3568 spdif .../devicetree/bindings/sound/rockchip-spdif.yaml | 1 + sound/soc/rockchip/rockchip_spdif.c | 38 ++++++++++++---------- 2 files changed, 22 insertions(+), 17 deletions(-) -- 2.7.4

3 6

[PATCH 1/2] ALSA: hda/cs8409: Prevent pops and clicks during suspend
by Vitaly Rodionov 26 Aug '21

26 Aug '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Some of the register values set for type detection cause pops during suspend, ensure these are cleaned up after type detection completes, as well ensuring that these are cleared when we suspend. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 272497b6cfcb..9db16b6292f4 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -708,6 +708,10 @@ static int cs42l42_jack_unsol_event(struct sub_codec *cs42l42) cs42l42->mic_jack_in = 1; } } + /* Configure the HSDET mode. */ + cs8409_i2c_write(cs42l42, 0x1120, 0x80); + /* Enable the HPOUT ground clamp and configure the HP pull-down */ + cs8409_i2c_write(cs42l42, 0x1F06, 0x02); /* Re-Enable Tip Sense Interrupt */ cs8409_i2c_write(cs42l42, 0x1320, 0xF3); } else { @@ -756,6 +760,8 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) unsigned int gpio_data; int reg_cdc_status = 0; const struct cs8409_i2c_param cs42l42_pwr_down_seq[] = { + { 0x1F06, 0x02 }, + { 0x1129, 0x00 }, { 0x2301, 0x3F }, { 0x2302, 0x3F }, { 0x2303, 0x3F }, -- 2.25.1

2 10

azx_get_pos_skl() induced system slowness
by Jens Axboe 26 Aug '21

26 Aug '21

Hi, Got a new notebook recently, it's a Lenovo X1 Carbon 9th gen. Sound works fine, but sometimes I get really stuttering playback from nestopia and I finally decided to look into it. When this happens, azx_get_pos_skl() is seemingly called a lot, at least it uses a ton of CPU cycles. This comes and goes, sometimes 1 minute in between, sometimes 2, and sometimes 30 seconds. If I comment out the udelay() in that function it does seems to be noticeably better, though it's not a complete fix. I guess it just reduces the pain of calling it so many times? This is running 5.14-rc7, but it's not a recent regression. Any clues as to what this might be? diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 470753b36c8a..3c1f233e463f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -878,7 +878,9 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) * for the possible boundary overlap; the read of DPIB fetches the * actual posbuf */ +#if 0 udelay(20); +#endif azx_skl_get_dpib_pos(chip, azx_dev); return azx_get_pos_posbuf(chip, azx_dev); } -- Jens Axboe

2 3

[PATCH] ASoC: Intel: bytcr_rt5640: Make rt5640_jack_gpio/rt5640_jack2_gpio static
by Peter Ujfalusi 25 Aug '21

25 Aug '21

Marking the two jack gpio as static fixes the following Sparse errors: sound/soc/intel/boards/bytcr_rt5640.c:468:26: error: symbol 'rt5640_jack_gpio' was not declared. Should it be static? sound/soc/intel/boards/bytcr_rt5640.c:475:26: error: symbol 'rt5640_jack2_gpio' was not declared. Should it be static? Fixes: 9ba00856686ad ("ASoC: Intel: bytcr_rt5640: Add support for HP Elite Pad 1000G2 jack-detect") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 05667a33d919..a6e837290c7d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -465,14 +465,14 @@ static struct snd_soc_jack_pin rt5640_pins2[] = { }, }; -struct snd_soc_jack_gpio rt5640_jack_gpio = { +static struct snd_soc_jack_gpio rt5640_jack_gpio = { .name = "hp-detect", .report = SND_JACK_HEADSET, .invert = true, .debounce_time = 200, }; -struct snd_soc_jack_gpio rt5640_jack2_gpio = { +static struct snd_soc_jack_gpio rt5640_jack2_gpio = { .name = "hp2-detect", .report = SND_JACK_HEADSET, .invert = true, -- 2.33.0

3 2

[PATCH RESEND] ASoC: rt5682: Fix the vol+ button detection issue
by Derek [方德義] 25 Aug '21

25 Aug '21

Fix the wrong button vol+ detection issue with some brand headsets by fine tuning the threshold of button vol+ and SAR ADC button accuracy. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index fcf442b8a732..e822fa1b9d4b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,7 +43,9 @@ static const struct reg_sequence patch_list[] = { {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, - {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_1, 0x22b7}, + {RT5682_SAR_IL_CMD_3, 0x0365}, + {RT5682_SAR_IL_CMD_6, 0x0110}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -1726,8 +1728,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, ARRAY_SIZE(rt5682_sto1_adc_r_mix)), - SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, - 14, 1, NULL, 0), /* ADC PGA */ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1898,8 +1898,6 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, - {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, - {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, @@ -2949,7 +2947,8 @@ static int rt5682_suspend(struct snd_soc_component *component) /* enter SAR ADC power saving mode */ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, - RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, 0); + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | + RT5682_SAR_BUTDET_RST_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, 0); snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV | RT5682_SAR_BUTDET_RST_NORMAL); @@ -2971,6 +2970,9 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_sync(rt5682->regmap); if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, + RT5682_SAR_BUTDET_POW_NORM | RT5682_SAR_SEL_MB1_MB2_AUTO); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, RT5682_CTRL_MB1_FSM | RT5682_CTRL_MB2_FSM); -- 2.17.1

2 2

[PATCH v2] ASoC: rt5682: Fix the vol+ button detection issue
by derek.fang＠realtek.com 25 Aug '21

25 Aug '21

From: Derek Fang <derek.fang(a)realtek.com> Fix the wrong button vol+ detection issue with some brand headsets by fine tuning the threshold of button vol+ and SAR ADC button accuracy. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index fcf442b8a732..e822fa1b9d4b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,7 +43,9 @@ static const struct reg_sequence patch_list[] = { {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, - {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_1, 0x22b7}, + {RT5682_SAR_IL_CMD_3, 0x0365}, + {RT5682_SAR_IL_CMD_6, 0x0110}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -1726,8 +1728,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, ARRAY_SIZE(rt5682_sto1_adc_r_mix)), - SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, - 14, 1, NULL, 0), /* ADC PGA */ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1898,8 +1898,6 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, - {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, - {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, @@ -2949,7 +2947,8 @@ static int rt5682_suspend(struct snd_soc_component *component) /* enter SAR ADC power saving mode */ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, - RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, 0); + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | + RT5682_SAR_BUTDET_RST_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, 0); snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV | RT5682_SAR_BUTDET_RST_NORMAL); @@ -2971,6 +2970,9 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_sync(rt5682->regmap); if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, + RT5682_SAR_BUTDET_POW_NORM | RT5682_SAR_SEL_MB1_MB2_AUTO); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, RT5682_CTRL_MB1_FSM | RT5682_CTRL_MB2_FSM); -- 2.17.1

2 1

[PATCH] ASoC: imx-rpmsg: change dev_err to dev_err_probe for -EPROBE_DEFER
by Shengjiu Wang 25 Aug '21

25 Aug '21

Change dev_err to dev_err_probe for no need print error message when defer probe happens. Fixes: 39f8405c3e50 ("ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index f0cae8c59d54..f96fe4ff8425 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -125,7 +125,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&data->card, data); ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); goto fail; } -- 2.17.1

2 1

[PATCH] ASoC: wm_adsp: Put debugfs_remove_recursive back in
by Charles Keepax 25 Aug '21

25 Aug '21

This patch reverts commit acbf58e53041 ("ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs"), and adds an alternate solution to the issue. That patch removes the call to debugfs_remove_recursive, which cleans up the DSPs debugfs. The intention was to avoid an unbinding issue on an out of tree driver/platform. The issue with the patch is it means the driver no longer cleans up its own debugfs, instead relying on ASoC to remove recurive on the parent debugfs node. This is conceptually rather unclean, but also it would prevent DSPs being added/removed independently of ASoC and soon we are going to be upstreaming some non-audio parts with these DSPs, which will require this. Finally, it seems the issue on the platform is a result of the wm_adsp2_cleanup_debugfs getting called twice. This is very likely a problem on the platform side and will be resolved there. But in the mean time make the code a little more robust to such issues, and again conceptually a bit nicer, but clearing the debugfs_root variable in the DSP structure when the debugfs is removed. Fixes: acbf58e53041 ("ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs" Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bbe27ab3b1fc3..f7c800927cb2f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -747,6 +747,8 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); + debugfs_remove_recursive(dsp->debugfs_root); + dsp->debugfs_root = NULL; } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.11.0

2 1

[PATCH linux-next] ASoC: SOF: intel: remove duplicate include
by CGEL 25 Aug '21

25 Aug '21

From: Changcheng Deng <deng.changcheng(a)zte.com.cn> Clean up the following includecheck warning: ./sound/soc/sof/intel/pci-tng.c: shim.h is included more than once. No functional change. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Changcheng Deng <deng.changcheng(a)zte.com.cn> --- sound/soc/sof/intel/pci-tng.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 4ee1da3..4bded66 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -15,7 +15,6 @@ #include <sound/sof.h> #include "../ops.h" #include "atom.h" -#include "shim.h" #include "../sof-pci-dev.h" #include "../sof-audio.h" -- 1.8.3.1

3 2

[PATCH v2 0/15] Patches to update for rockchip i2s
by Sugar Zhang 25 Aug '21

25 Aug '21

These patches fixup or update for rockchip i2s. Changes in v2: - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - drop change-id Sugar Zhang (13): ASoC: rockchip: i2s: Add support for set bclk ratio ASoC: rockchip: i2s: Fixup clk div error ASoC: rockchip: i2s: Improve dma data transfer efficiency ASoC: rockchip: i2s: Fix regmap_ops hang ASoC: rockchip: i2s: Fix concurrency between tx/rx ASoC: rockchip: i2s: Reset the controller if soft reset failed ASoC: dt-bindings: rockchip: Document reset property for i2s ASoC: rockchip: i2s: Add property to specify play/cap capability ASoC: dt-bindings: rockchip: i2s: Document property for playback/capture ASoC: rockchip: i2s: Add compatible for more SoCs ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs ASoC: rockchip: i2s: Add support for frame inversion ASoC: dt-bindings: rockchip: i2s: Document property TRCM Xiaotan Luo (1): ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B Xing Zheng (1): ASoC: rockchip: i2s: Add support for TRCM property .../devicetree/bindings/sound/rockchip-i2s.yaml | 29 ++++ sound/soc/rockchip/rockchip_i2s.c | 161 +++++++++++++++++---- sound/soc/rockchip/rockchip_i2s.h | 10 +- 3 files changed, 164 insertions(+), 36 deletions(-) -- 2.7.4

2 20

[bug report] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Dan Carpenter 25 Aug '21

25 Aug '21

Hello Kuninori Morimoto, The patch cc64c390b215: "ASoC: rsnd: adg: clearly handle clock error / NULL case" from Aug 20, 2021, leads to the following Smatch static checker warning: sound/soc/sh/rcar/adg.c:396 rsnd_adg_create_null_clk() warn: passing zero to 'ERR_CAST' sound/soc/sh/rcar/adg.c 386 static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, 387 const char * const name, 388 const char *parent) 389 { 390 struct device *dev = rsnd_priv_to_dev(priv); 391 struct clk *clk; 392 393 clk = clk_register_fixed_rate(dev, name, parent, 0, 0); 394 if (IS_ERR_OR_NULL(clk)) { 395 dev_err(dev, "create null clk error\n"); The clk_register_fixed_rate() function doesn't have any documentation, but generally when functions return a mix of error pointers and NULL then the NULL case means the feature is deliberately disabled. So it should not print an error message. Perhaps an info message at most. --> 396 return ERR_CAST(clk); No need to cast this. The types are the same. 397 } 398 399 return clk; 400 } regards, dan carpenter

2 1

[PATCH v2 0/4] Rockchip I2S/TDM controller
by Nicolas Frattaroli 24 Aug '21

24 Aug '21

Changes in v2: - remove ad-hoc writeq and needless (and broken) optimisation in reset assert/deassert. This wouldn't have worked on Big Endian, and would've been pointless on any other platform, as the overhead for saving one write was comparatively big - fix various checkpatch issues - get rid of leftover clk-trcm in schema - set status = "okay" in example in schema instead of "disabled" - change dma-names so rx is first, adjust device trees as necessary - properly reference uint32-array for rx-route and tx-route instead of uint32 - replace trcm-sync with two boolean properties, adjust DT changes accordingly and also get rid of the header file - get rid of rockchip,no-dmaengine. This was only needed for some downstream driver and shouldn't be in the DT - get rid of rockchip,capture-only/playback-only. Rationale being that I have no way to test whether they're needed, and unconditionally setting channels_min to 0 breaks everything - change hclk description in "clocks" Additionally, I have been made aware that registers and code seem to be an extension of the rockchip_i2s driver. This is true, though merging the two seems like more effort than just slapping a few compatibles and properties into the mix. I'll investigate it further in the coming weeks, but would likely still prefer basing any unified driver on mine rather than the other way around, since I think it has better code quality now. Original cover letter: Hello, I come bearing four patches for your consideration. The first of these four patches adds a driver for the Rockchip I2S/TDM controller, used in interfacing between the AHB bus and the I2S bus on some Rockchip SoCs. This allows for audio playback with a matching codec. The controller has three different modes: I2S, I2S/TDM and PCM. It is distinct from the earlier Rockchip I2S controller, and therefore not just an extension of that driver. The driver is based on the downstream version, though various changes have been made to hopefully make it more palatable to upstream. Some needless code duplication has been refactored, and the probe function will no longer let wrong device tree values write nonsense to hardware registers. Properties have been renamed and had their semantics changed. I won't bore you with the details of what downstream did, since that's not what I'm submitting, but the changes are significant enough that I've added myself to the list of authors. The second patch adds the YAML device tree bindings for this, which have been written from scratch by yours truly. Since I didn't like having random integers mean things, I defined them as constants in a header file for the bindings. The third patch adds the i2s1 controller to the rk356x device tree. I didn't add any of the other i2s controllers on that SoC for now as I have no way of testing them; in particular, i2s0 is tied to HDMI, so needs a functioning VOP2 driver to even have a chance of working. The fourth patch makes use of the i2s1 controller to enable analog audio output on the Quartz64 Model A through its RK817 codec. I've tested this to work properly at both 44.1 kHz and 96 kHz, so both mclk_root0 and mclk_root1 are definitely functioning. This is my first kernel contribution, so I most likely did something horribly wrong. That's why I'm more than happy to receive any criticisms and concerns over how the driver is implemented, because I've run out of ideas on how to make it clearly better myself. I'd also like to extend my thanks to Peter Geis, who has been acting as somewhat of a mentor and gave me occasional feedback and ideas during the writing of this patch series. Regards, Nicolas Frattaroli Nicolas Frattaroli (4): ASoC: rockchip: add support for i2s-tdm controller dt-bindings: sound: add rockchip i2s-tdm binding arm64: dts: rockchip: add i2s1 on rk356x arm64: dts: rockchip: add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 193 ++ .../boot/dts/rockchip/rk3566-quartz64-a.dts | 35 +- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 26 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1737 +++++++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 398 ++++ 7 files changed, 2401 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h -- 2.32.0

5 10

[PATCH] [v2] ASoC: Intel: sof_rt5682: Add support for max98390 speaker amp
by Mark Hsieh 24 Aug '21

24 Aug '21

Configure adl_max98390_rt5682 to support the rt5682 headset codec and max98390 speaker BUG=b:191811888 TEST=emerge-brya chromeos-kernel-5_10 Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 59 +++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 12 ++++ sound/soc/intel/boards/sof_rt5682.c | 20 ++++++- .../intel/common/soc-acpi-intel-adl-match.c | 13 ++++ 5 files changed, 104 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7e29b0d911e2..0017c08c5a62 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -470,6 +470,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1015 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..477883d571ab 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,65 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { /* For Right */ + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { /* For Left */ + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_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; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 2, 16); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 2, 16); + } + } + return 0; +} + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..2458758866f3 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,18 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..dc4966056b7d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,7 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -781,6 +782,13 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); } @@ -917,7 +925,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); - + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682); if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1043,6 +1052,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); 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 a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.17.1

4 3

alsactl: flush stdout
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-utils pull request #109 was opened from Hello71: Signed-off-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Request URL : https://github.com/alsa-project/alsa-utils/pull/109 Patch URL : https://github.com/alsa-project/alsa-utils/pull/109.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Pinebook Pro's audio does not work
by GitHub issues - edited 24 Aug '21

24 Aug '21

alsa-project/alsa-ucm-conf issue #110 was edited from starnight: I have a [Pinebook Pro](https://www.pine64.org/pinebook-pro/) whose audio (including the internal speaker, headphone and internal MIC) does not work by default with general Linux. Here is the [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7037916/… I notice Manjaro ARM has [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… in [pinebookpro-post-install](https://gitlab.manjaro.org/manjaro-arm/packages/c… package. If I install the [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… into `/var/lib/alsa/`, then `alsactl restore`, all of the audio works! However, I guess that **asound.state** is produced as a result, not the original configuration. If this relates to alsa-ucm-conf project, how should I configure the ucm for Pinebook Pro's audio? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/110 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Pinebook Pro's audio does not work
by GitHub issues - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-ucm-conf issue #110 was opened from starnight: I have a Pinebook Pro whose audio (including the internal speaker, headphone and internal MIC) does not work by default with general Linux. Here is the [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7037916/… I notice Manjaro ARM has [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… in [pinebookpro-post-install](https://gitlab.manjaro.org/manjaro-arm/packages/c… package. If I install the [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… into `/var/lib/alsa/`, then `alsactl restore`, all of the audio works! However, I guess that **asound.state** is produced as a result, not the original configuration. If this relates to alsa-ucm-conf project, how should I configure the ucm for Pinebook Pro's audio? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/110 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Timestamp
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-utils pull request #108 was opened from folkertvanheusden: Added -T / --timestamp (with parameter "realtime", "monotonic" or "raw") which adds a timestamp before each received message when using --dump. Request URL : https://github.com/alsa-project/alsa-utils/pull/108 Patch URL : https://github.com/alsa-project/alsa-utils/pull/108.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

arecord is failing with -V stereo
by sujith kumar reddy 24 Aug '21

24 Aug '21

Hi All, arecord is failing vumeter option -V stereo only. localhost ~ # arecord -Dhw:1,2 -r48000 -c2 -fS32_LE /tmp/test_record.wav -M -d 1 -V stereo Recording WAVE '/tmp/test_record.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo *** buffer overflow detected ***: terminated Aborted by signal Aborted... Please provide pointers to debug this option. Thanks Sujith

3 16

[PATCH alsa-utils 0/5] aplay: Fix bugs in peak calculations
by Takashi Iwai 24 Aug '21

24 Aug '21

The code in aplay for sample peak calculations and VU-meter representation seems to have a lot of bugs. Let's fix them. Takashi === Takashi Iwai (5): aplay: Fix conversion of unsigned samples in peak calculation aplay: Handle 16bit sample negative overflow in peak calculations aplay: Don't pass most negative integer to abs() in peak calculations aplay: Handle upper bound in peak calculations aplay: Fix out-of-bound access in stereo VU meter drawing aplay/aplay.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) -- 2.26.2

1 5

rawmidi: hide the frame structure using a decoder function
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-lib pull request #173 was opened from perexg: The frame structure should be hidden. Remove it and add snd_rawmidi_decode_frame0() which returns the timestamp and the midi chunk pointer to the application. BugLink: https://github.com/alsa-project/alsa-lib/issues/172 Request URL : https://github.com/alsa-project/alsa-lib/pull/173 Patch URL : https://github.com/alsa-project/alsa-lib/pull/173.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH for-5.15] ALSA: usb-audio: Move set-interface-first workaround into common quirk
by Takashi Iwai 24 Aug '21

24 Aug '21

The recent quirk for WALKMAN (commit 7af5a14371c1: "ALSA: usb-audio: Fix regression on Sony WALKMAN NW-A45 DAC") may be required for other devices and is worth to be put into the common quirk flags. This patch adds a new quirk flag bit QUIRK_FLAG_SET_IFACE_FIRST and a quirk table entry for the device. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 1 + sound/usb/endpoint.c | 6 ++---- sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 3 +++ 4 files changed, 8 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index e3c33b8d6d78..65f61695f561 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2278,6 +2278,7 @@ quirk_flags * bit 13: Disable runtime PM autosuspend * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format + * bit 16: Set up the interface at first like UAC1 This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6d58d7674a0..79a8c569c62b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1293,10 +1293,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to be set up before parameter setups */ iface_first = ep->cur_audiofmt->protocol == UAC_VERSION_1; - /* Workaround for Sony WALKMAN NW-A45 DAC; - * it requires the interface setup at first like UAC1 - */ - if (chip->usb_id == USB_ID(0x054c, 0x0b8c)) + /* Workaround for devices that require the interface setup at first like UAC1 */ + if (chip->quirk_flags & QUIRK_FLAG_SET_IFACE_FIRST) iface_first = true; if (iface_first) { err = endpoint_set_interface(chip, ep, true); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 670abc6318f2..4479a590194f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1790,6 +1790,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ QUIRK_FLAG_SKIP_CLOCK_SELECTOR | QUIRK_FLAG_CTL_MSG_DELAY_5M), + DEVICE_FLG(0x054c, 0x0b8c, /* Sony WALKMAN NW-A45 DAC */ + QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x0556, 0x0014, /* Phoenix Audio TMX320VC */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x05a3, 0x9420, /* ELP HD USB Camera */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4e93668a2a48..94261d19cceb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,6 +161,8 @@ extern bool snd_usb_skip_validation; * Ignore errors for mixer access * QUIRK_FLAG_DSD_RAW: * Support generic DSD raw U32_BE format + * QUIRK_FLAG_SET_IFACE_FIRST: + * Set up the interface at first like UAC1 */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -179,5 +181,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_DISABLE_AUTOSUSPEND (1U << 13) #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) +#define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) #endif /* __USBAUDIO_H */ -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: Fix regression on Sony WALKMAN NW-A45 DAC
by Takashi Iwai 24 Aug '21

24 Aug '21

We've got a regression report for USB-audio with Sony WALKMAN NW-A45 DAC device where no sound is audible on recent kernel. The bisection resulted in the code change wrt endpoint management, and the further debug session revealed that it was caused by the order of the USB audio interface. In the earlier code, we always set up the USB interface at first before other setups, but it was changed to be done at the last for UAC2/3, which is more standard way, while keeping the old way for UAC1. OTOH, this device seems requiring the setup of the interface at first just like UAC1. This patch works around the regression by applying the interface setup specifically for the WALKMAN at the beginning of the endpoint setup procedure. This change is written straightforwardly to be easily backported in old kernels. A further cleanup to move the workaround into a generic quirk section will follow in a later patch. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214105 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d33685874a6b..e6d58d7674a0 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1293,6 +1293,11 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to be set up before parameter setups */ iface_first = ep->cur_audiofmt->protocol == UAC_VERSION_1; + /* Workaround for Sony WALKMAN NW-A45 DAC; + * it requires the interface setup at first like UAC1 + */ + if (chip->usb_id == USB_ID(0x054c, 0x0b8c)) + iface_first = true; if (iface_first) { err = endpoint_set_interface(chip, ep, true); if (err < 0) -- 2.31.1

1 0

FAILED: Patch "ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8" failed to apply to 4.4-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.4-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From 8903376dc69949199301b290cc22dc64ae5d8a6d Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Wed, 18 Aug 2021 22:41:18 +0800 Subject: [PATCH] ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8 The mic has lots of noises if mic boost is enabled. So disable mic boost to get crystal clear audio capture. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96f32eaa24df..7ad689f991e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6658,6 +6658,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8438,8 +8445,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop" failed to apply to 4.4-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.4-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From da94692001ea45ffa1f5e9f17ecdef7aecd90c27 Mon Sep 17 00:00:00 2001 From: Kristin Paget <kristin(a)tombom.co.uk> Date: Sat, 14 Aug 2021 15:46:05 -0700 Subject: [PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk 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 a065260d0d20..96f32eaa24df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8332,6 +8332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8" failed to apply to 4.9-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.9-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From 8903376dc69949199301b290cc22dc64ae5d8a6d Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Wed, 18 Aug 2021 22:41:18 +0800 Subject: [PATCH] ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8 The mic has lots of noises if mic boost is enabled. So disable mic boost to get crystal clear audio capture. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96f32eaa24df..7ad689f991e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6658,6 +6658,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8438,8 +8445,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop" failed to apply to 4.9-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.9-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From da94692001ea45ffa1f5e9f17ecdef7aecd90c27 Mon Sep 17 00:00:00 2001 From: Kristin Paget <kristin(a)tombom.co.uk> Date: Sat, 14 Aug 2021 15:46:05 -0700 Subject: [PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk 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 a065260d0d20..96f32eaa24df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8332,6 +8332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8" failed to apply to 4.14-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.14-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From 8903376dc69949199301b290cc22dc64ae5d8a6d Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Wed, 18 Aug 2021 22:41:18 +0800 Subject: [PATCH] ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8 The mic has lots of noises if mic boost is enabled. So disable mic boost to get crystal clear audio capture. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96f32eaa24df..7ad689f991e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6658,6 +6658,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8438,8 +8445,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop" failed to apply to 4.14-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.14-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From da94692001ea45ffa1f5e9f17ecdef7aecd90c27 Mon Sep 17 00:00:00 2001 From: Kristin Paget <kristin(a)tombom.co.uk> Date: Sat, 14 Aug 2021 15:46:05 -0700 Subject: [PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk 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 a065260d0d20..96f32eaa24df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8332,6 +8332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8" failed to apply to 4.19-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.19-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From 8903376dc69949199301b290cc22dc64ae5d8a6d Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Wed, 18 Aug 2021 22:41:18 +0800 Subject: [PATCH] ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8 The mic has lots of noises if mic boost is enabled. So disable mic boost to get crystal clear audio capture. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96f32eaa24df..7ad689f991e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6658,6 +6658,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8438,8 +8445,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop" failed to apply to 4.19-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.19-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From da94692001ea45ffa1f5e9f17ecdef7aecd90c27 Mon Sep 17 00:00:00 2001 From: Kristin Paget <kristin(a)tombom.co.uk> Date: Sat, 14 Aug 2021 15:46:05 -0700 Subject: [PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk 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 a065260d0d20..96f32eaa24df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8332,6 +8332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.30.2

1 0

FAILED: Patch "slimbus: ngd: set correct device for pm" failed to apply to 4.19-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 4.19-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From c0e38eaa8d5102c138e4f16658ea762417d42a8f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Mon, 9 Aug 2021 09:24:27 +0100 Subject: [PATCH] slimbus: ngd: set correct device for pm For some reason we ended up using wrong device in some places for pm_runtime calls. Fix this so that NGG driver can do runtime pm correctly. Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210809082428.11236-4-srinivas.kandagatla@linaro… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..f3ee8e036372 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -618,7 +618,7 @@ static void qcom_slim_ngd_rx(struct qcom_slim_ngd_ctrl *ctrl, u8 *buf) (mc == SLIM_USR_MC_GENERIC_ACK && mt == SLIM_MSG_MT_SRC_REFERRED_USER)) { slim_msg_response(&ctrl->ctrl, &buf[4], buf[3], len - 4); - pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_mark_last_busy(ctrl->ctrl.dev); } } @@ -1257,13 +1257,14 @@ static int qcom_slim_ngd_enable(struct qcom_slim_ngd_ctrl *ctrl, bool enable) } /* controller state should be in sync with framework state */ complete(&ctrl->qmi.qmi_comp); - if (!pm_runtime_enabled(ctrl->dev) || - !pm_runtime_suspended(ctrl->dev)) - qcom_slim_ngd_runtime_resume(ctrl->dev); + if (!pm_runtime_enabled(ctrl->ctrl.dev) || + !pm_runtime_suspended(ctrl->ctrl.dev)) + qcom_slim_ngd_runtime_resume(ctrl->ctrl.dev); else - pm_runtime_resume(ctrl->dev); - pm_runtime_mark_last_busy(ctrl->dev); - pm_runtime_put(ctrl->dev); + pm_runtime_resume(ctrl->ctrl.dev); + + pm_runtime_mark_last_busy(ctrl->ctrl.dev); + pm_runtime_put(ctrl->ctrl.dev); ret = slim_register_controller(&ctrl->ctrl); if (ret) { @@ -1389,7 +1390,7 @@ static int qcom_slim_ngd_ssr_pdr_notify(struct qcom_slim_ngd_ctrl *ctrl, /* Make sure the last dma xfer is finished */ mutex_lock(&ctrl->tx_lock); if (ctrl->state != QCOM_SLIM_NGD_CTRL_DOWN) { - pm_runtime_get_noresume(ctrl->dev); + pm_runtime_get_noresume(ctrl->ctrl.dev); ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; qcom_slim_ngd_down(ctrl); qcom_slim_ngd_exit_dma(ctrl); -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8" failed to apply to 5.4-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 5.4-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From 8903376dc69949199301b290cc22dc64ae5d8a6d Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Date: Wed, 18 Aug 2021 22:41:18 +0800 Subject: [PATCH] ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8 The mic has lots of noises if mic boost is enabled. So disable mic boost to get crystal clear audio capture. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96f32eaa24df..7ad689f991e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6658,6 +6658,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8438,8 +8445,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.30.2

1 0

FAILED: Patch "ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop" failed to apply to 5.4-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 5.4-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From da94692001ea45ffa1f5e9f17ecdef7aecd90c27 Mon Sep 17 00:00:00 2001 From: Kristin Paget <kristin(a)tombom.co.uk> Date: Sat, 14 Aug 2021 15:46:05 -0700 Subject: [PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk 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 a065260d0d20..96f32eaa24df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8332,6 +8332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.30.2

1 0

FAILED: Patch "slimbus: ngd: set correct device for pm" failed to apply to 5.4-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 5.4-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From c0e38eaa8d5102c138e4f16658ea762417d42a8f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Mon, 9 Aug 2021 09:24:27 +0100 Subject: [PATCH] slimbus: ngd: set correct device for pm For some reason we ended up using wrong device in some places for pm_runtime calls. Fix this so that NGG driver can do runtime pm correctly. Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210809082428.11236-4-srinivas.kandagatla@linaro… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..f3ee8e036372 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -618,7 +618,7 @@ static void qcom_slim_ngd_rx(struct qcom_slim_ngd_ctrl *ctrl, u8 *buf) (mc == SLIM_USR_MC_GENERIC_ACK && mt == SLIM_MSG_MT_SRC_REFERRED_USER)) { slim_msg_response(&ctrl->ctrl, &buf[4], buf[3], len - 4); - pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_mark_last_busy(ctrl->ctrl.dev); } } @@ -1257,13 +1257,14 @@ static int qcom_slim_ngd_enable(struct qcom_slim_ngd_ctrl *ctrl, bool enable) } /* controller state should be in sync with framework state */ complete(&ctrl->qmi.qmi_comp); - if (!pm_runtime_enabled(ctrl->dev) || - !pm_runtime_suspended(ctrl->dev)) - qcom_slim_ngd_runtime_resume(ctrl->dev); + if (!pm_runtime_enabled(ctrl->ctrl.dev) || + !pm_runtime_suspended(ctrl->ctrl.dev)) + qcom_slim_ngd_runtime_resume(ctrl->ctrl.dev); else - pm_runtime_resume(ctrl->dev); - pm_runtime_mark_last_busy(ctrl->dev); - pm_runtime_put(ctrl->dev); + pm_runtime_resume(ctrl->ctrl.dev); + + pm_runtime_mark_last_busy(ctrl->ctrl.dev); + pm_runtime_put(ctrl->ctrl.dev); ret = slim_register_controller(&ctrl->ctrl); if (ret) { @@ -1389,7 +1390,7 @@ static int qcom_slim_ngd_ssr_pdr_notify(struct qcom_slim_ngd_ctrl *ctrl, /* Make sure the last dma xfer is finished */ mutex_lock(&ctrl->tx_lock); if (ctrl->state != QCOM_SLIM_NGD_CTRL_DOWN) { - pm_runtime_get_noresume(ctrl->dev); + pm_runtime_get_noresume(ctrl->ctrl.dev); ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; qcom_slim_ngd_down(ctrl); qcom_slim_ngd_exit_dma(ctrl); -- 2.30.2

1 0

FAILED: Patch "slimbus: ngd: set correct device for pm" failed to apply to 5.10-stable tree
by Sasha Levin 24 Aug '21

24 Aug '21

The patch below does not apply to the 5.10-stable tree. If someone wants it applied there, or to any other stable or longterm tree, then please email the backport, including the original git commit id to <stable(a)vger.kernel.org>. Thanks, Sasha ------------------ original commit in Linus's tree ------------------ >From c0e38eaa8d5102c138e4f16658ea762417d42a8f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Mon, 9 Aug 2021 09:24:27 +0100 Subject: [PATCH] slimbus: ngd: set correct device for pm For some reason we ended up using wrong device in some places for pm_runtime calls. Fix this so that NGG driver can do runtime pm correctly. Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20210809082428.11236-4-srinivas.kandagatla@linaro… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..f3ee8e036372 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -618,7 +618,7 @@ static void qcom_slim_ngd_rx(struct qcom_slim_ngd_ctrl *ctrl, u8 *buf) (mc == SLIM_USR_MC_GENERIC_ACK && mt == SLIM_MSG_MT_SRC_REFERRED_USER)) { slim_msg_response(&ctrl->ctrl, &buf[4], buf[3], len - 4); - pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_mark_last_busy(ctrl->ctrl.dev); } } @@ -1257,13 +1257,14 @@ static int qcom_slim_ngd_enable(struct qcom_slim_ngd_ctrl *ctrl, bool enable) } /* controller state should be in sync with framework state */ complete(&ctrl->qmi.qmi_comp); - if (!pm_runtime_enabled(ctrl->dev) || - !pm_runtime_suspended(ctrl->dev)) - qcom_slim_ngd_runtime_resume(ctrl->dev); + if (!pm_runtime_enabled(ctrl->ctrl.dev) || + !pm_runtime_suspended(ctrl->ctrl.dev)) + qcom_slim_ngd_runtime_resume(ctrl->ctrl.dev); else - pm_runtime_resume(ctrl->dev); - pm_runtime_mark_last_busy(ctrl->dev); - pm_runtime_put(ctrl->dev); + pm_runtime_resume(ctrl->ctrl.dev); + + pm_runtime_mark_last_busy(ctrl->ctrl.dev); + pm_runtime_put(ctrl->ctrl.dev); ret = slim_register_controller(&ctrl->ctrl); if (ret) { @@ -1389,7 +1390,7 @@ static int qcom_slim_ngd_ssr_pdr_notify(struct qcom_slim_ngd_ctrl *ctrl, /* Make sure the last dma xfer is finished */ mutex_lock(&ctrl->tx_lock); if (ctrl->state != QCOM_SLIM_NGD_CTRL_DOWN) { - pm_runtime_get_noresume(ctrl->dev); + pm_runtime_get_noresume(ctrl->ctrl.dev); ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; qcom_slim_ngd_down(ctrl); qcom_slim_ngd_exit_dma(ctrl); -- 2.30.2

1 0

[PATCH v1 0/15] Patches to update for rockchip i2s
by Sugar Zhang 24 Aug '21

24 Aug '21

These patches fixup or update for rockchip i2s. Sugar Zhang (13): ASoC: rockchip: i2s: Add support for set bclk ratio ASoC: rockchip: i2s: Fixup clk div error ASoC: rockchip: i2s: Improve dma data transfer efficiency ASoC: rockchip: i2s: Fix regmap_ops hang ASoC: rockchip: i2s: Fix concurrency between tx/rx ASoC: rockchip: i2s: Reset the controller if soft reset failed ASoC: dt-bindings: rockchip: Document reset property for i2s ASoC: rockchip: i2s: Add property to specify play/cap capability ASoC: dt-bindings: rockchip: i2s: Document property for playback/capture ASoC: rockchip: i2s: Add compatible for more SoCs ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs ASoC: rockchip: i2s: Add support for frame inversion ASoC: dt-bindings: rockchip: i2s: Document property 'clk-trcm' Xiaotan Luo (1): ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B Xing Zheng (1): ASoC: rockchip: i2s: Add support for 'rockchip,clk-trcm' property .../devicetree/bindings/sound/rockchip-i2s.yaml | 30 ++++ sound/soc/rockchip/rockchip_i2s.c | 153 ++++++++++++++++----- sound/soc/rockchip/rockchip_i2s.h | 10 +- 3 files changed, 157 insertions(+), 36 deletions(-) -- 2.7.4

3 20

Out of bounds
by GitHub pull_request - opened 23 Aug '21

23 Aug '21

alsa-project/alsa-utils pull request #107 was opened from folkertvanheusden: Drevent out-of-bounds writing in arecord during recording with high volume and vu-meter enabled. Request URL : https://github.com/alsa-project/alsa-utils/pull/107 Patch URL : https://github.com/alsa-project/alsa-utils/pull/107.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: rt1015: remove possible unused variable `bclk_ms'
by Tzung-Bi Shih 23 Aug '21

23 Aug '21

bclk_ms is possible unused by using the given config (see [1]). sound/soc/codecs/rt1015.c:724:2: warning: Value stored to 'bclk_ms' is never read [clang-analyzer-deadcode.DeadStores] bclk_ms = frame_size > 32; ^ ~~~~~~~~~~~~~~~ In addition, bclk_ms is meaningless and confusing after applying commit a5db2ca51367 ("ASoC: rt1015: remove unneeded variables in rt1015_priv"). The "> 32" in the expression looks like a typo but it was not. Let's remove the confusing variable bclk_ms. [1]: https://lkml.org/lkml/2021/8/23/305 Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- For the other two reported files: tas2770.c and wm8731.c, it seems there is no straightforward way to fix the warning. sound/soc/codecs/rt1015.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index b3e69bf245dd..c0c5952cdff7 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -689,7 +689,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); - int pre_div, bclk_ms, frame_size, lrck; + int pre_div, frame_size, lrck; unsigned int val_len = 0; lrck = params_rate(params); @@ -706,10 +706,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_ms = frame_size > 32; - - dev_dbg(component->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", - bclk_ms, pre_div, dai->id); + dev_dbg(component->dev, "pre_div is %d for iis %d\n", pre_div, dai->id); dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d\n", lrck, pre_div, dai->id); -- 2.33.0.rc2.250.ged5fa647cd-goog

2 1

[PATCH 00/11] ASoC: Intel: Skylake: Fix and support complex
by Cezary Rojewski 23 Aug '21

23 Aug '21

Existing skylake-driver supports very basic scenarios with limited range of modules and their control. Attached changes first fix code as several advanced configurations are 'mentioned' throughout the files but are not actually functional. Follow up are changes adding missing support for said configurations. Cezary Rojewski (5): ASoC: Intel: kbl_da7219_max98927: Fix format selection for max98373 ASoC: Intel: Skylake: Leave data as is when invoking TLV IPCs ASoC: Intel: Skylake: Fix module resource and format selection ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER ASoC: Intel: Skylake: Select first entry for singular pipe config arrays Gustaw Lewandowski (2): ASoC: Intel: Skylake: Fix passing loadable flag for module ASoC: Intel: Skylake: Simplify m_state for loadable modules Kareem Shaik (1): ASoC: Intel: Skylake: Support multiple format configs Pawel Harlozinski (1): ASoC: Intel: Skylake: Properly configure modules with generic extension Piotr Maziarz (1): ASoC: Intel: Skylake: Select proper format for NHLT blob Szymon Mielczarek (1): ASoC: Intel: Skylake: Support modules with generic extension include/uapi/sound/snd_sst_tokens.h | 6 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 55 +------ sound/soc/intel/skylake/skl-messages.c | 155 ++++++++++++------- sound/soc/intel/skylake/skl-pcm.c | 25 ++- sound/soc/intel/skylake/skl-topology.c | 155 +++++++++++-------- sound/soc/intel/skylake/skl-topology.h | 26 +++- 6 files changed, 231 insertions(+), 191 deletions(-) -- 2.25.1

3 13

[PATCH] ASoC: Intel: bytcr_rt5640: Mark hp_elitepad_1000g2_jack?_check functions static
by Hans de Goede 23 Aug '21

23 Aug '21

The byt_rt5640_hp_elitepad_1000g2_jack?_check functions are only used inside bytcr_rt5640.c, mark them as static. Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2f151ba80be5..05667a33d919 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -490,7 +490,7 @@ static const struct acpi_gpio_mapping byt_rt5640_hp_elitepad_1000g2_gpios[] = { { }, }; -int byt_rt5640_hp_elitepad_1000g2_jack1_check(void *data) +static int byt_rt5640_hp_elitepad_1000g2_jack1_check(void *data) { struct byt_rt5640_private *priv = data; int jack_status, mic_status; @@ -506,7 +506,7 @@ int byt_rt5640_hp_elitepad_1000g2_jack1_check(void *data) return SND_JACK_HEADSET; } -int byt_rt5640_hp_elitepad_1000g2_jack2_check(void *data) +static int byt_rt5640_hp_elitepad_1000g2_jack2_check(void *data) { struct snd_soc_component *component = data; int jack_status, report; -- 2.31.1

2 1

[RESEND 1/3] ASoC: ics43432: add CMM-4030D-261 support
by Angelo Dureghello 23 Aug '21

23 Aug '21

Despite minimal datasheet differences, the driver seems perfectly compatible with CMM-4030D-261. Tested CMM-4030D-261 to work with this changes: / { cmm4030d: cmm4030d { #sound-dai-cells = <0>; compatible = "cui,cmm-4030d-261"; }; sound_mic { compatible = "simple-audio-card"; simple-audio-card,name = "i2s mem mic"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_master>; simple-audio-card,frame-master = <&sound_master>; sound_master: simple-audio-card,cpu { sound-dai = <&ssi2>; system-clock-frequency = <2822400>; }; simple-audio-card,codec { sound-dai = <&cmm4030d>; }; }; }; Audio has been captured and tested successfully by: arecord -D "hw:1,0" -f S24_LE > test.wav aplay test.wav Signed-off-by: Angelo Dureghello <angelo.dureghello(a)timesys.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/ics43432.c | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index db16071205ba..cd1976aae328 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -847,7 +847,7 @@ config SND_SOC_HDAC_HDA select SND_HDA config SND_SOC_ICS43432 - tristate + tristate "ICS43423 and compatible i2s microphones" config SND_SOC_INNO_RK3036 tristate "Inno codec driver for RK3036 SoC" diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 47e749f03940..de4c8460ab3d 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * I2S MEMS microphone driver for InvenSense ICS-43432 + * I2S MEMS microphone driver for InvenSense ICS-43432 and similar + * MEMS-based microphones. * * - Non configurable. * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data @@ -53,6 +54,7 @@ static int ics43432_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id ics43432_ids[] = { { .compatible = "invensense,ics43432", }, + { .compatible = "cui,cmm-4030d-261", }, { } }; MODULE_DEVICE_TABLE(of, ics43432_ids); -- 2.32.0

2 5

[PATCH 2/2] plugio: Check for pointer callback error codes
by Arkadiusz Bokowy 23 Aug '21

23 Aug '21

By checking error code returned by the pointer callback, we can determine more precisely PCM state. Previous implementation assumed, that a software PCM can only produce overrun or underrun. It was impossible to mark software PCM as disconnected. Signed-off-by: Arkadiusz Bokowy <arkadiusz.bokowy(a)gmail.com> --- src/pcm/pcm_ioplug.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index c96104e9..0e27e121 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -81,11 +81,21 @@ static int snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) } io->last_hw = (snd_pcm_uframes_t)hw; } else { + switch (hw) { + case -ESTRPIPE: + io->data->state = SND_PCM_STATE_SUSPENDED; + break; + case -ENODEV: + io->data->state = SND_PCM_STATE_DISCONNECTED; + break; + default: + io->data->state = SND_PCM_STATE_XRUN; + } if (io->data->state == SND_PCM_STATE_DRAINING) snd_pcm_ioplug_drop(pcm); else io->data->state = SNDRV_PCM_STATE_XRUN; - return -EPIPE; + return hw; } return 0; } -- 2.31.1

2 1

[PATCH 1/2] ioplug: Check for callback error codes
by Arkadiusz Bokowy 23 Aug '21

23 Aug '21

Signed-off-by: Arkadiusz Bokowy <arkadiusz.bokowy(a)gmail.com> --- src/pcm/pcm_ioplug.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index 98184398..c96104e9 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -54,7 +54,7 @@ static int snd_pcm_ioplug_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsi /* update the hw pointer */ /* called in lock */ -static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) +static int snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) { ioplug_priv_t *io = pcm->private_data; snd_pcm_sframes_t hw; @@ -85,7 +85,9 @@ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) snd_pcm_ioplug_drop(pcm); else io->data->state = SNDRV_PCM_STATE_XRUN; + return -EPIPE; } + return 0; } static int snd_pcm_ioplug_info(snd_pcm_t *pcm, snd_pcm_info_t *info) @@ -115,7 +117,10 @@ static int snd_pcm_ioplug_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) io->data->callback->delay) return io->data->callback->delay(io->data, delayp); else { - snd_pcm_ioplug_hw_ptr_update(pcm); + int err; + err = snd_pcm_ioplug_hw_ptr_update(pcm); + if (err < 0) + return err; *delayp = snd_pcm_mmap_delay(pcm); } return 0; @@ -499,11 +504,14 @@ static int snd_pcm_ioplug_start(snd_pcm_t *pcm) static int snd_pcm_ioplug_drop(snd_pcm_t *pcm) { ioplug_priv_t *io = pcm->private_data; + int err; if (io->data->state == SND_PCM_STATE_OPEN) return -EBADFD; - io->data->callback->stop(io->data); + err = io->data->callback->stop(io->data); + if (err < 0) + return err; gettimestamp(&io->trigger_tstamp, pcm->tstamp_type); io->data->state = SND_PCM_STATE_SETUP; @@ -625,7 +633,7 @@ static int snd_pcm_ioplug_resume(snd_pcm_t *pcm) ioplug_priv_t *io = pcm->private_data; if (io->data->callback->resume) - io->data->callback->resume(io->data); + return io->data->callback->resume(io->data); return 0; } @@ -898,13 +906,14 @@ static void clear_io_params(ioplug_priv_t *io) static int snd_pcm_ioplug_close(snd_pcm_t *pcm) { ioplug_priv_t *io = pcm->private_data; + int err = 0; clear_io_params(io); if (io->data->callback->close) - io->data->callback->close(io->data); + err = io->data->callback->close(io->data); free(io); - return 0; + return err; } static const snd_pcm_ops_t snd_pcm_ioplug_ops = { -- 2.31.1

2 1

[PATCH] soundwire: cadence: do not extend reset delay
by Bard Liao 23 Aug '21

23 Aug '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The duration of the hw_reset is defined as 4096 cycles. The Cadence IP allows for an additional delay which doesn't seem necessary in practice: the actual reset sequence duration is defined by the sync_go mechanism, not by the IP itself. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 0b7f037e6cd0..4fcc3ba93004 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1032,10 +1032,7 @@ EXPORT_SYMBOL(sdw_cdns_check_self_clearing_bits); */ int sdw_cdns_exit_reset(struct sdw_cdns *cdns) { - /* program maximum length reset to be safe */ - cdns_updatel(cdns, CDNS_MCP_CONTROL, - CDNS_MCP_CONTROL_RST_DELAY, - CDNS_MCP_CONTROL_RST_DELAY); + /* keep reset delay unchanged to 4096 cycles */ /* use hardware generated reset */ cdns_updatel(cdns, CDNS_MCP_CONTROL, -- 2.17.1

2 1

[PATCH v2 0/3] soundwire: intel: exit clock-stop mode before system suspend
by Bard Liao 23 Aug '21

23 Aug '21

Intel validation reported an issue where the HW_RST self-clearing bit is not cleared in hardware, which as a ripple effect creates issues with the clock stop mode. This happens is a specific sequence where the Intel manager is pm_runtime suspended with the clock-stop mode enabled. During the system suspend, we currently do nothing, which can lead to potential issues on system resume and the following pm_runtime suspend, depending on the hardware state. This patch suggests a full resume if the clock-stop mode is used. This may require extra time but will make the suspend/resume flows completely symmetric. This also removes a race condition where we could not access SHIM registers if the parent was suspended as well. Resuming the link also resumes the parent by construction. BugLink: https://github.com/thesofproject/linux/issues/2606 v2: - Better comments and commit messages. - Modified the .prepare callback to only deal with the corner case that is NOT covered today instead of systematically doing a full resume. Pierre-Louis Bossart (3): soundwire: intel: fix potential race condition during power down soundwire: intel: skip suspend/resume/wake when link was not started soundwire: intel: conditionally exit clock stop mode on system suspend drivers/soundwire/intel.c | 150 ++++++++++++++++++++++++++++++-------- drivers/soundwire/intel.h | 1 + 2 files changed, 119 insertions(+), 32 deletions(-) -- 2.17.1

2 4

[PATCH for-5.15] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Takashi Iwai 23 Aug '21

23 Aug '21

We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- The patch is to be applied on my for-next branch once after merging for-linus branch. sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 905c7965f653..5db2f4865bbb 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.26.2

1 1

[PATCH] ALSA: doc: Fix indentation warning
by Takashi Iwai 23 Aug '21

23 Aug '21

Fix a trivial warning for the indentation by putting an empty line: Documentation/sound/alsa-configuration.rst:2258: WARNING: Unexpected indentation. Fixes: a39978ed6df1 ("ALSA: doc: Add the description of quirk_flags option for snd-usb-audio") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index e61edd1295fc..e3c33b8d6d78 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2261,6 +2261,7 @@ delayed_register quirk_flags Contains the bit flags for various device specific workarounds. Applied to the corresponding card index. + * bit 0: Skip reading sample rate for devices * bit 1: Create Media Controller API entries * bit 2: Allow alignment on audio sub-slot at transfer -- 2.31.1

1 0

[PATCH] ALSA: firewire-motu: add support for MOTU 896HD
by Takashi Sakamoto 23 Aug '21

23 Aug '21

Mark of the Unicorn (MOTU) shipped 896HD 2003 as one of models in second generation of its FireWire series, and already discontinued it. The model consists of below ICs: * Texas Instruments TSB41AB2 * Phillips Semiconductors PDI1394L40 * Altera cyclone EP1C3 * Texas Instruments TMS320VC5402 It supports sampling transmission frequency up to 192.0 kHz. The packet format differs depending on both of sampling transfer frequency and enabling ADAT channels. The model doesn't support MIDI message transmission. This commit adds support for it. $ python3 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04101b66 bus_info_length 4, crc_length 16, crc 7014 404 31333934 bus_name "1394" 408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4) 40c 0001f200 company_id 0001f2 | 410 0001dbce device_id 000001dbce | EUI-64 0001f2000001dbce root directory ----------------------------------------------------------------- 414 0004c65c directory_length 4, crc 50780 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 0003dcc1 directory_length 3, crc 56513 42c 120001f2 specifier id 430 13000005 version 434 17102800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 000264f2 leaf_length 2, crc 25842 43c 0001f200 company_id 0001f2 | 440 0001dbce device_id 000001dbce | EUI-64 0001f2000001dbce Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v2.c | 22 ++++++++++++++++++---- sound/firewire/motu/motu.c | 1 + sound/firewire/motu/motu.h | 1 + 3 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 93d5df1ae550..2bd4485e4bc7 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -12,11 +12,11 @@ #define V2_CLOCK_RATE_SHIFT 3 #define V2_CLOCK_SRC_MASK 0x00000007 #define V2_CLOCK_SRC_SHIFT 0 -#define V2_CLOCK_SRC_AESEBU_ON_XLR 0x07 +#define V2_CLOCK_SRC_AESEBU_ON_XLR 0x07 // In Traveler. #define V2_CLOCK_SRC_ADAT_ON_DSUB 0x05 #define V2_CLOCK_SRC_WORD_ON_BNC 0x04 #define V2_CLOCK_SRC_SPH 0x03 -#define V2_CLOCK_SRC_SPDIF 0x02 // on either coaxial or optical +#define V2_CLOCK_SRC_SPDIF 0x02 // on either coaxial or optical. AES/EBU in 896HD. #define V2_CLOCK_SRC_ADAT_ON_OPT 0x01 #define V2_CLOCK_SRC_INTERNAL 0x00 #define V2_CLOCK_FETCH_ENABLE 0x02000000 @@ -100,7 +100,9 @@ static int get_clock_source(struct snd_motu *motu, u32 data, bool support_iec60958_on_opt = (motu->spec == &snd_motu_spec_828mk2 || motu->spec == &snd_motu_spec_traveler); - if (!support_iec60958_on_opt) { + if (motu->spec == &snd_motu_spec_896hd) { + *src = SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR; + } else if (!support_iec60958_on_opt) { *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; } else { __be32 reg; @@ -129,6 +131,7 @@ static int get_clock_source(struct snd_motu *motu, u32 data, *src = SND_MOTU_CLOCK_SOURCE_ADAT_ON_DSUB; break; case V2_CLOCK_SRC_AESEBU_ON_XLR: + // For Traveler. *src = SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR; break; default: @@ -153,7 +156,7 @@ int snd_motu_protocol_v2_get_clock_source(struct snd_motu *motu, return get_clock_source(motu, be32_to_cpu(reg), src); } -// Expected for Traveler and 896HD, which implements Altera Cyclone EP1C3. +// Expected for Traveler, which implements Altera Cyclone EP1C3. static int switch_fetching_mode_cyclone(struct snd_motu *motu, u32 *data, bool enable) { @@ -190,6 +193,9 @@ int snd_motu_protocol_v2_switch_fetching_mode(struct snd_motu *motu, if (motu->spec == &snd_motu_spec_828mk2) { // 828mkII implements Altera ACEX 1K EP1K30. Nothing to do. return 0; + } else if (motu->spec == &snd_motu_spec_896hd) { + // 896HD implements Altera Cyclone EP1C3 but nothing to do. + return 0; } else { __be32 reg; u32 data; @@ -274,6 +280,14 @@ const struct snd_motu_spec snd_motu_spec_828mk2 = { .rx_fixed_pcm_chunks = {14, 14, 0}, }; +const struct snd_motu_spec snd_motu_spec_896hd = { + .name = "896HD", + .protocol_version = SND_MOTU_PROTOCOL_V2, + // No support for MIDI. + .tx_fixed_pcm_chunks = {14, 14, 8}, + .rx_fixed_pcm_chunks = {14, 14, 8}, +}; + const struct snd_motu_spec snd_motu_spec_traveler = { .name = "Traveler", .protocol_version = SND_MOTU_PROTOCOL_V2, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 543136578c70..f65426238d4c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -153,6 +153,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000001, &snd_motu_spec_828), SND_MOTU_DEV_ENTRY(0x000002, &snd_motu_spec_896), SND_MOTU_DEV_ENTRY(0x000003, &snd_motu_spec_828mk2), + SND_MOTU_DEV_ENTRY(0x000005, &snd_motu_spec_896hd), SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 73f36d1be515..f1a830b358d4 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -126,6 +126,7 @@ extern const struct snd_motu_spec snd_motu_spec_828; extern const struct snd_motu_spec snd_motu_spec_896; extern const struct snd_motu_spec snd_motu_spec_828mk2; +extern const struct snd_motu_spec snd_motu_spec_896hd; extern const struct snd_motu_spec snd_motu_spec_traveler; extern const struct snd_motu_spec snd_motu_spec_ultralite; extern const struct snd_motu_spec snd_motu_spec_8pre; -- 2.30.2

2 1

[PATCH 0/3] ALSA: hda: SSID aliasing via model option
by Takashi Iwai 23 Aug '21

23 Aug '21

Hi, this is a patch set for enhancement of HD-audio configuration. There are lots of cases where a quick is applied only with PCI or codec SSID matching but no corresponding model option entry is provided. In such a case, user had to modify and recompile the code, or using the early patching with patch option, both of which are cumbersome. This option makes it easier to apply a quirk of another device. Now you can pass model option string in a form of 1234:abcd for specifying the SSID of the alias target. Takashi === Takashi Iwai (3): ALSA: hda: Code refactoring snd_hda_pick_fixup() ALSA: hda: Allow model option to specify PCI SSID alias ALSA: hda: Update documentation for aliasing via the model option Documentation/sound/alsa-configuration.rst | 6 ++ Documentation/sound/hd-audio/notes.rst | 11 +++ sound/pci/hda/hda_auto_parser.c | 84 +++++++++++++--------- 3 files changed, 66 insertions(+), 35 deletions(-) -- 2.26.2

1 3

pulseaudio bug
by Callum Parker 23 Aug '21

23 Aug '21

Hello, I have been prompted by my logs that a bug is in the ALSA driver. [image.png] http://alsa-project.org/db/?f=a1c764daf38a6b532fa62ce3d069ca23a9896ac8 here is the upload I uploaded from the command distro I am using is Linux Mint20.2 Kind regards Callum Parker Sent with [ProtonMail](https://protonmail.com/) Secure Email.

1 0

OpenAL fails to build with recent "rawmidi" commit
by GitHub issues - edited 22 Aug '21

22 Aug '21

alsa-project/alsa-lib issue #172 was edited from shoober420: The recent alsa-lib commit https://github.com/alsa-project/alsa-lib/commit/95eb312fade1908a2c944e9de46… borks OpenAL from building. I thought it was an issue with OpenAL, but he mentions that its an issue with alsa-lib (https://github.com/kcat/openal-soft/issues/592#issuecomment-903211857). Here is the OpenAL build log just in case and a snippet of the OpenAL build error. https://github.com/kcat/openal-soft/files/7026546/buildopenal.log OpenAL build error. ``` [84/97] /usr/bin/x86_64-pc-linux-gnu-g++ -m32 -DALC_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_ALEXT_PROTOTYPES -DAL_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_BUILD_LIBRARY -DOpenAL_EXPORTS -DRESTRICT=__restrict -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/include -I/usr/include/pipewire-0.3 -I/usr/include/spa-0.2 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999_build-abi_x86_32.x86 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/common -march=native -Ofast -pipe -fno-plt -fno-common -fno-semantic-interposition -fipa-pta -falign-functions=32 -fdevirtualize-at-ltrans -fuse-linker-plugin -floop-nest-optimize -fgraphite-identity -fno-signed-zeros -fno-trapping-math -fassociative-math -freciprocal-math -fno-math-errno -ffinite-math-only -fno-rounding-math -fno-signaling-nans -fcx-limited-range -fexcess-precision=fast -w -flto=auto -fPIC -Winline -Wunused -Wall -Wextra -Wshadow -Wconversion -Wcast-align -Wpedantic -Wold-style-cast -Wnon-virtual-dtor -Woverloaded-virtual -fno-math-errno -fvisibility=hidden -msse2 -mfpmath=sse -pthread -std=gnu++14 -MD -MT CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -MF CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o.d -o CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -c /var/tmp/portage/media-libs/openal-9999/work/openal-9999/alc/backends/alsa.cpp FAILED: CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o /usr/bin/x86_64-pc-linux-gnu-g++ -m32 -DALC_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_ALEXT_PROTOTYPES -DAL_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_BUILD_LIBRARY -DOpenAL_EXPORTS -DRESTRICT=__restrict -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/include -I/usr/include/pipewire-0.3 -I/usr/include/spa-0.2 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999_build-abi_x86_32.x86 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/common -march=native -Ofast -pipe -fno-plt -fno-common -fno-semantic-interposition -fipa-pta -falign-functions=32 -fdevirtualize-at-ltrans -fuse-linker-plugin -floop-nest-optimize -fgraphite-identity -fno-signed-zeros -fno-trapping-math -fassociative-math -freciprocal-math -fno-math-errno -ffinite-math-only -fno-rounding-math -fno-signaling-nans -fcx-limited-range -fexcess-precision=fast -w -flto=auto -fPIC -Winline -Wunused -Wall -Wextra -Wshadow -Wconversion -Wcast-align -Wpedantic -Wold-style-cast -Wnon-virtual-dtor -Woverloaded-virtual -fno-math-errno -fvisibility=hidden -msse2 -mfpmath=sse -pthread -std=gnu++14 -MD -MT CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -MF CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o.d -o CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -c /var/tmp/portage/media-libs/openal-9999/work/openal-9999/alc/backends/alsa.cpp In file included from /usr/include/alsa/asoundlib.h:55, from /var/tmp/portage/media-libs/openal-9999/work/openal-9999/alc/backends/alsa.cpp:51: /usr/include/alsa/rawmidi.h:107:9: error: '__u8' does not name a type 107 | __u8 frame_type; | ^~~~ /usr/include/alsa/rawmidi.h:108:9: error: '__u8' does not name a type 108 | __u8 length; /* number of valid bytes in data field */ | ^~~~ /usr/include/alsa/rawmidi.h:109:9: error: '__u8' does not name a type 109 | __u8 reserved[2]; | ^~~~ /usr/include/alsa/rawmidi.h:110:9: error: '__u32' does not name a type; did you mean '__m32'? 110 | __u32 tv_nsec; /* nanoseconds */ | ^~~~~ | __m32 /usr/include/alsa/rawmidi.h:111:9: error: '__u64' does not name a type; did you mean '__m64'? 111 | __u64 tv_sec; /* seconds */ | ^~~~~ | __m64 /usr/include/alsa/rawmidi.h:112:9: error: '__u8' does not name a type 112 | __u8 data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; | ^~~~ ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/172 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

OpenAL fails to build with https://github.com/alsa-project/alsa-lib/commit/95eb312fade1908a2c944e9de4626c0ff60b2203 commit
by GitHub issues - opened 22 Aug '21

22 Aug '21

alsa-project/alsa-lib issue #172 was opened from shoober420: The recent alsa-lib commit https://github.com/alsa-project/alsa-lib/commit/95eb312fade1908a2c944e9de46… borks OpenAL from building. I thought it was an issue with OpenAL, but he mentions that its an issue with alsa-lib (https://github.com/kcat/openal-soft/issues/592#issuecomment-903211857). Here is the OpenAL build log just in case and a snippet of the OpenAL build error. https://github.com/kcat/openal-soft/files/7026546/buildopenal.log OpenAL build error. ``` [84/97] /usr/bin/x86_64-pc-linux-gnu-g++ -m32 -DALC_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_ALEXT_PROTOTYPES -DAL_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_BUILD_LIBRARY -DOpenAL_EXPORTS -DRESTRICT=__restrict -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/include -I/usr/include/pipewire-0.3 -I/usr/include/spa-0.2 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999_build-abi_x86_32.x86 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/common -march=native -Ofast -pipe -fno-plt -fno-common -fno-semantic-interposition -fipa-pta -falign-functions=32 -fdevirtualize-at-ltrans -fuse-linker-plugin -floop-nest-optimize -fgraphite-identity -fno-signed-zeros -fno-trapping-math -fassociative-math -freciprocal-math -fno-math-errno -ffinite-math-only -fno-rounding-math -fno-signaling-nans -fcx-limited-range -fexcess-precision=fast -w -flto=auto -fPIC -Winline -Wunused -Wall -Wextra -Wshadow -Wconversion -Wcast-align -Wpedantic -Wold-style-cast -Wnon-virtual-dtor -Woverloaded-virtual -fno-math-errno -fvisibility=hidden -msse2 -mfpmath=sse -pthread -std=gnu++14 -MD -MT CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -MF CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o.d -o CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -c /var/tmp/portage/media-libs/openal-9999/work/openal-9999/alc/backends/alsa.cpp FAILED: CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o /usr/bin/x86_64-pc-linux-gnu-g++ -m32 -DALC_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_ALEXT_PROTOTYPES -DAL_API="__attribute__((visibility(\"protected\"))) __attribute__((force_align_arg_pointer))" -DAL_BUILD_LIBRARY -DOpenAL_EXPORTS -DRESTRICT=__restrict -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/include -I/usr/include/pipewire-0.3 -I/usr/include/spa-0.2 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999_build-abi_x86_32.x86 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999 -I/var/tmp/portage/media-libs/openal-9999/work/openal-9999/common -march=native -Ofast -pipe -fno-plt -fno-common -fno-semantic-interposition -fipa-pta -falign-functions=32 -fdevirtualize-at-ltrans -fuse-linker-plugin -floop-nest-optimize -fgraphite-identity -fno-signed-zeros -fno-trapping-math -fassociative-math -freciprocal-math -fno-math-errno -ffinite-math-only -fno-rounding-math -fno-signaling-nans -fcx-limited-range -fexcess-precision=fast -w -flto=auto -fPIC -Winline -Wunused -Wall -Wextra -Wshadow -Wconversion -Wcast-align -Wpedantic -Wold-style-cast -Wnon-virtual-dtor -Woverloaded-virtual -fno-math-errno -fvisibility=hidden -msse2 -mfpmath=sse -pthread -std=gnu++14 -MD -MT CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -MF CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o.d -o CMakeFiles/OpenAL.dir/alc/backends/alsa.cpp.o -c /var/tmp/portage/media-libs/openal-9999/work/openal-9999/alc/backends/alsa.cpp In file included from /usr/include/alsa/asoundlib.h:55, from /var/tmp/portage/media-libs/openal-9999/work/openal-9999/alc/backends/alsa.cpp:51: /usr/include/alsa/rawmidi.h:107:9: error: '__u8' does not name a type 107 | __u8 frame_type; | ^~~~ /usr/include/alsa/rawmidi.h:108:9: error: '__u8' does not name a type 108 | __u8 length; /* number of valid bytes in data field */ | ^~~~ /usr/include/alsa/rawmidi.h:109:9: error: '__u8' does not name a type 109 | __u8 reserved[2]; | ^~~~ /usr/include/alsa/rawmidi.h:110:9: error: '__u32' does not name a type; did you mean '__m32'? 110 | __u32 tv_nsec; /* nanoseconds */ | ^~~~~ | __m32 /usr/include/alsa/rawmidi.h:111:9: error: '__u64' does not name a type; did you mean '__m64'? 111 | __u64 tv_sec; /* seconds */ | ^~~~~ | __m64 /usr/include/alsa/rawmidi.h:112:9: error: '__u8' does not name a type 112 | __u8 data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; | ^~~~ ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/172 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

HDA codec problem
by Randy Dunlap 21 Aug '21

21 Aug '21

Hi, I am having problems getting audio working on my computer: 00:1f.3 Audio device: Intel Corporation Device f1c8 I did an internet search that said that I would need 5.13 or later to have support for this device. openSUSE 15.3 kernel 5.3.18-59.19-default says: snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC233: line_outs=1 (0x1b/0x0/0x0/0x0/0x0) type:line snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 snd_hda_codec_realtek hdaudioC0D0: inputs: snd_hda_codec_realtek hdaudioC0D0: Mic=0x1a snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 but produces no sound output. Kernel 5.13.10 or 5.14-rc6 say: calling azx_driver_init+0x0/0x20 @ 1 snd_hda_intel 0000:00:1f.3: enabling device (0000 -> 0002) snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops) snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: last cmd=0x200f0000 snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x200f0000 snd_hda_intel 0000:00:1f.3: Codec #2 probe error; disabling it... snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to single_cmd mode: last cmd=0x000f0001 hdaudio hdaudioC0D0: no AFG or MFG node found snd_hda_intel 0000:00:1f.3: no codecs initialized and of course no sound output. Anyone have suggestions or has a fix for this already been posted? or maybe I am just doing something wrong? thanks. -- ~Randy

2 5

[RESEND 1/3] ASoC: ics43432: add CMM-4030D-261 support
by Angelo Dureghello 21 Aug '21

21 Aug '21

Despite minimal datasheet differences, the driver seems perfectly compatible with CMM-4030D-261. Tested CMM-4030D-261 to work with this changes: / { cmm4030d: cmm4030d { #sound-dai-cells = <0>; compatible = "cui,cmm-4030d-261"; }; sound_mic { compatible = "simple-audio-card"; simple-audio-card,name = "i2s mem mic"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_master>; simple-audio-card,frame-master = <&sound_master>; sound_master: simple-audio-card,cpu { sound-dai = <&ssi2>; system-clock-frequency = <2822400>; }; simple-audio-card,codec { sound-dai = <&cmm4030d>; }; }; }; Audio has been captured and tested successfully by: arecord -D "hw:1,0" -f S24_LE > test.wav aplay test.wav Signed-off-by: Angelo Dureghello <angelo.dureghello(a)timesys.com> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/ics43432.c | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index db16071205ba..cd1976aae328 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -847,7 +847,7 @@ config SND_SOC_HDAC_HDA select SND_HDA config SND_SOC_ICS43432 - tristate + tristate "ICS43423 and compatible i2s microphones" config SND_SOC_INNO_RK3036 tristate "Inno codec driver for RK3036 SoC" diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 47e749f03940..de4c8460ab3d 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * I2S MEMS microphone driver for InvenSense ICS-43432 + * I2S MEMS microphone driver for InvenSense ICS-43432 and similar + * MEMS-based microphones. * * - Non configurable. * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data @@ -53,6 +54,7 @@ static int ics43432_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id ics43432_ids[] = { { .compatible = "invensense,ics43432", }, + { .compatible = "cui,cmm-4030d-261", }, { } }; MODULE_DEVICE_TABLE(of, ics43432_ids); -- 2.32.0

1 2

(.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here
by GitHub issues - edited 20 Aug '21

20 Aug '21

alsa-project/alsa-lib issue #171 was edited from shoober420: Trying to compile alsa-lib from the latest two commits as of Aug 20th (https://github.com/alsa-project/alsa-lib/commit/868ecd509deb05938107faf6657…) (https://github.com/alsa-project/alsa-lib/commit/95eb312fade1908a2c944e9de46…), I get these errors during build. Im using Gentoo Linux with GCC 11.2 version. [buildalsalib.log](https://github.com/alsa-project/alsa-lib/files/7023522/bu… ``` (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: parser.o (symbol from plugin): in function `uc_mgr_config_load_file': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_cond.o (symbol from plugin): in function `uc_mgr_evaluate_condition': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_subs.o (symbol from plugin): in function `uc_mgr_get_substituted_value': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_include.o (symbol from plugin): in function `uc_mgr_config_tree_merge': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_regex.o (symbol from plugin): in function `uc_mgr_define_regex': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_exec.o (symbol from plugin): in function `uc_mgr_exec': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: main.o (symbol from plugin): in function `snd_use_case_identifier': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here collect2: error: ld returned 1 exit status make[2]: *** [Makefile:495: libasound.la] Error 1 make[2]: Leaving directory '/var/tmp/portage/media-libs/alsa-lib-9999/work/alsa-lib-9999-abi_x86_32.x86/src' make[1]: *** [Makefile:555: all-recursive] Error 1 make[1]: Leaving directory '/var/tmp/portage/media-libs/alsa-lib-9999/work/alsa-lib-9999-abi_x86_32.x86/src' make: *** [Makefile:407: all-recursive] Error 1 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/171 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

(.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here
by GitHub issues - opened 20 Aug '21

20 Aug '21

alsa-project/alsa-lib issue #171 was opened from shoober420: Trying to compile alsa-lib from the latest two commits as of Aug 20th (https://github.com/alsa-project/alsa-lib/commit/868ecd509deb05938107faf6657…) (https://github.com/alsa-project/alsa-lib/commit/95eb312fade1908a2c944e9de46…), I get this error during build. Im using Gentoo Linux with GCC 11.2 version. [buildalsalib.log](https://github.com/alsa-project/alsa-lib/files/7023522/bu… ``` (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: parser.o (symbol from plugin): in function `uc_mgr_config_load_file': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_cond.o (symbol from plugin): in function `uc_mgr_evaluate_condition': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_subs.o (symbol from plugin): in function `uc_mgr_get_substituted_value': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_include.o (symbol from plugin): in function `uc_mgr_config_tree_merge': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_regex.o (symbol from plugin): in function `uc_mgr_define_regex': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: ucm_exec.o (symbol from plugin): in function `uc_mgr_exec': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here /usr/lib/gcc/x86_64-pc-linux-gnu/11.2.0/../../../../x86_64-pc-linux-gnu/bin/ld: main.o (symbol from plugin): in function `snd_use_case_identifier': (.text+0x0): multiple definition of `__packed'; .libs/conf.o (symbol from plugin):(.text+0x0): first defined here collect2: error: ld returned 1 exit status make[2]: *** [Makefile:495: libasound.la] Error 1 make[2]: Leaving directory '/var/tmp/portage/media-libs/alsa-lib-9999/work/alsa-lib-9999-abi_x86_32.x86/src' make[1]: *** [Makefile:555: all-recursive] Error 1 make[1]: Leaving directory '/var/tmp/portage/media-libs/alsa-lib-9999/work/alsa-lib-9999-abi_x86_32.x86/src' make: *** [Makefile:407: all-recursive] Error 1 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/171 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/5] ASoC: Intel/rt5640: Add support for HP Elite Pad 1000G2 jack-detect
by Hans de Goede 20 Aug '21

20 Aug '21

The HP Elitepad 1000 G2 tablet has 2 headset jacks: 1. on the dock which uses the output of the codecs built-in HP-amp + the standard IN2 input which is always used with the headset-jack. 2. on the tablet itself, this uses the line-out of the codec + an external HP-amp, which gets enabled by the ALC5642 codec's GPIO1 pin; and IN1 for the headset-mic. The codec's GPIO1 is also its only IRQ output pin, so this means that the codec's IRQ cannot be used on this tablet. Instead the jack-detect is connected directly to GPIOs on the main SoC. The dock has a helper chip which also detects if a headset-mic is present or not, so there are 2 GPIOs for the jack-detect status of the dock. The tablet jack uses a single GPIO which indicates if a jack is present or not. Differentiating between between headphones vs a headset on the tablet jack is done by using the usual mic-bias over-current-detection mechanism. Regards, Hans Hans de Goede (5): ASoC: rt5640: Move rt5640_disable_jack_detect() up in the rt5640.c file ASoC: rt5640: Delay requesting IRQ until the machine-drv calls set_jack ASoC: rt5640: Add optional hp_det_gpio parameter to rt5640_detect_headset() ASoC: rt5640: Add rt5640_set_ovcd_params() helper ASoC: Intel: bytcr_rt5640: Add support for HP Elite Pad 1000G2 jack-detect sound/soc/codecs/rt5640.c | 133 ++++++++++++----------- sound/soc/codecs/rt5640.h | 6 ++ sound/soc/intel/boards/bytcr_rt5640.c | 146 +++++++++++++++++++++++++- 3 files changed, 225 insertions(+), 60 deletions(-) -- 2.31.1

3 10

[PATCH] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Kuninori Morimoto 20 Aug '21

20 Aug '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..5f1e72edfee0 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -391,9 +391,9 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, struct clk *clk; clk = clk_register_fixed_rate(dev, name, parent, 0, 0); - if (IS_ERR(clk)) { + if (IS_ERR_OR_NULL(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; @@ -582,7 +582,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clkout[CLKOUT] = clk; @@ -596,7 +596,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, req_rate[0]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clkout[i] = clk; -- 2.25.1

2 1

[PATCH] ASoC: rsnd: core: make some arrays static const, makes object smaller
by Kuninori Morimoto 20 Aug '21

20 Aug '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Don't populate arrays on the stack but instead them static const. Makes the object code smaller by 48 bytes. Before: text data bss dec hex filename 20938 916 104 21958 55c6 ./sound/soc/sh/rcar/core.o After: text data bss dec hex filename 20890 916 104 21910 5596 ./sound/soc/sh/rcar/core.o gcc version 11.1.0) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5e382b5c9d45..978bd0406729 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -426,19 +426,19 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - enum rsnd_mod_type playback_mods[] = { + static const enum rsnd_mod_type playback_mods[] = { RSND_MOD_SRC, RSND_MOD_CMD, RSND_MOD_SSIU, }; - enum rsnd_mod_type capture_mods[] = { + static const enum rsnd_mod_type capture_mods[] = { RSND_MOD_CMD, RSND_MOD_SRC, RSND_MOD_SSIU, }; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_mod *tmod = NULL; - enum rsnd_mod_type *mods = + const enum rsnd_mod_type *mods = rsnd_io_is_play(io) ? playback_mods : capture_mods; int i; -- 2.25.1

2 1

[PATCH v2 0/6] ASoC: Intel/rt5640: Add support for HP Elite Pad 1000G2 jack-detect
by Hans de Goede 20 Aug '21

20 Aug '21

Changes in v2: - Rebase on asoc/for-next - New patch: "ASoC: Intel: bytct_rt5640: Add a separate "Headset Mic 2" DAPM pin for the mic on the 2nd jack" - Addressed Pierre-Louis' comments about calling acpi_dev_add_driver_gpios() twice Original cover-letter: The HP Elitepad 1000 G2 tablet has 2 headset jacks: 1. on the dock which uses the output of the codecs built-in HP-amp + the standard IN2 input which is always used with the headset-jack. 2. on the tablet itself, this uses the line-out of the codec + an external HP-amp, which gets enabled by the ALC5642 codec's GPIO1 pin; and IN1 for the headset-mic. The codec's GPIO1 is also its only IRQ output pin, so this means that the codec's IRQ cannot be used on this tablet. Instead the jack-detect is connected directly to GPIOs on the main SoC. The dock has a helper chip which also detects if a headset-mic is present or not, so there are 2 GPIOs for the jack-detect status of the dock. The tablet jack uses a single GPIO which indicates if a jack is present or not. Differentiating between between headphones vs a headset on the tablet jack is done by using the usual mic-bias over-current-detection mechanism. Regards, Hans Hans de Goede (6): ASoC: rt5640: Move rt5640_disable_jack_detect() up in the rt5640.c file ASoC: rt5640: Delay requesting IRQ until the machine-drv calls set_jack ASoC: rt5640: Add optional hp_det_gpio parameter to rt5640_detect_headset() ASoC: rt5640: Add rt5640_set_ovcd_params() helper ASoC: Intel: bytct_rt5640: Add a separate "Headset Mic 2" DAPM pin for the mic on the 2nd jack ASoC: Intel: bytcr_rt5640: Add support for HP Elite Pad 1000G2 jack-detect sound/soc/codecs/rt5640.c | 136 ++++++++++++---------- sound/soc/codecs/rt5640.h | 6 + sound/soc/intel/boards/bytcr_rt5640.c | 158 +++++++++++++++++++++++++- 3 files changed, 234 insertions(+), 66 deletions(-) -- 2.31.1

4 9

[PATCH] ALSA: hda/realtek: Workaround for conflicting SSID on ASUS ROG Strix G17
by Takashi Iwai 20 Aug '21

20 Aug '21

ASUS ROG Strix G17 has the very same PCI and codec SSID (1043:103f) as ASUS TX300, and unfortunately, the existing quirk for TX300 is broken on ASUS ROG. Actually the device works without the quirk, so we'll need to clear the quirk before applying for this device. Since ASUS ROG has a different codec (ALC294 - while TX300 has ALC282), this patch adds a workaround for the device, just clearing the codec->fixup_id by checking the codec vendor_id. It's a bit ugly to add such a workaround there, but it seems to be the simplest way. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214101 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7ad689f991e7..3d525b79b848 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9521,6 +9521,16 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); + /* FIXME: both TX300 and ROG Strix G17 have the same SSID, and + * the quirk breaks the latter (bko#214101). + * Clear the wrong entry. + */ + if (codec->fixup_id == ALC282_FIXUP_ASUS_TX300 && + codec->core.vendor_id == 0x10ec0294) { + codec_dbg(codec, "Clear wrong fixup for ASUS ROG Strix G17\n"); + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; + } + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, -- 2.26.2

1 0

[PATCH v3 0/2] alsa-lib patches for rawmidi framing
by David Henningsson 20 Aug '21

20 Aug '21

v3: Adjusted according to comments from tiwai David Henningsson (2): Add rawmidi framing API Add test for rawmidi framing API include/rawmidi.h | 38 +++++++++++++++++ include/sound/uapi/asound.h | 30 +++++++++++++- src/rawmidi/rawmidi.c | 83 +++++++++++++++++++++++++++++++++++++ src/rawmidi/rawmidi_hw.c | 2 + src/rawmidi/rawmidi_local.h | 2 + test/rawmidi.c | 65 ++++++++++++++++++++++++++--- 6 files changed, 212 insertions(+), 8 deletions(-) -- 2.25.1

2 3

[PATCH v2] ASoC: Intel: Fix platform ID matching for kbl_da7219_max98373
by Lukasz Majczak 19 Aug '21

19 Aug '21

Sparse warnings triggered truncating the IDs of some platform device tables. Unfortunately kbl_da7219_max98373 was also truncated. This patch is reverting the original ID. Tested on Atlas chromebook. Fixes: 94efd726b947 ("ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters") Cc: <stable(a)vger.kernel.org> # 5.4+ Tested-by: Lukasz Majczak <lma(a)semihalf.com> Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> Suggested-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 741bf2f9e081..8cab91a00b1a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -113,7 +113,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_mx98373", + .drv_name = "kbl_da7219_max98373", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, -- 2.33.0.rc2.250.ged5fa647cd-goog

4 4

[PATCH v3] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 19 Aug '21

19 Aug '21

dma_request_chan() returns error pointer in case of failures, but the rz_ssi_dma_request() checked for NULL pointer instead. This patch fixes the issue by checking for ERR_PTR() instead of NULL and sets the DMA pointers to NULL in error case so that ssi can fallback to PIO mode. Fixes: 26ac471c5354 ("ASoC: sh: rz-ssi: Add SSI DMAC support") Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Reviewed-by: Lad Prabhakar <prabhakar.mahadev-lad.rj(a)bp.renesas.com> --- v2->v3: * Added Fixes tag * For SSI with with full-duplex transmission and reception, Add support for fallback to PIO mode, if both channels are not available. v1->v2: * Replaced IS_ERR_OR_NULL with IS_ERR for error pointer check, as the dma_request_chan function never returns NULL pointer for error case. --- sound/soc/sh/rz-ssi.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index f097c773d413..fa0cc08f70ec 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -185,7 +185,7 @@ rz_ssi_stream_get(struct rz_ssi_priv *ssi, struct snd_pcm_substream *substream) static inline bool rz_ssi_is_dma_enabled(struct rz_ssi_priv *ssi) { - return (ssi->playback.dma_ch || ssi->capture.dma_ch); + return (ssi->playback.dma_ch && (ssi->dma_rt || ssi->capture.dma_ch)); } static int rz_ssi_stream_is_valid(struct rz_ssi_priv *ssi, @@ -676,15 +676,26 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } + if (!rz_ssi_is_dma_enabled(ssi)) + goto no_dma; + if (ssi->playback.dma_ch && (rz_ssi_dma_slave_config(ssi, ssi->playback.dma_ch, true) < 0)) goto no_dma; -- 2.17.1

2 1

[PATCH] ASoC: uniphier: make arrays mul and div static const, makes object smaller
by Colin King 19 Aug '21

19 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the arrays mul and div on the stack but instead make them static const. Makes the object code smaller by 4 bytes. Before: text data bss dec hex filename 16226 4984 64 21274 531a ./sound/soc/uniphier/aio-cpu.o After: text data bss dec hex filename 16062 5144 64 21270 5316 ./sound/soc/uniphier/aio-cpu.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/uniphier/aio-cpu.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index cf9814130067..4e8d5f7532ba 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -128,8 +128,8 @@ static const struct uniphier_aio_spec *find_spec(struct uniphier_aio *aio, static int find_divider(struct uniphier_aio *aio, int pll_id, unsigned int freq) { struct uniphier_aio_pll *pll; - int mul[] = { 1, 1, 1, 2, }; - int div[] = { 2, 3, 1, 3, }; + static const int mul[] = { 1, 1, 1, 2, }; + static const int div[] = { 2, 3, 1, 3, }; int i; if (!is_valid_pll(aio->chip, pll_id)) -- 2.32.0

3 2

[PATCH 0/4] Rockchip I2S/TDM controller
by Nicolas Frattaroli 19 Aug '21

19 Aug '21

Hello, I come bearing four patches for your consideration. The first of these four patches adds a driver for the Rockchip I2S/TDM controller, used in interfacing between the AHB bus and the I2S bus on some Rockchip SoCs. This allows for audio playback with a matching codec. The controller has three different modes: I2S, I2S/TDM and PCM. It is distinct from the earlier Rockchip I2S controller, and therefore not just an extension of that driver. The driver is based on the downstream version, though various changes have been made to hopefully make it more palatable to upstream. Some needless code duplication has been refactored, and the probe function will no longer let wrong device tree values write nonsense to hardware registers. Properties have been renamed and had their semantics changed. I won't bore you with the details of what downstream did, since that's not what I'm submitting, but the changes are significant enough that I've added myself to the list of authors. The second patch adds the YAML device tree bindings for this, which have been written from scratch by yours truly. Since I didn't like having random integers mean things, I defined them as constants in a header file for the bindings. The third patch adds the i2s1 controller to the rk356x device tree. I didn't add any of the other i2s controllers on that SoC for now as I have no way of testing them; in particular, i2s0 is tied to HDMI, so needs a functioning VOP2 driver to even have a chance of working. The fourth patch makes use of the i2s1 controller to enable analog audio output on the Quartz64 Model A through its RK817 codec. I've tested this to work properly at both 44.1 kHz and 96 kHz, so both mclk_root0 and mclk_root1 are definitely functioning. This is my first kernel contribution, so I most likely did something horribly wrong. That's why I'm more than happy to receive any criticisms and concerns over how the driver is implemented, because I've run out of ideas on how to make it clearly better myself. I'd also like to extend my thanks to Peter Geis, who has been acting as somewhat of a mentor and gave me occasional feedback and ideas during the writing of this patch series. Regards, Nicolas Frattaroli Nicolas Frattaroli (4): ASoC: rockchip: add support for i2s-tdm controller dt-bindings: sound: add rockchip i2s-tdm binding arm64: dts: rockchip: add i2s1 on rk356x arm64: dts: rockchip: add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 221 ++ .../boot/dts/rockchip/rk3566-quartz64-a.dts | 36 +- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 26 + include/dt-bindings/sound/rockchip,i2s-tdm.h | 9 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1804 +++++++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 400 ++++ 8 files changed, 2508 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml create mode 100644 include/dt-bindings/sound/rockchip,i2s-tdm.h create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h -- 2.32.0

5 9

[PATCH] ASoC: intel: atom: Fix breakage for PCM buffer address setup
by Takashi Iwai 19 Aug '21

19 Aug '21

The commit 2e6b836312a4 ("ASoC: intel: atom: Fix reference to PCM buffer address") changed the reference of PCM buffer address to substream->runtime->dma_addr as the buffer address may change dynamically. However, I forgot that the dma_addr field is still not set up for the CONTINUOUS buffer type (that this driver uses) yet in 5.14 and earlier kernels, and it resulted in garbage I/O. The problem will be fixed in 5.15, but we need to address it quickly for now. The fix is to deduce the address again from the DMA pointer with virt_to_phys(), but from the right one, substream->runtime->dma_area. Fixes: 2e6b836312a4 ("ASoC: intel: atom: Fix reference to PCM buffer address") Reported-and-tested-by: Hans de Goede <hdegoede(a)redhat.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/2048c6aa-2187-46bd-6772-36a4fb3c5aeb@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 5db2f4865bbb..905c7965f653 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = substream->runtime->dma_addr; + u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.26.2

2 3

[PATCH] ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8
by Kai-Heng Feng 19 Aug '21

19 Aug '21

The mic has lots of noises if mic boost is enabled. So disable mic boost to get crystal clear audio capture. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b72064f83211..cdd87917dd8cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6645,6 +6645,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8229,6 +8230,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8425,8 +8432,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.32.0

2 1

[5.14 regression] "ASoC: intel: atom: Fix reference to PCM buffer address" breaks Intel SST audio
by Hans de Goede 19 Aug '21

19 Aug '21

Hi All, After rebasing a set of bytcr_rt5640 patches, on top of asoc/for-next I noticed that playing back audio would only generate random-noise / buzzing (I did not try recording any audio). After poking at this for a while I've found the culprit: 2e6b836312a4 ("ASoC: intel: atom: Fix reference to PCM buffer address") If I revert that single commit then audio on Intel Bay Trail and Cherry Trail devices works fine again with 5.14. This is with a Fedora 34 userspace using pipewire as audiodaemon I'm not sure what is going on here, but since the old code changed by the broken commit has worked fine for ages and given where we are in the devel-cycle I think it might be best to just revert 2e6b836312a4 again. Regards, Hans

2 4

huge mixers
by Philippe Bekaert 19 Aug '21

19 Aug '21

Dear all, I am working on a new driver for the RME HDSPe cards, which eventually could replace the hdspm driver. These cards have a hardware mixer / matrox router, freely mixing tens of hardware inputs and software playbacks into tens of outputs (8192 mixer controls on the HDSPe MADI). Right now, mixer state (cached by the driver) can be read efficiently in one ad-hoc ioctl call, and individual channels modified through a HWDEP ALSA control element with 3 parameters (input/playback index, output index, gain value). I understand there is a desire to get rid of ad hoc ioctls and am therefore looking for a more generic way to read and write huge mixer state. Has anyone been facing similar issues? How have you been solving it? Is there any “common practice” or “habits” in this community concerning this topic? Would it make sense to define a kind of generic huge mixer interface? Then, has anyone an idea or preference how it should look like? Looking forward to your feedback. Best regards and thanks in advance, Philippe.

2 1

PCM channel names
by Philippe Bekaert 19 Aug '21

19 Aug '21

Same context as my previous mail … The RME HDSPe cards have a large number of input and outputs for which the standard ALSA channels names and mappings make little sense. Names depend on the card, e.g. Analog.L, Analog.R, AES.1/1 … AES.1/8, ADAT.1 … ADAT.8 etc… Right now, the hdspm driver provides a ad-hoc virtual file in /proc/asound/card<n> containing these names. Isn’t there a more generic interface to communicate PCM channel names from driver to user space Same questions as before: Has anyone been facing similar issues? How have you been solving it? Is there any “common practice” or “habits” in this community concerning this topic? Would it make sense to define a kind of generic interface for PCM naming? Then, has anyone an idea or preference how it should look like? Again looking forward to your feedback. Best regards and thanks in advance, Philippe.

2 1

[RFC PATCH 0/2] driver core: kick deferred probe from delayed context
by Pierre-Louis Bossart 18 Aug '21

18 Aug '21

The deferred probe mechanism uses a successful driver probe/attach as a trigger to revisit the list of deferred probe devices. This works in most cases, except when the probe success is not a valid indicator of resources being available. In that case, a race condition may occur, where the device/driver core framework will attempt to probe a device that depends on resources before those resources are available, resulting in a -EPROBE_DEFER error and a deferred probe device that will never be initialized. The example provided in this RFC relies on the probe workqueue used for the HDaudio support where we simultaneously: a) need to use request_module() b) cannot use an async probe due to the use of request_module() c) cannot block the probe of other drivers In this example, the deferred probe can be kicked when the workqueue completes. The use of request_firmware_nowait() is another conceptual example, where a domain-specific callback can enable resources *after* the probe returns, for example by downloading the firmware, booting a processor and waiting for the processor to be ready for interaction with the Linux host. In this second example, the deferred probe could be kicked when the 'cont' callback completes. This patchset suggests a 7-line change to solve race conditions in these examples with delayed work. Discussion: a) During Intel internal reviews, Andy Shevchenko pointed out another known issue with deferred probe [1]. This patchset is unrelated and does not claim to solve the problem raised by Andy. b) one possible objection is that this patchset does not suppress a possibly unnecessary round of evaluation of deferred probe devices. It did not feel necessary to any of us to minimize the occurrences of EPROBE_DEFER but instead to make sure the device waiting for resources successfully probes in the end. c) another objection might be that the driver core should know about such dependencies. It would be desirable but in the cases we've encountered such dependencies are highly domain-specific and not necessarily straightforward to describe. There's been multiple endeavors to improve the description of dependencies, this patchset only focuses on the deferred probe framework, with an improvement when the provider of resources makes these resources available after its probe returns. [1] https://lore.kernel.org/lkml/20200324175719.62496-1-andriy.shevchenko@linux… Pierre-Louis Bossart (2): driver core: export driver_deferred_probe_trigger() ASoC: SOF: trigger re-probing of deferred devices from workqueue drivers/base/dd.c | 3 ++- include/linux/device/driver.h | 1 + sound/soc/sof/core.c | 3 +++ 3 files changed, 6 insertions(+), 1 deletion(-) base-commit: 8d1998893cd5e3488cd95529f60a187e3009d14b -- 2.25.1

4 16

[PATCH] ASoC: fsl_rpmsg: Check -EPROBE_DEFER for getting clocks
by Shengjiu Wang 18 Aug '21

18 Aug '21

The devm_clk_get() may return -EPROBE_DEFER, then clocks will be assigned to NULL wrongly. As the clocks are optional so we can use devm_clk_get_optional() instead of devm_clk_get(). Fixes: b73d9e6225e8 ("ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_rpmsg.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index ea5c973e2e84..d60f4dac6c1b 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -165,25 +165,25 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) } /* Get the optional clocks */ - rpmsg->ipg = devm_clk_get(&pdev->dev, "ipg"); + rpmsg->ipg = devm_clk_get_optional(&pdev->dev, "ipg"); if (IS_ERR(rpmsg->ipg)) - rpmsg->ipg = NULL; + return PTR_ERR(rpmsg->ipg); - rpmsg->mclk = devm_clk_get(&pdev->dev, "mclk"); + rpmsg->mclk = devm_clk_get_optional(&pdev->dev, "mclk"); if (IS_ERR(rpmsg->mclk)) - rpmsg->mclk = NULL; + return PTR_ERR(rpmsg->mclk); - rpmsg->dma = devm_clk_get(&pdev->dev, "dma"); + rpmsg->dma = devm_clk_get_optional(&pdev->dev, "dma"); if (IS_ERR(rpmsg->dma)) - rpmsg->dma = NULL; + return PTR_ERR(rpmsg->dma); - rpmsg->pll8k = devm_clk_get(&pdev->dev, "pll8k"); + rpmsg->pll8k = devm_clk_get_optional(&pdev->dev, "pll8k"); if (IS_ERR(rpmsg->pll8k)) - rpmsg->pll8k = NULL; + return PTR_ERR(rpmsg->pll8k); - rpmsg->pll11k = devm_clk_get(&pdev->dev, "pll11k"); + rpmsg->pll11k = devm_clk_get_optional(&pdev->dev, "pll11k"); if (IS_ERR(rpmsg->pll11k)) - rpmsg->pll11k = NULL; + return PTR_ERR(rpmsg->pll11k); platform_set_drvdata(pdev, rpmsg); pm_runtime_enable(&pdev->dev); -- 2.27.0

2 1

[PATCH v4 00/11] ASoC: mediatek: Add support for MT8195 SoC
by Trevor Wu 18 Aug '21

18 Aug '21

This series of patches adds support for Mediatek AFE of MT8195 SoC. Patches are based on broonie tree "for-next" branch. Changes since v3: - fixed warnings found by kernel test robot - removed unused critical section - corrected the lock protected sections on etdm driver - added DPTX and HDMITX audio support Changes since v2: - added audio clock gate control - added 'mediatek' prefix to private dts properties - added consumed clocks to dt-bindins and adopted suggestions from Rob - refined clock usage and remove unused clock and control code - fixed typos Changes since v1: - fixed some problems related to dt-bindings - added some missing properties to dt-bindings - added depency declaration on dt-bindings - fixed some warnings found by kernel test robot Trevor Wu (11): ASoC: mediatek: mt8195: update mediatek common driver ASoC: mediatek: mt8195: support audsys clock control ASoC: mediatek: mt8195: support etdm in platform driver ASoC: mediatek: mt8195: support adda in platform driver ASoC: mediatek: mt8195: support pcm in platform driver ASoC: mediatek: mt8195: add platform driver dt-bindings: mediatek: mt8195: add audio afe document ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 ASoC: mediatek: mt8195: add DPTX audio support ASoC: mediatek: mt8195: add HDMITX audio support dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682 document .../bindings/sound/mt8195-afe-pcm.yaml | 184 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 24 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 22 +- sound/soc/mediatek/common/mtk-base-afe.h | 10 +- sound/soc/mediatek/mt8195/Makefile | 15 + sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 441 +++ sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 109 + sound/soc/mediatek/mt8195/mt8195-afe-common.h | 158 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3281 +++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 214 ++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 15 + .../soc/mediatek/mt8195/mt8195-audsys-clkid.h | 93 + sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 830 +++++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2639 +++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 389 ++ .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 1093 ++++++ sound/soc/mediatek/mt8195/mt8195-reg.h | 2796 ++++++++++++++ 19 files changed, 12356 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1019-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/Makefile create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-adda.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h -- 2.18.0

1 11

[PATCH -next] ASoC: sh: rz-ssi: Fix return value check in rz_ssi_dma_request()
by Yang Yingliang 18 Aug '21

18 Aug '21

In case of error, the function dma_request_chan() returns ERR_PTR() and never returns NULL. Set 'dma_ch' to NULL, if dma_request_chan() returns error, so the code using 'dma_ch' can work correctly. Fixes: 26ac471c5354 ("ASoC: sh: rz-ssi: Add SSI DMAC support") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sh/rz-ssi.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..5ec78fd94d94 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,17 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.25.1

2 2

[PATCH 1/2] ALSA: hda/sigmatel - Sink stac_shutup() into stac_suspend()
by Nathan Chancellor 18 Aug '21

18 Aug '21

When CONFIG_PM is not set, there is an unused function warning: sound/pci/hda/patch_sigmatel.c:4383:13: warning: unused function 'stac_shutup' [-Wunused-function] static void stac_shutup(struct hda_codec *codec) ^ 1 warning generated. Sink the contents of stac_shutup() into stac_suspend() since stac_shutup() is only called in that one location now. Fixes: 327b34f2a97d ("ALSA: hda: Nuke unused reboot_notify callback") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/pci/hda/patch_sigmatel.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3e00116db86e..61df4d33c48f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4380,18 +4380,6 @@ static int stac_init(struct hda_codec *codec) return 0; } -static void stac_shutup(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - snd_hda_shutup_pins(codec); - - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); -} - #define stac_free snd_hda_gen_free #ifdef CONFIG_SND_PROC_FS @@ -4444,7 +4432,15 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #ifdef CONFIG_PM static int stac_suspend(struct hda_codec *codec) { - stac_shutup(codec); + struct sigmatel_spec *spec = codec->spec; + + snd_hda_shutup_pins(codec); + + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); + return 0; } #else base-commit: 8fc8e903156f42c66245838441d03607e9067381 -- 2.33.0

2 3

[PATCH] ASoC: nau8821: new driver
by Seven Lee 18 Aug '21

18 Aug '21

Add driver for NAU88L21. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8821.c | 1818 +++++++++++++++++ sound/soc/codecs/nau8821.h | 533 +++++ 5 files changed, 2413 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 sound/soc/codecs/nau8821.c create mode 100644 sound/soc/codecs/nau8821.h diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt new file mode 100644 index 000000000000..6c3baf7a5f21 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -0,0 +1,55 @@ +Nuvoton NAU88L21 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8821" + + - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. + + +Example: + + headset: nau8821@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,dmic-clk-threshold = 3072000; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fe5e558635ad..6757f30779ff 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -136,6 +136,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 + imply SND_SOC_NAU8821 imply SND_SOC_NAU8822 imply SND_SOC_NAU8824 imply SND_SOC_NAU8825 @@ -1904,6 +1905,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8821 + tristate "Nuvoton Technology Corporation NAU88L21 CODEC" + depends on I2C + config SND_SOC_NAU8822 tristate "Nuvoton Technology Corporation NAU88C22 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..5cb0bbaba1a5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -141,6 +141,7 @@ snd-soc-mt6660-objs := mt6660.o snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8821-objs := nau8821.o snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o @@ -468,6 +469,7 @@ obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8821) += snd-soc-nau8821.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c new file mode 100644 index 000000000000..50b89841cfd8 --- /dev/null +++ b/sound/soc/codecs/nau8821.c @@ -0,0 +1,1818 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * nau8821.c -- Nuvoton NAU88L21 audio codec driver + * + * Copyright 2021 Nuvoton Technology Corp. + * Author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Lee <wtli(a)nuvoton.com> + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/acpi.h> +#include <linux/math64.h> +#include <linux/semaphore.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "nau8821.h" + +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MIN 90000000 + +/* the maximum frequency of CLK_ADC and CLK_DAC */ +#define CLK_DA_AD_MAX 6144000 + +static int nau8821_configure_sysclk(struct nau8821 *nau8821, + int clk_id, unsigned int freq); + +struct nau8821_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau8821_fll_attr { + unsigned int param; + unsigned int val; +}; + +/* scaling for mclk from sysclk_src output */ +static const struct nau8821_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, + { 48, 0xd }, + { 96, 0xe }, + { 5, 0xf }, +}; + +/* ratio for input clk freq */ +static const struct nau8821_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau8821_fll_attr fll_pre_scalar[] = { + { 0, 0x0 }, + { 1, 0x1 }, + { 2, 0x2 }, + { 3, 0x3 }, +}; + +/* over sampling rate */ +struct nau8821_osr_attr { + unsigned int osr; + unsigned int clk_src; +}; + +static const struct nau8821_osr_attr osr_dac_sel[] = { + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 256, 0 }, /* OSR 256, SRC 1 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 0, 0 }, + { 32, 3 }, /* OSR 32, SRC 1/8 */ +}; + +static const struct nau8821_osr_attr osr_adc_sel[] = { + { 32, 3 }, /* OSR 32, SRC 1/8 */ + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 256, 0 }, /* OSR 256, SRC 1 */ +}; + +struct nau8821_dmic_speed { + unsigned int param; + unsigned int val; +}; + +static const struct nau8821_dmic_speed dmic_speed_sel[] = { + { 0, 0x0 }, /*SPEED 1, SRC 1 */ + { 1, 0x1 }, /*SPEED 2, SRC 1/2 */ + { 2, 0x2 }, /*SPEED 4, SRC 1/4 */ + { 3, 0x3 }, /*SPEED 8, SRC 1/8 */ +}; + +static const struct reg_default nau8821_reg_defaults[] = { + { NAU8821_R01_ENA_CTRL, 0x00ff }, + { NAU8821_R03_CLK_DIVIDER, 0x0050 }, + { NAU8821_R04_FLL1, 0x0 }, + { NAU8821_R05_FLL2, 0x00bc }, + { NAU8821_R06_FLL3, 0x0008 }, + { NAU8821_R07_FLL4, 0x0010 }, + { NAU8821_R08_FLL5, 0x4000 }, + { NAU8821_R09_FLL6, 0x6900 }, + { NAU8821_R0A_FLL7, 0x0031 }, + { NAU8821_R0B_FLL8, 0x26e9 }, + { NAU8821_R0D_JACK_DET_CTRL, 0x0 }, + { NAU8821_R0F_INTERRUPT_MASK, 0x0 }, + { NAU8821_R12_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8821_R13_DMIC_CTRL, 0x0 }, + { NAU8821_R1A_GPIO12_CTRL, 0x0 }, + { NAU8821_R1B_TDM_CTRL, 0x0 }, + { NAU8821_R1C_I2S_PCM_CTRL1, 0x000a }, + { NAU8821_R1D_I2S_PCM_CTRL2, 0x8010 }, + { NAU8821_R1E_LEFT_TIME_SLOT, 0x0 }, + { NAU8821_R1F_RIGHT_TIME_SLOT, 0x0 }, + { NAU8821_R21_BIQ0_COF1, 0x0 }, + { NAU8821_R22_BIQ0_COF2, 0x0 }, + { NAU8821_R23_BIQ0_COF3, 0x0 }, + { NAU8821_R24_BIQ0_COF4, 0x0 }, + { NAU8821_R25_BIQ0_COF5, 0x0 }, + { NAU8821_R26_BIQ0_COF6, 0x0 }, + { NAU8821_R27_BIQ0_COF7, 0x0 }, + { NAU8821_R28_BIQ0_COF8, 0x0 }, + { NAU8821_R29_BIQ0_COF9, 0x0 }, + { NAU8821_R2A_BIQ0_COF10, 0x0 }, + { NAU8821_R2B_ADC_RATE, 0x0002 }, + { NAU8821_R2C_DAC_CTRL1, 0x0082 }, + { NAU8821_R2D_DAC_CTRL2, 0x0 }, + { NAU8821_R2F_DAC_DGAIN_CTRL, 0x0 }, + { NAU8821_R30_ADC_DGAIN_CTRL, 0x0 }, + { NAU8821_R31_MUTE_CTRL, 0x0 }, + { NAU8821_R32_HSVOL_CTRL, 0x0 }, + { NAU8821_R34_DACR_CTRL, 0xcfcf }, + { NAU8821_R35_ADC_DGAIN_CTRL1, 0xcfcf }, + { NAU8821_R36_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8821_R37_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8821_R38_ADC_DRC_SLOPES, 0x25ff }, + { NAU8821_R39_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8821_R3A_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8821_R3B_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8821_R3C_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8821_R3D_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8821_R41_BIQ1_COF1, 0x0 }, + { NAU8821_R42_BIQ1_COF2, 0x0 }, + { NAU8821_R43_BIQ1_COF3, 0x0 }, + { NAU8821_R44_BIQ1_COF4, 0x0 }, + { NAU8821_R45_BIQ1_COF5, 0x0 }, + { NAU8821_R46_BIQ1_COF6, 0x0 }, + { NAU8821_R47_BIQ1_COF7, 0x0 }, + { NAU8821_R48_BIQ1_COF8, 0x0 }, + { NAU8821_R49_BIQ1_COF9, 0x0 }, + { NAU8821_R4A_BIQ1_COF10, 0x0 }, + { NAU8821_R4B_CLASSG_CTRL, 0x0 }, + { NAU8821_R4C_IMM_MODE_CTRL, 0x0 }, + { NAU8821_R4D_IMM_RMS_L, 0x0 }, + { NAU8821_R53_OTPDOUT_1, 0xaad8 }, + { NAU8821_R54_OTPDOUT_2, 0x0002 }, + { NAU8821_R55_MISC_CTRL, 0x0 }, + { NAU8821_R66_BIAS_ADJ, 0x0 }, + { NAU8821_R68_TRIM_SETTINGS, 0x0 }, + { NAU8821_R69_ANALOG_CONTROL_1, 0x0 }, + { NAU8821_R6A_ANALOG_CONTROL_2, 0x0 }, + { NAU8821_R6B_PGA_MUTE, 0x0 }, + { NAU8821_R71_ANALOG_ADC_1, 0x0011 }, + { NAU8821_R72_ANALOG_ADC_2, 0x0020 }, + { NAU8821_R73_RDAC, 0x0008 }, + { NAU8821_R74_MIC_BIAS, 0x0006 }, + { NAU8821_R76_BOOST, 0x0 }, + { NAU8821_R77_FEPGA, 0x0 }, + { NAU8821_R7E_PGA_GAIN, 0x0 }, + { NAU8821_R7F_POWER_UP_CONTROL, 0x0 }, + { NAU8821_R80_CHARGE_PUMP, 0x0 }, +}; + +/** + * nau8821_sema_acquire - acquire the semaphore of nau8821 + * @nau8821: component to register the codec private data with + * @timeout: how long in jiffies to wait before failure or zero to wait + * until release + * + * Attempts to acquire the semaphore with number of jiffies. If no more + * tasks are allowed to acquire the semaphore, calling this function will + * put the task to sleep. If the semaphore is not released within the + * specified number of jiffies, this function returns. + * If the semaphore is not released within the specified number of jiffies, + * this function returns -ETIME. If the sleep is interrupted by a signal, + * this function will return -EINTR. It returns 0 if the semaphore was + * acquired successfully. + * + * Acquires the semaphore without jiffies. Try to acquire the semaphore + * atomically. Returns 0 if the semaphore has been acquired successfully + * or 1 if it cannot be acquired. + */ +static int nau8821_sema_acquire(struct nau8821 *nau8821, long timeout) +{ + int ret; + + if (!nau8821->irq) + return 1; + + if (timeout) { + ret = down_timeout(&nau8821->jd_sem, timeout); + if (ret < 0) + dev_dbg(nau8821->dev, "Acquire semaphore timeout\n"); + } else { + ret = down_trylock(&nau8821->jd_sem); + if (ret) + dev_dbg(nau8821->dev, "Acquire semaphore fail\n"); + } + + return ret; +} + +/** + * nau8821_sema_release - release the semaphore of nau8821 + * @nau8821: component to register the codec private data with + * + * Release the semaphore which may be called from any context and + * even by tasks which have never called down(). + */ +static inline void nau8821_sema_release(struct nau8821 *nau8821) +{ + if (!nau8821->irq) + return; + up(&nau8821->jd_sem); +} + +/** + * nau8821_sema_reset - reset the semaphore for nau8821 + * @nau8821: component to register the codec private data with + * + * Reset the counter of the semaphore. Call this function to restart + * a new round task management. + */ +static inline void nau8821_sema_reset(struct nau8821 *nau8821) +{ + nau8821->jd_sem.count = 1; +} + +static bool nau8821_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET ... NAU8821_R01_ENA_CTRL: + case NAU8821_R03_CLK_DIVIDER ... NAU8821_R0B_FLL8: + case NAU8821_R0D_JACK_DET_CTRL: + case NAU8821_R0F_INTERRUPT_MASK ... NAU8821_R13_DMIC_CTRL: + case NAU8821_R1A_GPIO12_CTRL ... NAU8821_R1F_RIGHT_TIME_SLOT: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2D_DAC_CTRL2: + case NAU8821_R2F_DAC_DGAIN_CTRL ... NAU8821_R32_HSVOL_CTRL: + case NAU8821_R34_DACR_CTRL ... NAU8821_R3D_DAC_DRC_ATKDCY: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4F_FUSE_CTRL3: + case NAU8821_R51_FUSE_CTRL1: + case NAU8821_R53_OTPDOUT_1 ... NAU8821_R55_MISC_CTRL: + case NAU8821_R58_I2C_DEVICE_ID ... NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R66_BIAS_ADJ: + case NAU8821_R68_TRIM_SETTINGS ... NAU8821_R6B_PGA_MUTE: + case NAU8821_R71_ANALOG_ADC_1 ... NAU8821_R74_MIC_BIAS: + case NAU8821_R76_BOOST ... NAU8821_R77_FEPGA: + case NAU8821_R7E_PGA_GAIN ... NAU8821_R82_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static bool nau8821_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET ... NAU8821_R01_ENA_CTRL: + case NAU8821_R03_CLK_DIVIDER ... NAU8821_R0B_FLL8: + case NAU8821_R0D_JACK_DET_CTRL: + case NAU8821_R0F_INTERRUPT_MASK: + case NAU8821_R11_INT_CLR_KEY_STATUS ... NAU8821_R13_DMIC_CTRL: + case NAU8821_R1A_GPIO12_CTRL ... NAU8821_R1F_RIGHT_TIME_SLOT: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2D_DAC_CTRL2: + case NAU8821_R2F_DAC_DGAIN_CTRL ... NAU8821_R32_HSVOL_CTRL: + case NAU8821_R34_DACR_CTRL ... NAU8821_R3D_DAC_DRC_ATKDCY: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4C_IMM_MODE_CTRL: + case NAU8821_R4E_FUSE_CTRL2 ... NAU8821_R4F_FUSE_CTRL3: + case NAU8821_R51_FUSE_CTRL1: + case NAU8821_R55_MISC_CTRL: + case NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R66_BIAS_ADJ: + case NAU8821_R68_TRIM_SETTINGS ... NAU8821_R6B_PGA_MUTE: + case NAU8821_R71_ANALOG_ADC_1 ... NAU8821_R74_MIC_BIAS: + case NAU8821_R76_BOOST ... NAU8821_R77_FEPGA: + case NAU8821_R7E_PGA_GAIN ... NAU8821_R80_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8821_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET: + case NAU8821_R10_IRQ_STATUS ... NAU8821_R11_INT_CLR_KEY_STATUS: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2A_BIQ0_COF10: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4A_BIQ1_COF10: + case NAU8821_R4D_IMM_RMS_L: + case NAU8821_R53_OTPDOUT_1 ... NAU8821_R54_OTPDOUT_2: + case NAU8821_R58_I2C_DEVICE_ID ... NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R81_CHARGE_PUMP_INPUT_READ ... NAU8821_R82_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8821_biq_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + if (!component->regmap) + return -EINVAL; + + regmap_raw_read(component->regmap, NAU8821_R41_BIQ1_COF1, + ucontrol->value.bytes.data, params->max); + + return 0; +} + +static int nau8821_biq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + + if (!component->regmap) + return -EINVAL; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + regmap_raw_write(component->regmap, NAU8821_R41_BIQ1_COF1, + data, params->max); + + kfree(data); + + return 0; +} + +static const char * const nau8821_adc_decimation[] = { + "32", "64", "128", "256" }; + +static const struct soc_enum nau8821_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8821_R2B_ADC_RATE, NAU8821_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8821_adc_decimation), nau8821_adc_decimation); + +static const char * const nau8821_dac_oversampl[] = { + "64", "256", "128", "", "32" }; + +static const struct soc_enum nau8821_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8821_dac_oversampl), nau8821_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -6600, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -900, 0); +static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -6600, 50, 1); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); +static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -7000, 2400); + +static const struct snd_kcontrol_new nau8821_controls[] = { + SOC_DOUBLE_TLV("Mic Volume", NAU8821_R35_ADC_DGAIN_CTRL1, + NAU8821_ADCL_CH_VOL_SFT, NAU8821_ADCR_CH_VOL_SFT, + 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8821_R30_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("Headphone Volume", NAU8821_R32_HSVOL_CTRL, + NAU8821_HPL_VOL_SFT, NAU8821_HPR_VOL_SFT, 0x3, 1, hp_vol_tlv), + SOC_DOUBLE_TLV("Digital Playback Volume", NAU8821_R34_DACR_CTRL, + NAU8821_DACL_CH_VOL_SFT, NAU8821_DACR_CH_VOL_SFT, + 0xcf, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Frontend PGA Volume", NAU8821_R7E_PGA_GAIN, + NAU8821_PGA_GAIN_L_SFT, NAU8821_PGA_GAIN_R_SFT, + 37, 0, fepga_gain_tlv), + SOC_DOUBLE_TLV("Headphone Crosstalk Volume", + NAU8821_R2F_DAC_DGAIN_CTRL, + 0, 8, 0xff, 0, crosstalk_vol_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8821_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8821_dac_oversampl_enum), + SND_SOC_BYTES_EXT("BIQ Coefficients", 20, + nau8821_biq_coeff_get, nau8821_biq_coeff_put), +}; + +static const struct snd_kcontrol_new nau8821_adc_ch0_dmic = + SOC_DAPM_SINGLE("Switch", NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_EN_SFT, 1, 0); + +static int dmic_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + int i, speed_selection, clk_adc_src, clk_adc; + unsigned int clk_divider_r03; + + /* The DMIC clock is gotten from adc clock divided by + * CLK_DMIC_SRC (1, 2, 4, 8). The clock has to be equal or + * less than nau8821->dmic_clk_threshold. + */ + regmap_read(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + &clk_divider_r03); + clk_adc_src = (clk_divider_r03 & NAU8821_CLK_ADC_SRC_MASK) + >> NAU8821_CLK_ADC_SRC_SFT; + + switch (clk_adc_src) { + case 0: + clk_adc = nau8821->fs * 256; + break; + case 1: + clk_adc = (nau8821->fs * 256) >> 1; + break; + case 2: + clk_adc = (nau8821->fs * 256) >> 2; + break; + case 3: + clk_adc = (nau8821->fs * 256) >> 3; + break; + } + + for (i = 0 ; i < 4 ; i++) { + if ((clk_adc >> dmic_speed_sel[i].param) <= + nau8821->dmic_clk_threshold) { + speed_selection = dmic_speed_sel[i].val; + break; + } + } + + dev_dbg(nau8821->dev, + "clk_adc=%d, dmic_clk_threshold = %d, param=%d, val = %d\n", + clk_adc, nau8821->dmic_clk_threshold, + dmic_speed_sel[i].param, dmic_speed_sel[i].val); + regmap_update_bits(nau8821->regmap, NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_SRC_MASK, + (speed_selection << NAU8821_DMIC_SRC_SFT)); + + return 0; +} + +static int nau8821_left_adc_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 nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(125); + regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCL, NAU8821_EN_ADCL); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, + NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCL, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_right_adc_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 nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(125); + regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCR, NAU8821_EN_ADCR); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, + NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_pump_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 nau8821 *nau8821 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(20); + regmap_update_bits(nau8821->regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_JAMNODCLOW, NAU8821_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8821->regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_output_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 nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8821->regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_TESTDAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_TESTDAC_EN, NAU8821_BIAS_TESTDAC_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget nau8821_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_POWERUP_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA L", NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_PGA_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Frontend PGA R", NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_PGA_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCL Power", NAU8821_R72_ANALOG_ADC_2, + NAU8821_POWERUP_ADCL_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCR Power", NAU8821_R72_ANALOG_ADC_2, + NAU8821_POWERUP_ADCR_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Clock", SND_SOC_NOPM, 0, 0, + dmic_clock_control, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCL", NULL, SND_SOC_NOPM, 0, 0, + nau8821_left_adc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADCR", NULL, SND_SOC_NOPM, 0, 0, + nau8821_right_adc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("DMIC Enable", SND_SOC_NOPM, + 0, 0, &nau8821_adc_ch0_dmic), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_TRISTATE_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8821_R73_RDAC, + NAU8821_DACL_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8821_R73_RDAC, + NAU8821_DACR_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL Clock", 3, NAU8821_R73_RDAC, + NAU8821_DACL_CLK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR Clock", 3, NAU8821_R73_RDAC, + NAU8821_DACR_CLK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8821_R01_ENA_CTRL, + NAU8821_EN_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8821_R01_ENA_CTRL, + NAU8821_EN_DACL_SFT, 0), + SND_SOC_DAPM_PGA_S("HP amp L", 0, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_LDAC_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP amp R", 0, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_RDAC_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8821_R80_CHARGE_PUMP, + NAU8821_CHANRGE_PUMP_EN_SFT, 0, nau8821_pump_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA_S("Output Driver R Stage 1", 4, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_INTEG_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 1", 4, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_INTEG_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 2", 5, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_DRV_INSTG_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 2", 5, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_DRV_INSTG_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 6, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_MAIN_DRV_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 6, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_MAIN_DRV_L_SFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 7, + NAU8821_R80_CHARGE_PUMP, NAU8821_POWER_DOWN_DACL_SFT, + 0, nau8821_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 7, + NAU8821_R80_CHARGE_PUMP, NAU8821_POWER_DOWN_DACR_SFT, + 0, nau8821_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* HPOL/R are ungrounded by disabling 16 Ohm pull-downs on playback */ + SND_SOC_DAPM_PGA_S("HPOL Pulldown", 8, + NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HPOR Pulldown", 8, + NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R_SFT, 0, NULL, 0), + + /* High current HPOL/R boost driver */ + SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, + NAU8821_R76_BOOST, NAU8821_HP_BOOST_DIS_SFT, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Class G", NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8821_dapm_routes[] = { + {"Frontend PGA L", NULL, "MICL"}, + {"Frontend PGA R", NULL, "MICR"}, + + {"DMIC Enable", "Switch", "DMIC"}, + {"DMIC Enable", NULL, "DMIC Clock"}, + {"ADCL", NULL, "DMIC Enable"}, + {"ADCR", NULL, "DMIC Enable"}, + + {"Frontend PGA L", NULL, "ADCL Power"}, + {"Frontend PGA R", NULL, "ADCR Power"}, + {"Frontend PGA L", NULL, "MICBIAS"}, + {"Frontend PGA R", NULL, "MICBIAS"}, + + {"ADCL", NULL, "Frontend PGA L"}, + {"ADCR", NULL, "Frontend PGA R"}, + + {"AIFTX", NULL, "ADCL"}, + {"AIFTX", NULL, "ADCR"}, + {"AIFTX", NULL, "DMIC Enable"}, + + {"DDACL", NULL, "AIFRX"}, + {"DDACR", NULL, "AIFRX"}, + + {"HP amp L", NULL, "DDACL"}, + {"HP amp R", NULL, "DDACR"}, + {"Charge Pump", NULL, "HP amp L"}, + {"Charge Pump", NULL, "HP amp R"}, + {"ADACL", NULL, "Charge Pump"}, + {"ADACR", NULL, "Charge Pump"}, + {"ADACL Clock", NULL, "ADACL"}, + {"ADACR Clock", NULL, "ADACR"}, + {"Output Driver L Stage 1", NULL, "ADACL Clock"}, + {"Output Driver R Stage 1", NULL, "ADACR Clock"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL Pulldown", NULL, "Output DACL"}, + {"HPOR Pulldown", NULL, "Output DACR"}, + {"HP Boost Driver", NULL, "HPOL Pulldown"}, + {"HP Boost Driver", NULL, "HPOR Pulldown"}, + {"Class G", NULL, "HP Boost Driver"}, + {"HPOL", NULL, "Class G"}, + {"HPOR", NULL, "Class G"}, +}; + +static int nau8821_clock_check(struct nau8821 *nau8821, + int stream, int rate, int osr) +{ + int osrate = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (osr >= ARRAY_SIZE(osr_dac_sel)) + return -EINVAL; + osrate = osr_dac_sel[osr].osr; + } else { + if (osr >= ARRAY_SIZE(osr_adc_sel)) + return -EINVAL; + osrate = osr_adc_sel[osr].osr; + } + + if (!osrate || rate * osrate > CLK_DA_AD_MAX) { + dev_err(nau8821->dev, + "exceed the maximum frequency of CLK_ADC or CLK_DAC\n"); + return -EINVAL; + } + + return 0; +} + +static int nau8821_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 nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, clk_div; + + nau8821_sema_acquire(nau8821, HZ); + nau8821->fs = params_rate(params); + /* CLK_DAC or CLK_ADC = OSR * FS + * DAC or ADC clock frequency is defined as Over Sampling Rate (OSR) + * multiplied by the audio sample rate (Fs). Note that the OSR and Fs + * values must be selected such that the maximum frequency is less + * than 6.144 MHz. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8821->regmap, NAU8821_R2C_DAC_CTRL1, &osr); + osr &= NAU8821_DAC_OVERSAMPLE_MASK; + if (nau8821_clock_check(nau8821, substream->stream, + nau8821->fs, osr)) { + nau8821_sema_release(nau8821); + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_DAC_SRC_MASK, + osr_dac_sel[osr].clk_src << NAU8821_CLK_DAC_SRC_SFT); + } else { + regmap_read(nau8821->regmap, NAU8821_R2B_ADC_RATE, &osr); + osr &= NAU8821_ADC_SYNC_DOWN_MASK; + if (nau8821_clock_check(nau8821, substream->stream, + nau8821->fs, osr)) { + nau8821_sema_release(nau8821); + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_ADC_SRC_MASK, + osr_adc_sel[osr].clk_src << NAU8821_CLK_ADC_SRC_SFT); + } + + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, &ctrl_val); + if (ctrl_val & NAU8821_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / nau8821->fs; + if (bclk_fs <= 32) + clk_div = 3; + else if (bclk_fs <= 64) + clk_div = 2; + else if (bclk_fs <= 128) + clk_div = 1; + else { + nau8821_sema_release(nau8821); + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_LRC_DIV_MASK | NAU8821_I2S_BLK_DIV_MASK, + (clk_div << NAU8821_I2S_LRC_DIV_SFT) | clk_div); + } + + switch (params_width(params)) { + case 16: + val_len |= NAU8821_I2S_DL_16; + break; + case 20: + val_len |= NAU8821_I2S_DL_20; + break; + case 24: + val_len |= NAU8821_I2S_DL_24; + break; + case 32: + val_len |= NAU8821_I2S_DL_32; + break; + default: + nau8821_sema_release(nau8821); + return -EINVAL; + } + + regmap_update_bits(nau8821->regmap, NAU8821_R1C_I2S_PCM_CTRL1, + NAU8821_I2S_DL_MASK, val_len); + + nau8821_sema_release(nau8821); + + return 0; +} + +static int nau8821_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8821_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8821_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8821_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8821_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8821_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8821_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8821_I2S_DF_PCM_AB; + ctrl1_val |= NAU8821_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + nau8821_sema_acquire(nau8821, HZ); + + regmap_update_bits(nau8821->regmap, NAU8821_R1C_I2S_PCM_CTRL1, + NAU8821_I2S_DL_MASK | NAU8821_I2S_DF_MASK | + NAU8821_I2S_BP_MASK | NAU8821_I2S_PCMB_MASK, ctrl1_val); + regmap_update_bits(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, ctrl2_val); + + nau8821_sema_release(nau8821); + + return 0; +} + +static int nau8821_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int val = 0; + + if (mute) + val = NAU8821_DAC_SOFT_MUTE; + + return regmap_update_bits(nau8821->regmap, + NAU8821_R31_MUTE_CTRL, NAU8821_DAC_SOFT_MUTE, val); +} + +static const struct snd_soc_dai_ops nau8821_dai_ops = { + .hw_params = nau8821_hw_params, + .set_fmt = nau8821_set_dai_fmt, + .mute_stream = nau8821_digital_mute, +}; + +#define NAU8821_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8821_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8821_dai = { + .name = NUVOTON_CODEC_DAI, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8821_RATES, + .formats = NAU8821_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8821_RATES, + .formats = NAU8821_FORMATS, + }, + .ops = &nau8821_dai_ops, +}; + + +static bool nau8821_is_jack_inserted(struct regmap *regmap) +{ + bool active_high, is_high; + int status, jkdet; + + regmap_read(regmap, NAU8821_R0D_JACK_DET_CTRL, &jkdet); + active_high = jkdet & NAU8821_JACK_POLARITY; + regmap_read(regmap, NAU8821_R82_GENERAL_STATUS, &status); + is_high = status & NAU8821_GPIO2_IN; + /* return jack connection status according to jack insertion logic + * active high or active low. + */ + return active_high == is_high; +} + +static void nau8821_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8821_R10_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8821_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8821_R11_INT_CLR_KEY_STATUS, clear_irq); + } +} + +static void nau8821_eject_jack(struct nau8821 *nau8821) +{ + struct snd_soc_dapm_context *dapm = nau8821->dapm; + struct regmap *regmap = nau8821->regmap; + + /* Reset semaphore */ + nau8821_sema_reset(nau8821); + + /* Detach 2kOhm Resistors from MICBIAS to MICGND */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, 0); + /* HPL/HPR short to ground */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R | NAU8821_SPKR_DWN1L, 0); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8821_int_status_clear_all(regmap); + + /* Enable the insertion interruption, disable the ejection inter- + * ruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_EJECT_DIS | NAU8821_IRQ_INSERT_DIS, + NAU8821_IRQ_EJECT_DIS); + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_EJECT_EN); + + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, NAU8821_JACK_DET_DB_BYPASS); + + /* Close clock for jack type detection at manual mode */ + if (dapm->bias_level < SND_SOC_BIAS_PREPARE) + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + + /* Recover to normal channel input */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_R_SRC_EN, 0); +} + +static void nau8821_jdet_work(struct work_struct *work) +{ + struct nau8821 *nau8821 = + container_of(work, struct nau8821, jdet_work); + struct snd_soc_dapm_context *dapm = nau8821->dapm; + struct regmap *regmap = nau8821->regmap; + int jack_status_reg, mic_detected, event = 0, event_mask = 0; + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + msleep(20); + + regmap_read(regmap, NAU8821_R58_I2C_DEVICE_ID, &jack_status_reg); + mic_detected = !(jack_status_reg & NAU8821_KEYDET); + + if (mic_detected) { + dev_dbg(nau8821->dev, "Headset connected\n"); + event |= SND_JACK_HEADSET; + + /* 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, NAU8821_MICBIAS_JKR2); + /* Latch Right Channel Analog data + * input into the Right Channel Filter + */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_R_SRC_EN, NAU8821_ADC_R_SRC_EN); + } else { + dev_dbg(nau8821->dev, "Headphone connected\n"); + event |= SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); + } + event_mask |= SND_JACK_HEADSET; + snd_soc_jack_report(nau8821->jack, event, event_mask); + nau8821_sema_release(nau8821); +} + +/* Enable interruptions with internal clock. */ +static void nau8821_setup_inserted_irq(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Enable internal VCO needed for interruptions */ + if (nau8821->dapm->bias_level < SND_SOC_BIAS_PREPARE) + nau8821_configure_sysclk(nau8821, NAU8821_CLK_INTERNAL, 0); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, NAU8821_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, NAU8821_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, 0); + + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN, 0); + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_EJECT_DIS, 0); +} + +static irqreturn_t nau8821_interrupt(int irq, void *data) +{ + struct nau8821 *nau8821 = (struct nau8821 *)data; + struct regmap *regmap = nau8821->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + if (regmap_read(regmap, NAU8821_R10_IRQ_STATUS, &active_irq)) { + dev_err(nau8821->dev, "failed to read irq status\n"); + return IRQ_NONE; + } + + dev_dbg(nau8821->dev, "IRQ %d\n", active_irq); + + if ((active_irq & NAU8821_JACK_EJECT_IRQ_MASK) == + NAU8821_JACK_EJECT_DETECTED) { + regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, + NAU8821_MICDET_MASK, NAU8821_MICDET_DIS); + nau8821_eject_jack(nau8821); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8821_JACK_EJECT_IRQ_MASK; + } else if ((active_irq & NAU8821_JACK_INSERT_IRQ_MASK) == + NAU8821_JACK_INSERT_DETECTED) { + regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, + NAU8821_MICDET_MASK, NAU8821_MICDET_EN); + if (nau8821_is_jack_inserted(regmap)) { + /* detect microphone and jack type */ + cancel_work_sync(&nau8821->jdet_work); + schedule_work(&nau8821->jdet_work); + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_INSERT_DIS, + NAU8821_IRQ_INSERT_DIS); + regmap_update_bits(regmap, + NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_INSERT_EN); + nau8821_setup_inserted_irq(nau8821); + } else { + dev_warn(nau8821->dev, + "Inserted IRQ fired but not connected\n"); + nau8821_eject_jack(nau8821); + } + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8821_R11_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8821->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static const struct regmap_config nau8821_regmap_config = { + .val_bits = NAU8821_REG_DATA_LEN, + .reg_bits = NAU8821_REG_ADDR_LEN, + + .max_register = NAU8821_REG_MAX, + .readable_reg = nau8821_readable_reg, + .writeable_reg = nau8821_writeable_reg, + .volatile_reg = nau8821_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8821_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8821_reg_defaults), +}; + +static int nau8821_component_probe(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + nau8821->dapm = dapm; + + return 0; +} + +static void nau8821_component_remove(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (nau8821->irq) + /* Reset semaphore */ + nau8821_sema_reset(nau8821); +} + +/** + * nau8821_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau8821_calc_fll_param(unsigned int fll_in, + unsigned int fs, struct nau8821_fll *fll_param) +{ + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by + * dividing freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8821_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in >> fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } + } + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; + + /* Calculate the FLL 10-bit integer input and the FLL 24-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco_max << 24, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 24) & 0x3ff; + fll_param->fll_frac = fvco & 0xffffff; + + return 0; +} + +static void nau8821_fll_apply(struct nau8821 *nau8821, + struct nau8821_fll *fll_param) +{ + struct regmap *regmap = nau8821->regmap; + + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK | NAU8821_CLK_MCLK_SRC_MASK, + NAU8821_CLK_SRC_MCLK | fll_param->mclk_src); + /* Make DSP operate at high speed for better performance. */ + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_FLL_RATIO_MASK | NAU8821_ICTRL_LATCH_MASK, + fll_param->ratio | (0x6 << NAU8821_ICTRL_LATCH_SFT)); + /* FLL 24-bit fractional input */ + regmap_write(regmap, NAU8821_R0A_FLL7, + (fll_param->fll_frac >> 16) & 0xff); + regmap_write(regmap, NAU8821_R0B_FLL8, fll_param->fll_frac & 0xffff); + /* FLL 10-bit integer input */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(regmap, NAU8821_R07_FLL4, + NAU8821_HIGHBW_EN | NAU8821_FLL_REF_DIV_MASK, + NAU8821_HIGHBW_EN | + (fll_param->clk_ref_div << NAU8821_FLL_REF_DIV_SFT)); + /* select divided VCO input */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_CLK_SW_MASK, NAU8821_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(regmap, + NAU8821_R09_FLL6, NAU8821_DCO_EN, 0); + if (fll_param->fll_frac) { + /* set FLL loop filter enable and cutoff frequency at 500Khz */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_MASK, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_FILTER); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN | NAU8821_CUTOFF500, + NAU8821_SDM_EN | NAU8821_CUTOFF500); + } else { + /* disable FLL loop filter and cutoff frequency */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_MASK, NAU8821_FLL_FTR_SW_ACCU); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN | NAU8821_CUTOFF500, 0); + } +} + +/** + * nau8821_set_fll - FLL configuration of nau8821 + * @codec: codec component + * @freq_in: frequency of input clock source + * @freq_out: must be 256*Fs in order to achieve the best performance + * + * The FLL function can select BCLK or MCLK as the input clock source. + * + * Returns 0 if the parameters have been applied successfully + * or negative error code. + */ +static int nau8821_set_fll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, unsigned int freq_out) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct nau8821_fll fll_set_param, *fll_param = &fll_set_param; + int ret, fs; + + fs = freq_out >> 8; + ret = nau8821_calc_fll_param(freq_in, fs, fll_param); + if (ret) { + dev_err(nau8821->dev, + "Unsupported input clock %d to output clock %d\n", + freq_in, freq_out); + return ret; + } + dev_dbg(nau8821->dev, + "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param->mclk_src, fll_param->ratio, fll_param->fll_frac, + fll_param->fll_int, fll_param->clk_ref_div); + + nau8821_fll_apply(nau8821, fll_param); + mdelay(2); + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_VCO); + + return 0; +} + +static void nau8821_configure_mclk_as_sysclk(struct regmap *regmap) +{ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_DCO_EN, 0); + /* Make DSP operate as default setting for power saving. */ + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_ICTRL_LATCH_MASK, 0); +} + +static int nau8821_configure_sysclk(struct nau8821 *nau8821, + int clk_id, unsigned int freq) +{ + struct regmap *regmap = nau8821->regmap; + + switch (clk_id) { + case NAU8821_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + nau8821_configure_mclk_as_sysclk(regmap); + break; + case NAU8821_CLK_MCLK: + nau8821_sema_acquire(nau8821, HZ); + nau8821_configure_mclk_as_sysclk(regmap); + /* MCLK not changed by clock tree */ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_MCLK_SRC_MASK, 0); + nau8821_sema_release(nau8821); + break; + case NAU8821_CLK_INTERNAL: + if (nau8821_is_jack_inserted(regmap)) { + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_DCO_EN, NAU8821_DCO_EN); + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_VCO); + /* Decrease the VCO frequency and make DSP operate + * as default setting for power saving. + */ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_ICTRL_LATCH_MASK | + NAU8821_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN, NAU8821_SDM_EN); + } + break; + case NAU8821_CLK_FLL_MCLK: + nau8821_sema_acquire(nau8821, HZ); + /* Higher FLL reference input frequency can only set lower + * gain error, such as 0000 for input reference from MCLK + * 12.288Mhz. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_MCLK | 0); + nau8821_sema_release(nau8821); + break; + case NAU8821_CLK_FLL_BLK: + nau8821_sema_acquire(nau8821, HZ); + /* If FLL reference input is from low frequency source, + * higher error gain can apply such as 0xf which has + * the most sensitive gain error correction threshold, + * Therefore, FLL has the most accurate DCO to + * target frequency. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_BLK | + (0xf << NAU8821_GAIN_ERR_SFT)); + nau8821_sema_release(nau8821); + break; + case NAU8821_CLK_FLL_FS: + nau8821_sema_acquire(nau8821, HZ); + /* If FLL reference input is from low frequency source, + * higher error gain can apply such as 0xf which has + * the most sensitive gain error correction threshold, + * Therefore, FLL has the most accurate DCO to + * target frequency. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_FS | + (0xf << NAU8821_GAIN_ERR_SFT)); + nau8821_sema_release(nau8821); + break; + default: + dev_err(nau8821->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + nau8821->clk_id = clk_id; + dev_dbg(nau8821->dev, "Sysclk is %dHz and clock id is %d\n", freq, + nau8821->clk_id); + + return 0; +} + +static int nau8821_set_sysclk(struct snd_soc_component *component, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + return nau8821_configure_sysclk(nau8821, clk_id, freq); +} + +static int nau8821_resume_setup(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + if (nau8821->irq) { + /* Clear all interruption status */ + nau8821_int_status_clear_all(regmap); + + /* Enable both insertion and ejection interruptions, and then + * bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, 0); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, + NAU8821_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_INSERT_DIS | NAU8821_IRQ_EJECT_DIS, 0); + } + + return 0; +} + +static int nau8821_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct regmap *regmap = nau8821->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + /* Setup codec configuration after resume */ + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) + nau8821_resume_setup(nau8821); + break; + + case SND_SOC_BIAS_OFF: + /* HPL/HPR short to ground */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R | NAU8821_SPKR_DWN1L, 0); + if (nau8821->irq) { + /* Reset semaphore */ + nau8821_sema_reset(nau8821); + /* Reset the configuration of jack type for detection. + * Detach 2kOhm Resistors from MICBIAS to MICGND1/2. + */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, 0); + /* Turn off all interruptions before system shutdown. + * Keep theinterruption quiet before resume + * setup completes. + */ + regmap_write(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, 0xffff); + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN); + } + break; + } + + return 0; +} + +static int __maybe_unused nau8821_suspend(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (nau8821->irq) + disable_irq(nau8821->irq); + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8821->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8821->dapm); + regcache_cache_only(nau8821->regmap, true); + regcache_mark_dirty(nau8821->regmap); + + return 0; +} + +static int __maybe_unused nau8821_resume(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(nau8821->regmap, false); + regcache_sync(nau8821->regmap); + if (nau8821->irq) { + /* Hold semaphore to postpone playback happening + * until jack detection done. + */ + nau8821_sema_acquire(nau8821, 0); + enable_irq(nau8821->irq); + } + + return 0; +} + +static const struct snd_soc_component_driver nau8821_component_driver = { + .probe = nau8821_component_probe, + .remove = nau8821_component_remove, + .set_sysclk = nau8821_set_sysclk, + .set_pll = nau8821_set_fll, + .set_bias_level = nau8821_set_bias_level, + .suspend = nau8821_suspend, + .resume = nau8821_resume, + .controls = nau8821_controls, + .num_controls = ARRAY_SIZE(nau8821_controls), + .dapm_widgets = nau8821_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8821_dapm_widgets), + .dapm_routes = nau8821_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8821_dapm_routes), + .suspend_bias_off = 1, + .non_legacy_dai_naming = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, +}; + +/** + * nau8821_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8821_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + int ret; + + nau8821->jack = jack; + /* Initiate jack detection work queue */ + INIT_WORK(&nau8821->jdet_work, nau8821_jdet_work); + ret = devm_request_threaded_irq(nau8821->dev, nau8821->irq, NULL, + nau8821_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8821", nau8821); + if (ret) { + dev_err(nau8821->dev, "Cannot request irq %d (%d)\n", + nau8821->irq, ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(nau8821_enable_jack_detect); + +static void nau8821_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8821_R00_RESET, 0xffff); + regmap_write(regmap, NAU8821_R00_RESET, 0xffff); +} + +static void nau8821_print_device_properties(struct nau8821 *nau8821) +{ + struct device *dev = nau8821->dev; + + dev_dbg(dev, "jkdet-enable: %d\n", nau8821->jkdet_enable); + dev_dbg(dev, "jkdet-pull-enable: %d\n", nau8821->jkdet_pull_enable); + dev_dbg(dev, "jkdet-pull-up: %d\n", nau8821->jkdet_pull_up); + dev_dbg(dev, "jkdet-polarity: %d\n", nau8821->jkdet_polarity); + dev_dbg(dev, "micbias-voltage: %d\n", nau8821->micbias_voltage); + dev_dbg(dev, "vref-impedance: %d\n", nau8821->vref_impedance); + dev_dbg(dev, "jack-insert-debounce: %d\n", + nau8821->jack_insert_debounce); + dev_dbg(dev, "jack-eject-debounce: %d\n", + nau8821->jack_eject_debounce); + dev_dbg(dev, "dmic-clk-threshold: %d\n", + nau8821->dmic_clk_threshold); +} + +static int nau8821_read_device_properties(struct device *dev, + struct nau8821 *nau8821) +{ + int ret; + + nau8821->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8821->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8821->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8821->jkdet_polarity); + if (ret) + nau8821->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8821->micbias_voltage); + if (ret) + nau8821->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8821->vref_impedance); + if (ret) + nau8821->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8821->jack_insert_debounce); + if (ret) + nau8821->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8821->jack_eject_debounce); + if (ret) + nau8821->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,dmic-clk-threshold", + &nau8821->dmic_clk_threshold); + if (ret) + nau8821->dmic_clk_threshold = 3072000; + + return 0; +} + +static void nau8821_init_regs(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Enable Bias/Vmid */ + regmap_update_bits(regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_VMID, NAU8821_BIAS_VMID); + regmap_update_bits(regmap, NAU8821_R76_BOOST, + NAU8821_GLOBAL_BIAS_EN, NAU8821_GLOBAL_BIAS_EN); + /* VMID Tieoff setting and enable TESTDAC. + * This sets the analog DAC inputs to a '0' input signal to avoid + * any glitches due to power up transients in both the analog and + * digital DAC circuit. + */ + regmap_update_bits(regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_VMID_SEL_MASK | NAU8821_BIAS_TESTDAC_EN, + (nau8821->vref_impedance << NAU8821_BIAS_VMID_SEL_SFT) | + NAU8821_BIAS_TESTDAC_EN); + /* Disable short Frame Sync detection logic */ + regmap_update_bits(regmap, NAU8821_R1E_LEFT_TIME_SLOT, + NAU8821_DIS_FS_SHORT_DET, NAU8821_DIS_FS_SHORT_DET); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8821_R76_BOOST, + NAU8821_PRECHARGE_DIS | NAU8821_HP_BOOST_DIS | + NAU8821_HP_BOOST_G_DIS | NAU8821_SHORT_SHUTDOWN_EN, + NAU8821_PRECHARGE_DIS | NAU8821_HP_BOOST_DIS | + NAU8821_HP_BOOST_G_DIS | NAU8821_SHORT_SHUTDOWN_EN); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_TIMER_MASK, + 0x20 << NAU8821_CLASSG_TIMER_SFT); + /* Class AB bias current to 2x, DAC Capacitor enable MSB/LSB */ + regmap_update_bits(regmap, NAU8821_R6A_ANALOG_CONTROL_2, + NAU8821_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8821_DAC_CAPACITOR_MSB | NAU8821_DAC_CAPACITOR_LSB, + NAU8821_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8821_DAC_CAPACITOR_MSB | NAU8821_DAC_CAPACITOR_LSB); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_POWER_DOWN_DACR | NAU8821_POWER_DOWN_DACL, 0); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8821_R73_RDAC, + NAU8821_DAC_CLK_DELAY_MASK | NAU8821_DAC_VREF_MASK, + (0x2 << NAU8821_DAC_CLK_DELAY_SFT) | + (0x3 << NAU8821_DAC_VREF_SFT)); + + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_VOLTAGE_MASK, nau8821->micbias_voltage); + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_SYNC_DOWN_MASK, NAU8821_ADC_SYNC_DOWN_64); + regmap_update_bits(regmap, NAU8821_R2C_DAC_CTRL1, + NAU8821_DAC_OVERSAMPLE_MASK, NAU8821_DAC_OVERSAMPLE_64); +} + +static int nau8821_setup_irq(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + sema_init(&nau8821->jd_sem, 1); + + /* Jack detection */ + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_OUTPUT_EN, + nau8821->jkdet_enable ? 0 : NAU8821_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_PULL_EN, + nau8821->jkdet_pull_enable ? 0 : NAU8821_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_PULL_UP, + nau8821->jkdet_pull_up ? NAU8821_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8821->jkdet_polarity ? 0 : NAU8821_JACK_POLARITY); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_INSERT_DEBOUNCE_MASK, + nau8821->jack_insert_debounce << + NAU8821_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_EJECT_DEBOUNCE_MASK, + nau8821->jack_eject_debounce << + NAU8821_JACK_EJECT_DEBOUNCE_SFT); + /* Pull up IRQ pin */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_PIN_PULL_UP | NAU8821_IRQ_PIN_PULL_EN | + NAU8821_IRQ_OUTPUT_EN, NAU8821_IRQ_PIN_PULL_UP | + NAU8821_IRQ_PIN_PULL_EN | NAU8821_IRQ_OUTPUT_EN); + /* Disable interruption before codec initiation done */ + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, 0x3f5, 0x3f5); + + return 0; +} + +static int nau8821_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8821 *nau8821 = dev_get_platdata(&i2c->dev); + int ret, value; + + if (!nau8821) { + nau8821 = devm_kzalloc(dev, sizeof(*nau8821), GFP_KERNEL); + if (!nau8821) + return -ENOMEM; + nau8821_read_device_properties(dev, nau8821); + } + i2c_set_clientdata(i2c, nau8821); + + nau8821->regmap = devm_regmap_init_i2c(i2c, &nau8821_regmap_config); + if (IS_ERR(nau8821->regmap)) + return PTR_ERR(nau8821->regmap); + + nau8821->dev = dev; + nau8821->irq = i2c->irq; + nau8821_print_device_properties(nau8821); + + nau8821_reset_chip(nau8821->regmap); + ret = regmap_read(nau8821->regmap, NAU8821_R58_I2C_DEVICE_ID, &value); + if (ret) { + dev_err(dev, "Failed to read device id (%d)\n", ret); + return ret; + } + nau8821_init_regs(nau8821); + + if (i2c->irq) + nau8821_setup_irq(nau8821); + + ret = devm_snd_soc_register_component(&i2c->dev, + &nau8821_component_driver, &nau8821_dai, 1); + + return ret; +} + +static const struct i2c_device_id nau8821_i2c_ids[] = { + { "nau8821", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8821_i2c_ids); + +#ifdef CONFIG_OF +static const struct of_device_id nau8821_of_ids[] = { + { .compatible = "nuvoton,nau8821", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8821_of_ids); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8821_acpi_match[] = { + { "NVTN2020", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8821_acpi_match); +#endif + +static struct i2c_driver nau8821_driver = { + .driver = { + .name = "nau8821", + .of_match_table = of_match_ptr(nau8821_of_ids), + .acpi_match_table = ACPI_PTR(nau8821_acpi_match), + }, + .probe = nau8821_i2c_probe, + .id_table = nau8821_i2c_ids, +}; +module_i2c_driver(nau8821_driver); + +MODULE_DESCRIPTION("ASoC nau8821 driver"); +MODULE_AUTHOR("John Hsu <kchsu0(a)nuvoton.com>"); +MODULE_AUTHOR("Seven Lee <wtli(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h new file mode 100644 index 000000000000..3541decc34f0 --- /dev/null +++ b/sound/soc/codecs/nau8821.h @@ -0,0 +1,533 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * NAU88L21 ALSA SoC audio driver + * + * Copyright 2021 Nuvoton Technology Corp. + * Author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Lee <wtli(a)nuvoton.com> + */ + +#ifndef __NAU8821_H__ +#define __NAU8821_H__ + +#define NAU8821_R00_RESET 0x00 +#define NAU8821_R01_ENA_CTRL 0x01 +#define NAU8821_R03_CLK_DIVIDER 0x03 +#define NAU8821_R04_FLL1 0x04 +#define NAU8821_R05_FLL2 0x05 +#define NAU8821_R06_FLL3 0x06 +#define NAU8821_R07_FLL4 0x07 +#define NAU8821_R08_FLL5 0x08 +#define NAU8821_R09_FLL6 0x09 +#define NAU8821_R0A_FLL7 0x0a +#define NAU8821_R0B_FLL8 0x0b +#define NAU8821_R0D_JACK_DET_CTRL 0x0d +#define NAU8821_R0F_INTERRUPT_MASK 0x0f +#define NAU8821_R10_IRQ_STATUS 0x10 +#define NAU8821_R11_INT_CLR_KEY_STATUS 0x11 +#define NAU8821_R12_INTERRUPT_DIS_CTRL 0x12 +#define NAU8821_R13_DMIC_CTRL 0x13 +#define NAU8821_R1A_GPIO12_CTRL 0x1a +#define NAU8821_R1B_TDM_CTRL 0x1b +#define NAU8821_R1C_I2S_PCM_CTRL1 0x1c +#define NAU8821_R1D_I2S_PCM_CTRL2 0x1d +#define NAU8821_R1E_LEFT_TIME_SLOT 0x1e +#define NAU8821_R1F_RIGHT_TIME_SLOT 0x1f +#define NAU8821_R21_BIQ0_COF1 0x21 +#define NAU8821_R22_BIQ0_COF2 0x22 +#define NAU8821_R23_BIQ0_COF3 0x23 +#define NAU8821_R24_BIQ0_COF4 0x24 +#define NAU8821_R25_BIQ0_COF5 0x25 +#define NAU8821_R26_BIQ0_COF6 0x26 +#define NAU8821_R27_BIQ0_COF7 0x27 +#define NAU8821_R28_BIQ0_COF8 0x28 +#define NAU8821_R29_BIQ0_COF9 0x29 +#define NAU8821_R2A_BIQ0_COF10 0x2a +#define NAU8821_R2B_ADC_RATE 0x2b +#define NAU8821_R2C_DAC_CTRL1 0x2c +#define NAU8821_R2D_DAC_CTRL2 0x2d +#define NAU8821_R2F_DAC_DGAIN_CTRL 0x2f +#define NAU8821_R30_ADC_DGAIN_CTRL 0x30 +#define NAU8821_R31_MUTE_CTRL 0x31 +#define NAU8821_R32_HSVOL_CTRL 0x32 +#define NAU8821_R34_DACR_CTRL 0x34 +#define NAU8821_R35_ADC_DGAIN_CTRL1 0x35 +#define NAU8821_R36_ADC_DRC_KNEE_IP12 0x36 +#define NAU8821_R37_ADC_DRC_KNEE_IP34 0x37 +#define NAU8821_R38_ADC_DRC_SLOPES 0x38 +#define NAU8821_R39_ADC_DRC_ATKDCY 0x39 +#define NAU8821_R3A_DAC_DRC_KNEE_IP12 0x3a +#define NAU8821_R3B_DAC_DRC_KNEE_IP34 0x3b +#define NAU8821_R3C_DAC_DRC_SLOPES 0x3c +#define NAU8821_R3D_DAC_DRC_ATKDCY 0x3d +#define NAU8821_R41_BIQ1_COF1 0x41 +#define NAU8821_R42_BIQ1_COF2 0x42 +#define NAU8821_R43_BIQ1_COF3 0x43 +#define NAU8821_R44_BIQ1_COF4 0x44 +#define NAU8821_R45_BIQ1_COF5 0x45 +#define NAU8821_R46_BIQ1_COF6 0x46 +#define NAU8821_R47_BIQ1_COF7 0x47 +#define NAU8821_R48_BIQ1_COF8 0x48 +#define NAU8821_R49_BIQ1_COF9 0x49 +#define NAU8821_R4A_BIQ1_COF10 0x4a +#define NAU8821_R4B_CLASSG_CTRL 0x4b +#define NAU8821_R4C_IMM_MODE_CTRL 0x4c +#define NAU8821_R4D_IMM_RMS_L 0x4d +#define NAU8821_R4E_FUSE_CTRL2 0x4e +#define NAU8821_R4F_FUSE_CTRL3 0x4f +#define NAU8821_R51_FUSE_CTRL1 0x51 +#define NAU8821_R53_OTPDOUT_1 0x53 +#define NAU8821_R54_OTPDOUT_2 0x54 +#define NAU8821_R55_MISC_CTRL 0x55 +#define NAU8821_R58_I2C_DEVICE_ID 0x58 +#define NAU8821_R59_SARDOUT_RAM_STATUS 0x59 +#define NAU8821_R5A_SOFTWARE_RST 0x5a +#define NAU8821_R66_BIAS_ADJ 0x66 +#define NAU8821_R68_TRIM_SETTINGS 0x68 +#define NAU8821_R69_ANALOG_CONTROL_1 0x69 +#define NAU8821_R6A_ANALOG_CONTROL_2 0x6a +#define NAU8821_R6B_PGA_MUTE 0x6b +#define NAU8821_R71_ANALOG_ADC_1 0x71 +#define NAU8821_R72_ANALOG_ADC_2 0x72 +#define NAU8821_R73_RDAC 0x73 +#define NAU8821_R74_MIC_BIAS 0x74 +#define NAU8821_R76_BOOST 0x76 +#define NAU8821_R77_FEPGA 0x77 +#define NAU8821_R7E_PGA_GAIN 0x7e +#define NAU8821_R7F_POWER_UP_CONTROL 0x7f +#define NAU8821_R80_CHARGE_PUMP 0x80 +#define NAU8821_R81_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8821_R82_GENERAL_STATUS 0x82 +#define NAU8821_REG_MAX NAU8821_R82_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8821_REG_ADDR_LEN 16 +#define NAU8821_REG_DATA_LEN 16 + +/* ENA_CTRL (0x01) */ +#define NAU8821_CLK_DAC_INV_SFT 14 +#define NAU8821_CLK_DAC_INV (0x1 << NAU8821_CLK_DAC_INV) +#define NAU8821_EN_DACR_SFT 11 +#define NAU8821_EN_DACR (0x1 << NAU8821_EN_DACR_SFT) +#define NAU8821_EN_DACL_SFT 10 +#define NAU8821_EN_DACL (0x1 << NAU8821_EN_DACL_SFT) +#define NAU8821_EN_ADCR_SFT 9 +#define NAU8821_EN_ADCR (0x1 << NAU8821_EN_ADCR_SFT) +#define NAU8821_EN_ADCL_SFT 8 +#define NAU8821_EN_ADCL (0x1 << NAU8821_EN_ADCL_SFT) +#define NAU8821_EN_ADC_CLK_SFT 7 +#define NAU8821_EN_ADC_CLK (0x1 << NAU8821_EN_ADC_CLK_SFT) +#define NAU8821_EN_DAC_CLK_SFT 6 +#define NAU8821_EN_DAC_CLK (0x1 << NAU8821_EN_DAC_CLK_SFT) +#define NAU8821_EN_I2S_CLK_SFT 4 +#define NAU8821_EN_I2S_CLK (0x1 << NAU8821_EN_I2S_CLK_SFT) +#define NAU8821_EN_DRC_CLK_SFT 0 +#define NAU8821_EN_DRC_CLK (0x1 << NAU8821_EN_DRC_CLK_SFT) + +/* CLK_DIVIDER (0x03) */ +#define NAU8821_CLK_SRC_SFT 15 +#define NAU8821_CLK_SRC_MASK (0x1 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_SRC_VCO (0x1 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_SRC_MCLK (0x0 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_SFT 13 +#define NAU8821_CLK_CODEC_SRC_MASK (0x1 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_VCO (0x1 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_MCLK (0x0 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_ADC_SRC_SFT 6 +#define NAU8821_CLK_ADC_SRC_MASK (0x3 << NAU8821_CLK_ADC_SRC_SFT) +#define NAU8821_CLK_DAC_SRC_SFT 4 +#define NAU8821_CLK_DAC_SRC_MASK (0x3 << NAU8821_CLK_DAC_SRC_SFT) +#define NAU8821_CLK_MCLK_SRC_MASK 0xf + +/* FLL1 (0x04) */ +#define NAU8821_ICTRL_LATCH_SFT 10 +#define NAU8821_ICTRL_LATCH_MASK (0x7 << NAU8821_ICTRL_LATCH_SFT) +#define NAU8821_FLL_RATIO_MASK 0x7f + +/* FLL3 (0x06) */ +#define NAU8821_GAIN_ERR_SFT 12 +#define NAU8821_GAIN_ERR_MASK (0xf << NAU8821_GAIN_ERR_SFT) +#define NAU8821_FLL_CLK_SRC_SFT 10 +#define NAU8821_FLL_CLK_SRC_MASK (0x3 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_FS (0x3 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_BLK (0x2 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_MCLK (0x0 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_INTEGER_MASK 0x3ff + +/* FLL4 (0x07) */ +#define NAU8821_HIGHBW_EN_SFT 15 +#define NAU8821_HIGHBW_EN (0x1 << NAU8821_HIGHBW_EN_SFT) +#define NAU8821_FLL_REF_DIV_SFT 10 +#define NAU8821_FLL_REF_DIV_MASK (0x3 << NAU8821_FLL_REF_DIV_SFT) + +/* FLL5 (0x08) */ +#define NAU8821_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8821_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8821_FLL_CLK_SW_SFT 13 +#define NAU8821_FLL_CLK_SW_MASK (0x1 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_CLK_SW_N2 (0x1 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_CLK_SW_REF (0x0 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_FTR_SW_SFT 12 +#define NAU8821_FLL_FTR_SW_MASK (0x1 << NAU8821_FLL_FTR_SW_SFT) +#define NAU8821_FLL_FTR_SW_ACCU (0x1 << NAU8821_FLL_FTR_SW_SFT) +#define NAU8821_FLL_FTR_SW_FILTER (0x0 << NAU8821_FLL_FTR_SW_SFT) + +/* FLL6 (0x09) */ +#define NAU8821_DCO_EN (0x1 << 15) +#define NAU8821_SDM_EN (0x1 << 14) +#define NAU8821_CUTOFF500 (0x1 << 13) + +/* FLL7 (0x0a) */ +#define NAU8821_FLL_FRACH_MASK 0xff + +/* FLL8 (0x0b) */ +#define NAU8821_FLL_FRACL_MASK 0xffff + +/* JACK_DET_CTRL (0x0d) */ +/* 0 - open, 1 - short to GND */ +#define NAU8821_SPKR_DWN1R_SFT 15 +#define NAU8821_SPKR_DWN1R (0x1 << NAU8821_SPKR_DWN1R_SFT) +#define NAU8821_SPKR_DWN1L_SFT 14 +#define NAU8821_SPKR_DWN1L (0x1 << NAU8821_SPKR_DWN1L_SFT) +#define NAU8821_JACK_DET_RESTART (0x1 << 9) +#define NAU8821_JACK_DET_DB_BYPASS (0x1 << 8) +#define NAU8821_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8821_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8821_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8821_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8821_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8821_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8821_JACK_POLARITY (0x1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0x0f) */ +#define NAU8821_IRQ_PIN_PULL_UP (0x1 << 14) +#define NAU8821_IRQ_PIN_PULL_EN (0x1 << 13) +#define NAU8821_IRQ_OUTPUT_EN (0x1 << 11) +#define NAU8821_IRQ_RMS_EN (0x1 << 8) +#define NAU8821_IRQ_KEY_RELEASE_EN (0x1 << 7) +#define NAU8821_IRQ_KEY_PRESS_EN (0x1 << 6) +#define NAU8821_IRQ_MIC_DET_EN (0x1 << 4) +#define NAU8821_IRQ_EJECT_EN (0x1 << 2) +#define NAU8821_IRQ_INSERT_EN 0x1 + +/* IRQ_STATUS (0x10) */ +#define NAU8821_SHORT_CIRCUIT_IRQ (0x1 << 9) +#define NAU8821_IMPEDANCE_MEAS_IRQ (0x1 << 8) +#define NAU8821_KEY_IRQ_SFT 6 +#define NAU8821_KEY_IRQ_MASK (0x3 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_KEY_RELEASE_IRQ (0x2 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_KEY_SHORT_PRESS_IRQ (0x1 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_MIC_DETECT_IRQ (0x1 << 4) +#define NAU8821_JACK_EJECT_IRQ_MASK (0x3 << 2) +#define NAU8821_JACK_EJECT_DETECTED (0x1 << 2) +#define NAU8821_JACK_INSERT_IRQ_MASK 0x3 +#define NAU8821_JACK_INSERT_DETECTED 0x1 + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8821_IRQ_KEY_RELEASE_DIS (0x1 << 7) +#define NAU8821_IRQ_KEY_PRESS_DIS (0x1 << 6) +#define NAU8821_IRQ_MIC_DIS (0x1 << 4) +#define NAU8821_IRQ_EJECT_DIS (0x1 << 2) +#define NAU8821_IRQ_INSERT_DIS 0x1 + +/* DMIC_CTRL (0x13) */ +#define NAU8821_DMIC_DS_SFT 7 +#define NAU8821_DMIC_DS_MASK (0x1 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_DS_HIGH (0x1 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_DS_LOW (0x0 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_SRC_SFT 1 +#define NAU8821_DMIC_SRC_MASK (0x3 << NAU8821_DMIC_SRC_SFT) +#define NAU8821_CLK_DMIC_SRC (0x2 << NAU8821_DMIC_SRC_SFT) +#define NAU8821_DMIC_EN_SFT 0 + +/* GPIO12_CTRL (0x1a) */ +#define NAU8821_JKDET_PULL_UP (0x1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8821_JKDET_PULL_EN (0x1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8821_JKDET_OUTPUT_EN (0x1 << 8) /* 0 - enable input, 1 - enable output */ + +/* TDM_CTRL (0x1b) */ +#define NAU8821_TDM_EN_SFT 15 +#define NAU8821_TDM_EN (0x1 << NAU8821_TDM_EN_SFT) +#define NAU8821_DACL_CH_SFT 7 +#define NAU8821_DACL_CH_MASK (0x7 << NAU8821_DACL_CH_SFT) +#define NAU8821_DACR_CH_SFT 4 +#define NAU8821_DACR_CH_MASK (0x7 << NAU8821_DACR_CH_SFT) +#define NAU8821_ADCL_CH_SFT 2 +#define NAU8821_ADCL_CH_MASK (0x3 << NAU8821_ADCL_CH_SFT) +#define NAU8821_ADCR_CH_SFT 0 +#define NAU8821_ADCR_CH_MASK 0x3 + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8821_I2S_BP_SFT 7 +#define NAU8821_I2S_BP_MASK (0x1 << NAU8821_I2S_BP_SFT) +#define NAU8821_I2S_BP_INV (0x1 << NAU8821_I2S_BP_SFT) +#define NAU8821_I2S_PCMB_SFT 6 +#define NAU8821_I2S_PCMB_MASK (0x1 << NAU8821_I2S_PCMB_SFT) +#define NAU8821_I2S_PCMB_EN (0x1 << NAU8821_I2S_PCMB_SFT) +#define NAU8821_I2S_DL_SFT 2 +#define NAU8821_I2S_DL_MASK (0x3 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_32 (0x3 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_24 (0x2 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_20 (0x1 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_16 (0x0 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DF_MASK 0x3 +#define NAU8821_I2S_DF_PCM_AB 0x3 +#define NAU8821_I2S_DF_I2S 0x2 +#define NAU8821_I2S_DF_LEFT 0x1 +#define NAU8821_I2S_DF_RIGTH 0x0 + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8821_I2S_TRISTATE_SFT 15 +#define NAU8821_I2S_TRISTATE (0x1 << NAU8821_I2S_TRISTATE_SFT) +#define NAU8821_I2S_LRC_DIV_SFT 12 +#define NAU8821_I2S_LRC_DIV_MASK (0x3 << NAU8821_I2S_LRC_DIV_SFT) +#define NAU8821_I2S_MS_SFT 3 +#define NAU8821_I2S_MS_MASK (0x1 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_MS_MASTER (0x1 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_MS_SLAVE (0x0 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_BLK_DIV_MASK 0x7 + +/* LEFT_TIME_SLOT (0x1e) */ +#define NAU8821_TSLOT_L_OFFSET_MASK 0x3ff +#define NAU8821_DIS_FS_SHORT_DET (0x1 << 13) + +/* RIGHT_TIME_SLOT (0x1f) */ +#define NAU8821_TSLOT_R_OFFSET_MASK 0x3ff + +/* BIQ0_COF10 (0x2a) */ +#define NAU8821_BIQ0_ADC_EN_SFT 3 +#define NAU8821_BIQ0_ADC_EN_EN (0x1 << NAU8821_BIQ0_ADC_EN_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8821_ADC_SYNC_DOWN_SFT 0 +#define NAU8821_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8821_ADC_SYNC_DOWN_256 0x3 +#define NAU8821_ADC_SYNC_DOWN_128 0x2 +#define NAU8821_ADC_SYNC_DOWN_64 0x1 +#define NAU8821_ADC_SYNC_DOWN_32 0x0 +#define NAU8821_ADC_L_SRC_SFT 15 +#define NAU8821_ADC_L_SRC_EN (0x1 << NAU8821_ADC_L_SRC_SFT) +#define NAU8821_ADC_R_SRC_SFT 14 +#define NAU8821_ADC_R_SRC_EN (0x1 << NAU8821_ADC_R_SRC_SFT) + +/* DAC_CTRL1 (0x2c) */ +#define NAU8821_DAC_OVERSAMPLE_SFT 0 +#define NAU8821_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8821_DAC_OVERSAMPLE_32 0x4 +#define NAU8821_DAC_OVERSAMPLE_128 0x2 +#define NAU8821_DAC_OVERSAMPLE_256 0x1 +#define NAU8821_DAC_OVERSAMPLE_64 0x0 + +/* DAC_DGAIN_CTRL (0x2f) */ +#define NAU8821_DAC1_TO_DAC0_ST_SFT 8 +#define NAU8821_DAC1_TO_DAC0_ST_MASK (0xff << NAU8821_DAC1_TO_DAC0_ST_SFT) +#define NAU8821_DAC0_TO_DAC1_ST_SFT 0 +#define NAU8821_DAC0_TO_DAC1_ST_MASK 0xff + +/* MUTE_CTRL (0x31) */ +#define NAU8821_DAC_ZC_EN (0x1 << 12) +#define NAU8821_DAC_SOFT_MUTE (0x1 << 9) +#define NAU8821_ADC_ZC_EN (0x1 << 2) +#define NAU8821_ADC_SOFT_MUTE (0x1 << 1) + +/* HSVOL_CTRL (0x32) */ +#define NAU8821_HP_MUTE (0x1 << 15) +#define NAU8821_HP_MUTE_AUTO (0x1 << 14) +#define NAU8821_HPL_MUTE (0x1 << 13) +#define NAU8821_HPR_MUTE (0x1 << 12) +#define NAU8821_HPL_VOL_SFT 4 +#define NAU8821_HPL_VOL_MASK (0x3 << NAU8821_HPL_VOL_SFT) +#define NAU8821_HPR_VOL_SFT 0 +#define NAU8821_HPR_VOL_MASK (0x3 << NAU8821_HPR_VOL_SFT) + +/* DACR_CTRL (0x34) */ +#define NAU8821_DACR_CH_VOL_SFT 8 +#define NAU8821_DACR_CH_VOL_MASK (0xff << NAU8821_DACR_CH_VOL_SFT) +#define NAU8821_DACL_CH_VOL_SFT 0 +#define NAU8821_DACL_CH_VOL_MASK 0xff + +/* ADC_DGAIN_CTRL1 (0x35) */ +#define NAU8821_ADCR_CH_VOL_SFT 8 +#define NAU8821_ADCR_CH_VOL_MASK (0xff << NAU8821_ADCR_CH_VOL_SFT) +#define NAU8821_ADCL_CH_VOL_SFT 0 +#define NAU8821_ADCL_CH_VOL_MASK 0xff + +/* BIQ1_COF10 (0x4a) */ +#define NAU8821_BIQ1_DAC_EN_SFT 3 +#define NAU8821_BIQ1_DAC_EN_EN (0x1 << NAU8821_BIQ1_DAC_EN_SFT) + +/* CLASSG_CTRL (0x4b) */ +#define NAU8821_CLASSG_TIMER_SFT 8 +#define NAU8821_CLASSG_TIMER_MASK (0x3f << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_64MS (0x20 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_32MS (0x10 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_16MS (0x8 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_8MS (0x4 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_2MS (0x2 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_1MS (0x1 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_RDAC_EN_SFT 2 +#define NAU8821_CLASSG_RDAC_EN (0x1 << NAU8821_CLASSG_RDAC_EN_SFT) +#define NAU8821_CLASSG_LDAC_EN_SFT 1 +#define NAU8821_CLASSG_LDAC_EN (0x1 << NAU8821_CLASSG_LDAC_EN_SFT) +#define NAU8821_CLASSG_EN_SFT 0 +#define NAU8821_CLASSG_EN 0x1 + +/* IMM_MODE_CTRL (0x4C) */ +#define NAU8821_IMM_THD_SFT 8 +#define NAU8821_IMM_THD_MASK (0x3f << NAU8821_IMM_THD_SFT) +#define NAU8821_IMM_GEN_VOL_SFT 6 +#define NAU8821_IMM_GEN_VOL_MASK (0x3 << NAU8821_IMM_GEN_VOL_SFT) +#define NAU8821_IMM_CYC_SFT 4 +#define NAU8821_IMM_CYC_MASK (0x3 << NAU8821_IMM_CYC_SFT) +#define NAU8821_IMM_EN (0x1 << 3) +#define NAU8821_IMM_DAC_SRC_MASK 0x3 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8821_KEYDET (0x1 << 7) +#define NAU8821_MICDET (0x1 << 6) +#define NAU8821_SOFTWARE_ID_MASK 0x3 + +/* BIAS_ADJ (0x66) */ +#define NAU8821_BIAS_HP_IMP (0x1 << 15) +#define NAU8821_BIAS_TESTDAC_SFT 8 +#define NAU8821_BIAS_TESTDAC_EN (0x3 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_TESTDACR_EN (0x2 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_TESTDACL_EN (0x1 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_VMID (0x1 << 6) +#define NAU8821_BIAS_VMID_SEL_SFT 4 +#define NAU8821_BIAS_VMID_SEL_MASK (0x3 << NAU8821_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_1 (0x69) */ +#define NAU8821_JD_POL_SFT 2 +#define NAU8821_JD_POL_MASK (0x1 << NAU8821_JD_POL_SFT) +#define NAU8821_JD_POL_INV (0x1 << NAU8821_JD_POL_SFT) +#define NAU8821_JD_OUT_POL_SFT 1 +#define NAU8821_JD_OUT_POL_MASK (0x1 << NAU8821_JD_OUT_POL_SFT) +#define NAU8821_JD_OUT_POL_INV (0x1 << NAU8821_JD_OUT_POL_SFT) +#define NAU8821_JD_EN_SFT 0 +#define NAU8821_JD_EN 0x1 + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8821_HP_NON_CLASSG_CURRENT_2xADJ (0x1 << 12) +#define NAU8821_DAC_CAPACITOR_MSB (0x1 << 1) +#define NAU8821_DAC_CAPACITOR_LSB 0x1 + +/* ANALOG_ADC_1 (0x71) */ +#define NAU8821_MICDET_EN_SFT 0 +#define NAU8821_MICDET_MASK 0x1 +#define NAU8821_MICDET_DIS 0x1 +#define NAU8821_MICDET_EN 0x0 + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8821_ADC_VREFSEL_SFT 8 +#define NAU8821_ADC_VREFSEL_MASK (0x3 << NAU8821_ADC_VREFSEL_SFT) +#define NAU8821_POWERUP_ADCL_SFT 6 +#define NAU8821_POWERUP_ADCL (0x1 << NAU8821_POWERUP_ADCL_SFT) +#define NAU8821_POWERUP_ADCR_SFT 4 +#define NAU8821_POWERUP_ADCR (0x1 << NAU8821_POWERUP_ADCR_SFT) + +/* RDAC (0x73) */ +#define NAU8821_DACR_EN_SFT 13 +#define NAU8821_DACR_EN (0x3 << NAU8821_DACR_EN_SFT) +#define NAU8821_DACL_EN_SFT 12 +#define NAU8821_DACL_EN (0x3 << NAU8821_DACL_EN_SFT) +#define NAU8821_DACR_CLK_EN_SFT 9 +#define NAU8821_DACR_CLK_EN (0x3 << NAU8821_DACR_CLK_EN_SFT) +#define NAU8821_DACL_CLK_EN_SFT 8 +#define NAU8821_DACL_CLK_EN (0x3 << NAU8821_DACL_CLK_EN_SFT) +#define NAU8821_DAC_CLK_DELAY_SFT 4 +#define NAU8821_DAC_CLK_DELAY_MASK (0x7 << NAU8821_DAC_CLK_DELAY_SFT) +#define NAU8821_DAC_VREF_SFT 2 +#define NAU8821_DAC_VREF_MASK (0x3 << NAU8821_DAC_VREF_SFT) + +/* MIC_BIAS (0x74) */ +#define NAU8821_MICBIAS_JKR2 (0x1 << 12) +#define NAU8821_MICBIAS_POWERUP_SFT 8 +#define NAU8821_MICBIAS_VOLTAGE_SFT 0 +#define NAU8821_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8821_PRECHARGE_DIS (0x1 << 13) +#define NAU8821_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8821_HP_BOOST_DIS_SFT 9 +#define NAU8821_HP_BOOST_DIS (0x1 << NAU8821_HP_BOOST_DIS_SFT) +#define NAU8821_HP_BOOST_G_DIS (0x1 << 8) +#define NAU8821_SHORT_SHUTDOWN_EN (0x1 << 6) + +/* FEPGA (0x77) */ +#define NAU8821_FEPGA_MODEL_SFT 4 +#define NAU8821_FEPGA_MODEL_MASK (0xf << NAU8821_FEPGA_MODEL_SFT) +#define NAU8821_FEPGA_MODER_SFT 0 +#define NAU8821_FEPGA_MODER_MASK 0xf + +/* PGA_GAIN (0x7e) */ +#define NAU8821_PGA_GAIN_L_SFT 8 +#define NAU8821_PGA_GAIN_L_MASK (0x3f << NAU8821_PGA_GAIN_L_SFT) +#define NAU8821_PGA_GAIN_R_SFT 0 +#define NAU8821_PGA_GAIN_R_MASK 0x3f + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8821_PUP_PGA_L_SFT 15 +#define NAU8821_PUP_PGA_L (0x1 << NAU8821_PUP_PGA_L_SFT) +#define NAU8821_PUP_PGA_R_SFT 14 +#define NAU8821_PUP_PGA_R (0x1 << NAU8821_PUP_PGA_R_SFT) +#define NAU8821_PUP_INTEG_R_SFT 5 +#define NAU8821_PUP_INTEG_R (0x1 << NAU8821_PUP_INTEG_R_SFT) +#define NAU8821_PUP_INTEG_L_SFT 4 +#define NAU8821_PUP_INTEG_L (0x1 << NAU8821_PUP_INTEG_L_SFT) +#define NAU8821_PUP_DRV_INSTG_R_SFT 3 +#define NAU8821_PUP_DRV_INSTG_R (0x1 << NAU8821_PUP_DRV_INSTG_R_SFT) +#define NAU8821_PUP_DRV_INSTG_L_SFT 2 +#define NAU8821_PUP_DRV_INSTG_L (0x1 << NAU8821_PUP_DRV_INSTG_L_SFT) +#define NAU8821_PUP_MAIN_DRV_R_SFT 1 +#define NAU8821_PUP_MAIN_DRV_R (0x1 << NAU8821_PUP_MAIN_DRV_R_SFT) +#define NAU8821_PUP_MAIN_DRV_L_SFT 0 +#define NAU8821_PUP_MAIN_DRV_L 0x1 + +/* CHARGE_PUMP (0x80) */ +#define NAU8821_JAMNODCLOW (0x1 << 10) +#define NAU8821_POWER_DOWN_DACR_SFT 9 +#define NAU8821_POWER_DOWN_DACR (0x1 << NAU8821_POWER_DOWN_DACR_SFT) +#define NAU8821_POWER_DOWN_DACL_SFT 8 +#define NAU8821_POWER_DOWN_DACL (0x1 << NAU8821_POWER_DOWN_DACL_SFT) +#define NAU8821_CHANRGE_PUMP_EN_SFT 5 +#define NAU8821_CHANRGE_PUMP_EN (0x1 << NAU8821_CHANRGE_PUMP_EN_SFT) + +/* GENERAL_STATUS (0x82) */ +#define NAU8821_GPIO2_IN_SFT 1 +#define NAU8821_GPIO2_IN (0x1 << NAU8821_GPIO2_IN_SFT) + +#define NUVOTON_CODEC_DAI "nau8821-hifi" + +/* System Clock Source */ +enum { + NAU8821_CLK_DIS, + NAU8821_CLK_MCLK, + NAU8821_CLK_INTERNAL, + NAU8821_CLK_FLL_MCLK, + NAU8821_CLK_FLL_BLK, + NAU8821_CLK_FLL_FS, +}; + +struct nau8821 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct semaphore jd_sem; + struct work_struct jdet_work; + int irq; + int clk_id; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int jack_insert_debounce; + int jack_eject_debounce; + int fs; + int dmic_clk_threshold; +}; + +int nau8821_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif /* __NAU8821_H__ */ -- 2.25.1

4 5

[PATCH] ASoC: sh: rz-ssi: Fix wrong operator used issue
by Biju Das 17 Aug '21

17 Aug '21

Fix wrong operator used issue reported by Coverity by replacing | operator with & operator. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Reported-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rz-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 69b45ee2d445..f097c773d413 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -368,7 +368,7 @@ static int rz_ssi_stop(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) /* Wait for idle */ timeout = 100; while (--timeout) { - if (rz_ssi_reg_readl(ssi, SSISR) | SSISR_IIRQ) + if (rz_ssi_reg_readl(ssi, SSISR) & SSISR_IIRQ) break; udelay(1); } -- 2.17.1

2 1

[PATCH] ASoC: Intel: bytcr_rt5640: Use cfg-lineout:2 in the components string
by Hans de Goede 17 Aug '21

17 Aug '21

Use "cfg-lineout:2" in the components string on boards with a lineout instead of "cfg-lineout:1", this better mirrors the speaker part of the components string where we use "cfg-spk:1" for devices with a single speaker and "cfg-spk:2" for stereo speakers. The lineout is stereo by default, so using ":2" makes more sense, this way we keep ":1" reserved in case we ever encounter a device with a mono lineout. We can make this change without breaking userspace because no kernel has shipped with "cfg-lineout:1" in the component-string yet; and there also are no userspace bits (UCM profiles) checking for this yet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 369642c07a5d..f79234f49c7c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1688,7 +1688,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) if (byt_rt5640_quirk & BYT_RT5640_LINEOUT_AS_HP2) lineout_string = " cfg-hp2:lineout"; else - lineout_string = " cfg-lineout:1"; + lineout_string = " cfg-lineout:2"; } if (byt_rt5640_quirk & BYT_RT5640_HSMIC2_ON_IN1) -- 2.31.1

3 2

[PATCH 0/2] ASoC: tegra30: Fix use of of_device_get_match_data
by Aakash Hemadri 17 Aug '21

17 Aug '21

Hi, This patchset fixes my previous incorrect patches. 356b94a32a75 ("ASoC: tegra30: i2s: Use of_device_get_match_data") 80165bb80433 ("ASoC: tegra30: ahub: Use of_device_get_match_data") Aakash Hemadri (2): ASoC: tegra30: ahub: Fix incorrect usage of of_device_get_match_data ASoC: tegra30: i2s: Fix incorrect usage of of_device_get_match_data sound/soc/tegra/tegra30_ahub.c | 6 ++---- sound/soc/tegra/tegra30_i2s.c | 8 ++++---- 2 files changed, 6 insertions(+), 8 deletions(-) -- 2.32.0

3 5

[PATCH 0/2] ASoC: tegra: Use of_device_get_match_data
by Aakash Hemadri 17 Aug '21

17 Aug '21

Hi, Prefer `of_device_get_match_data` over `of_match_device`. This patch replaces of_match_device with of_device_get_match_data. This patch series will apply cleanly on for-next Aakash Hemadri (2): ASoC: tegra30: ahub: Use of_device_get_match_data ASoC: tegra30: i2s: Use of_device_get_match_data sound/soc/tegra/tegra30_ahub.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.32.0

4 11

[PATCH v2 0/2] alsa-lib patches for rawmidi tstamp framing
by David Henningsson 17 Aug '21

17 Aug '21

v2: Improved commit messages David Henningsson (2): Add rawmidi framing API Add test for rawmidi framing API include/rawmidi.h | 26 ++++++++++++ include/sound/uapi/asound.h | 30 +++++++++++++- src/rawmidi/rawmidi.c | 83 +++++++++++++++++++++++++++++++++++++ src/rawmidi/rawmidi_hw.c | 2 + src/rawmidi/rawmidi_local.h | 2 + test/rawmidi.c | 65 ++++++++++++++++++++++++++--- 6 files changed, 200 insertions(+), 8 deletions(-) -- 2.25.1

2 3

[PATCH 0/2] alsa-lib patches for rawmidi tstamp framing
by David Henningsson 17 Aug '21

17 Aug '21

Well, now that we're about to release a kernel with support for rawmidi tstamp framing, how about some alsa-lib support for it as well? David Henningsson (2): Add rawmidi framing API Add test for rawmidi framing API include/rawmidi.h | 26 ++++++++++++ include/sound/uapi/asound.h | 30 +++++++++++++- src/rawmidi/rawmidi.c | 83 +++++++++++++++++++++++++++++++++++++ src/rawmidi/rawmidi_hw.c | 2 + src/rawmidi/rawmidi_local.h | 2 + test/rawmidi.c | 65 ++++++++++++++++++++++++++--- 6 files changed, 200 insertions(+), 8 deletions(-) -- 2.25.1

2 5

arecord is failing with -V stereo
by sujith kumar reddy 17 Aug '21

17 Aug '21

Hi All, arecord is failing vumeter option -V stereo only. localhost ~ # arecord -Dhw:1,2 -r48000 -c2 -fS32_LE /tmp/test_record.wav -M -d 1 -V stereo Recording WAVE '/tmp/test_record.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo *** buffer overflow detected ***: terminated Aborted by signal Aborted... Please provide pointers to debug this option. Thanks Sujith

1 0

[PATCH for-5.15] ALSA: hda: Drop workaround for a hang at shutdown again
by Takashi Iwai 17 Aug '21

17 Aug '21

The commit 0165c4e19f6e ("ALSA: hda: Fix hang during shutdown due to link reset") modified the shutdown callback of the HD-audio controller for working around a hang. Meanwhile, the actual culprit of the hang was identified to be the leftover active codecs that may interfere with the powered down controller somehow, but we took a minimal fix approach for 5.14, and that was the commit above. Now, since the codec drivers go runtime-suspend at shutdown for 5.15, we can revert the change and make sure that the full runtime-suspend is performed at shutdown of HD-audio controller again. This patch essentially reverts the commit above to restore the behavior. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214045 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 47777439961c..3aa432d814a2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,11 +883,10 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) +static void azx_shutdown_chip(struct azx *chip) { azx_stop_chip(chip); - if (!skip_link_reset) - azx_enter_link_reset(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -896,11 +895,6 @@ static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); -static void azx_shutdown_chip(struct azx *chip) -{ - __azx_shutdown_chip(chip, false); -} - static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2363,7 +2357,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - __azx_shutdown_chip(chip, true); + azx_shutdown_chip(chip); } /* PCI IDs */ -- 2.26.2

1 0

[PATCH v6 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by fengzheng923＠gmail.com 17 Aug '21

17 Aug '21

From: Ban Tao <fengzheng923(a)gmail.com> The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. --- v2->v3: None. --- v3->v4: 1.add sig_bits. --- v4->v5: None. --- v5->v6: 1.Modify RXFIFO_CTL_MODE to mode 1. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 403 ++++++++++++++++++++++++++++++++++ 4 files changed, 419 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e924f9e5df97..8d700baaa3ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..bbac836ba4de --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,403 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMPLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMPLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + static struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMPLE_RATE_MASK, + SUN50I_DMIC_SR_SAMPLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.17.1

3 3

[REGRESSION] "ALSA: HDA: Early Forbid of runtime PM" broke my laptop's internal audio
by Michael Catanzaro 17 Aug '21

17 Aug '21

Hi, On my System76 Gazelle Pro (gazp9) laptop, internal audio broke after upgrading to kernel 5.10. The laptop's speakers produce no sound. Audio from headphones still works fine. A quick test is to visit GNOME System Settings -> Sound -> Output, select Test, click Front Left or Front Right and notice there is no sound. I tested a mainline kernel revision from yesterday (5.11-rc5+) and the regression is not yet fixed. I bisected the regression to "ALSA: HDA: Early Forbid of runtime PM": https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… I created a downstream bug report here, including some bits from dmesg that may or may not be relevant in comment #2: https://bugzilla.redhat.com/show_bug.cgi?id=1918933 Audio devices: $ lspci | grep -i audio 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) OS: Fedora 33 I'm happy to provide any additional required info or test patches if you CC me, since I'm not a kernel developer and am not subscribed to these lists. Thanks.

6 16

[PATCH] ALSA: hda/via: Apply runtime PM workaround for ASUS B23E
by Takashi Iwai 17 Aug '21

17 Aug '21

ASUS B23E requires the same workaround like other machines with VT1802, otherwise it looses the codec power on a few nodes and the sound kept silence. Fixes: a0645daf1610 ("ALSA: HDA: Early Forbid of runtime PM") Link: https://lore.kernel.org/r/ac2232f142efcd67fe6ac38897f704f7176bd200.camel@gm… Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a5c1a2c4eae4..773a136161f1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1041,6 +1041,7 @@ static const struct hda_fixup via_fixups[] = { }; static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x13f7, "Asus B23E", VIA_FIXUP_POWER_SAVE), SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", VIA_FIXUP_POWER_SAVE), -- 2.26.2

1 0

[PATCH] ALSA: hda: Fix hang during shutdown due to link reset
by Imre Deak 17 Aug '21

17 Aug '21

During system shutdown codecs may be still active, and resetting the controller->codec HW link in this state - based on the bug reporter's tests - leads to the shutdown sequence to get stuck. This happens at least on the reporter's KBL system with an ALC662 codec. For now fix the issue by skipping the link reset step. Fixes: 472e18f63c42 ("ALSA: hda: Release controller display power during shutdown/reboot") References: https://bugzilla.kernel.org/show_bug.cgi?id=214045 References: https://gitlab.freedesktop.org/drm/intel/-/issues/3618#note_1024665 Reported-and-tested-by: youling257(a)gmail.com Cc: Takashi Iwai <tiwai(a)suse.de> Cc: youling257(a)gmail.com Signed-off-by: Imre Deak <imre.deak(a)intel.com> --- sound/pci/hda/hda_intel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0322b289505e7..0062c18b646af 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void azx_shutdown_chip(struct azx *chip) +static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) { azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!skip_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); +static void azx_shutdown_chip(struct azx *chip) +{ + __azx_shutdown_chip(chip, false); +} + static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2385,7 +2391,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_shutdown_chip(chip); + __azx_shutdown_chip(chip, true); } /* PCI IDs */ -- 2.27.0

2 1

[PATCH AUTOSEL 4.4 1/2] ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs
by Sasha Levin 17 Aug '21

17 Aug '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit acbf58e530416e167c3b323111f4013d9f2b0a7d ] soc_cleanup_component_debugfs will debugfs_remove_recursive the component->debugfs_root, so adsp doesn't need to also remove the same entry. By doing that adsp also creates a race with core component, which causes a NULL pointer dereference Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5ff0d3b10bcf..ceb1daf99526 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -391,7 +391,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 1/2] ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs
by Sasha Levin 17 Aug '21

17 Aug '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit acbf58e530416e167c3b323111f4013d9f2b0a7d ] soc_cleanup_component_debugfs will debugfs_remove_recursive the component->debugfs_root, so adsp doesn't need to also remove the same entry. By doing that adsp also creates a race with core component, which causes a NULL pointer dereference Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 00dd37f10daf..1347131e8b94 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -585,7 +585,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 1/2] ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs
by Sasha Levin 17 Aug '21

17 Aug '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit acbf58e530416e167c3b323111f4013d9f2b0a7d ] soc_cleanup_component_debugfs will debugfs_remove_recursive the component->debugfs_root, so adsp doesn't need to also remove the same entry. By doing that adsp also creates a race with core component, which causes a NULL pointer dereference Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1516252aa0a5..7ed5f4ef7de0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -658,7 +658,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 1/4] ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs
by Sasha Levin 17 Aug '21

17 Aug '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit acbf58e530416e167c3b323111f4013d9f2b0a7d ] soc_cleanup_component_debugfs will debugfs_remove_recursive the component->debugfs_root, so adsp doesn't need to also remove the same entry. By doing that adsp also creates a race with core component, which causes a NULL pointer dereference Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 02c557e1f779..47a4363ebde9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -651,7 +651,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 1/5] ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs
by Sasha Levin 17 Aug '21

17 Aug '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit acbf58e530416e167c3b323111f4013d9f2b0a7d ] soc_cleanup_component_debugfs will debugfs_remove_recursive the component->debugfs_root, so adsp doesn't need to also remove the same entry. By doing that adsp also creates a race with core component, which causes a NULL pointer dereference Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210728104416.636591-1-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 13672928da99..ba776f58d524 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -745,7 +745,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 1/9] ASoC: rt5682: Adjust headset volume button threshold
by Sasha Levin 17 Aug '21

17 Aug '21

From: Derek Fang <derek.fang(a)realtek.com> [ Upstream commit 6d20bf7c020f417fdef1810a22da17c126603472 ] Adjust the threshold of headset button volume+ to fix the wrong button detection issue with some brand headsets. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> Link: https://lore.kernel.org/r/20210721133121.12333-1-derek.fang@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 2e41b8c169e5..0486b1469799 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.30.2

1 2

[PATCH 0/3] ASoC: ti: Use of_device_get_match_data
by Aakash Hemadri 16 Aug '21

16 Aug '21

Hi, This patch series will apply cleanly on for-next Prefer `of_device_get_match_data` over `of_match_device`. Aakash Hemadri (3): ASoC: ti: davinci-mcasp: Use of_device_get_match_data ASoC: ti: omap-mcbsp: Use of_device_get_match_data ASoC: ti: davinci-evm: Use of_device_get_match_data sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) -- 2.32.0

2 7

re: ASoC: sh: Add RZ/G2L SSIF-2 driver
by Colin Ian King 16 Aug '21

16 Aug '21

Hi, Static analysis on linux-next with Coverity has detected an issue in rz_ssi_stop, sound/soc/sh/rz-ssi.c with the following commit: commit 03e786bd43410fa93e5d2459f7a43e90ff0ae801 Author: Biju Das <biju.das.jz(a)bp.renesas.com> Date: Fri Aug 13 10:11:54 2021 +0100 ASoC: sh: Add RZ/G2L SSIF-2 driver The analysis is as follows: 367 368 /* Wait for idle */ 369 timeout = 100; 370 while (--timeout) { Wrong operator used (CONSTANT_EXPRESSION_RESULT) operator_confusion: rz_ssi_reg_readl(ssi, 4) | (33554432UL /* 1UL << 25 */) is always 1/true regardless of the values of its operand. This occurs as the logical operand of if. 371 if (rz_ssi_reg_readl(ssi, SSISR) | SSISR_IIRQ) 372 break; 373 udelay(1); 374 } I suspect the & operator was intended to mask the specific bit. However I'm not sure if the intent is to detect if the SSISR_IIRQ bit should be zero or one at the point where the break should occur, hence I'm reporting this issue. Colin

2 1

[PATCH 0/9] ASoC: soc-xxx: cleanup cppcheck warning
by Kuninori Morimoto 16 Aug '21

16 Aug '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-xxx. Kuninori Morimoto (9): ASoC: soc-generic-dmaengine-pcm: cleanup cppcheck warning at dmaengine_pcm_hw_params() ASoC: soc-generic-dmaengine-pcm: cleanup cppcheck warning at dmaengine_pcm_new() ASoC: soc-generic-dmaengine-pcm: cleanup cppcheck warning at dmaengine_copy_user() ASoC: soc-dai: cleanup cppcheck warning at snd_soc_dai_link_set_capabilities() ASoC: soc-dai: cleanup cppcheck warning at snd_soc_pcm_dai_new() ASoC: soc-jack: cleanup cppcheck warning at snd_soc_jack_report() ASoC: soc-jack: cleanup cppcheck warning for CONFIG_GPIOLIB ASoC: soc-component: cleanup cppcheck warning at snd_soc_pcm_component_pm_runtime_get() ASoC: soc-ac97: cleanup cppcheck warning sound/soc/soc-ac97.c | 14 ++++++-------- sound/soc/soc-component.c | 4 ++-- sound/soc/soc-dai.c | 18 ++++++++---------- sound/soc/soc-generic-dmaengine-pcm.c | 9 +++------ sound/soc/soc-jack.c | 15 +++++++-------- 5 files changed, 26 insertions(+), 34 deletions(-) -- 2.25.1

2 10

[PATCH] ASoC: sh: rz-ssi: Fix dereference of noderef expression warning
by Biju Das 16 Aug '21

16 Aug '21

Fix following sparse warning: sound/soc/sh/rz-ssi.c:156:15: sparse: warning: dereference of noderef expression Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/sh/rz-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 2f9345415a15..7fa3737b90bf 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -148,7 +148,7 @@ static u32 rz_ssi_reg_readl(struct rz_ssi_priv *priv, uint reg) return readl(priv->base + reg); } -static void rz_ssi_reg_mask_setl(struct rz_ssi_priv __iomem *priv, uint reg, +static void rz_ssi_reg_mask_setl(struct rz_ssi_priv *priv, uint reg, u32 bclr, u32 bset) { u32 val; -- 2.17.1

2 1

[PATCH v3 0/8] ASoC: Intel: boards: use software node API
by Pierre-Louis Bossart 16 Aug '21

16 Aug '21

This is an update on an earlier contribution from Heikki Krogerus The function device_add_properties() is going to be removed. Replacing it with software node API equivalents. Thanks for Hans de Goede and Andy Shevchenko for their comments, suggestions and Reviewed-by tags on GitHub. The review thread can be found at https://github.com/thesofproject/linux/pull/3041) v3 changes: Fixed nit-picks from Andy: label, return value, missing commas/periods. Added Andy's Reviewed-by tag v2 changes: feedback from Andy and Hans Better error handling Codec reference is kept until the .remove callback Remove bus search to find device v1 changes from Heikki's patches: Avoid the use of devm_ routines for Baytrail machine drivers. Heikki Krogerus (1): ASoC: Intel: boards: use software node API in Atom boards Pierre-Louis Bossart (7): ASoC: Intel: boards: harden codec property handling ASoC: Intel: boards: handle errors with acpi_dev_get_first_match_dev() ASoC: Intel: boards: get codec device with ACPI instead of bus search ASoC: Intel: sof_sdw: pass card information to init/exit functions ASoC: Intel: sof_sdw_rt711*: keep codec device reference until remove ASoC: Intel: use software node API in SoundWire machines ASoC: Intel: remove device_properties for Atom boards sound/soc/intel/boards/bytcht_es8316.c | 31 ++++++++-- sound/soc/intel/boards/bytcr_rt5640.c | 60 +++++++++++++++----- sound/soc/intel/boards/bytcr_rt5651.c | 63 ++++++++++++++------- sound/soc/intel/boards/sof_sdw.c | 20 ++++--- sound/soc/intel/boards/sof_sdw_common.h | 37 +++++++----- sound/soc/intel/boards/sof_sdw_max98373.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1316.c | 3 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 3 +- sound/soc/intel/boards/sof_sdw_rt700.c | 3 +- sound/soc/intel/boards/sof_sdw_rt711.c | 51 +++++++++-------- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 52 +++++++++-------- sound/soc/intel/boards/sof_sdw_rt715.c | 3 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 3 +- 14 files changed, 223 insertions(+), 112 deletions(-) -- 2.25.1

3 10

[PATCH v2 00/14] ASoC: add Audio Graph Card2 driver
by Kuninori Morimoto 16 Aug '21

16 Aug '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. And some users want to use Audio-Graph-Card as basic connection and expand/customize its own settings on it. 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 update Audio-Graph-Card is that it is very difficult to keep compatibilty. Audio-Graph-Card2 supports Normal Connection, DPCM Connection, Multi CPU/Codec Connection, Codec2Codec Connection, and possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and customized 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. Sample DT settings which can be used for testing is using Test-Component. This patch set includes sample DT for custome driver and/or audio-graph-card2. You can easily try to use/test it if you added below line to your DT file. It needs below CONFIGs. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_SAMPLE_CUSTOM_CARD CONFIG_SND_TEST_COMPONENT There are some notes. 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. 1 - 2 : add Test-Component driver 3 - 9 : add Audio-Graph-Card2 driver 10 : add Audio-Graph-Card2 base custome driver sample 11 - 14 : Audio-Graph-Card2 driver / custome driver DT sample 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 v1 -> v2 - don't use "port" base for_each loop Kuninori Morimoto (14): 1 ASoC: test-component: add Test Component YAML bindings 2 ASoC: test-component: add Test Component for Sound debug/test 3 ASoC: simple-card-utils: add asoc_graph_is_ports0() 4 ASoC: simple-card-utils: add codec2codec support 5 ASoC: audio-graph-card2: add Audio Graph Card2 driver 6 ASoC: audio-graph-card2: add DPCM support 7 ASoC: audio-graph-card2: add Multi CPU/Codec support 8 ASoC: audio-graph-card2: add Codec2Codec support 9 ASoC: audio-graph-card2: add Yaml Document 10 ASoC: sample-custom-card: add Audio Graph Card2 custome sample 11 ASoC: audio-graph-card2-sample.dtsi: add Sample DT for Audio Graph Card2 12 ASoC: audio-graph-card2-sample.dtsi: add DPCM sample 13 ASoC: audio-graph-card2-sample.dtsi: add Multi CPU/Codec sample 14 ASoC: audio-graph-card2-sample.dtsi: add Codec2Codec sample. .../sound/audio-graph-card2-items.yaml | 80 ++ .../bindings/sound/audio-graph-card2.yaml | 51 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 24 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../soc/generic/audio-graph-card2-sample.dtsi | 140 ++ sound/soc/generic/audio-graph-card2.c | 1252 +++++++++++++++++ sound/soc/generic/sample-custom-card.c | 160 +++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 12 files changed, 2474 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2-items.yaml 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-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/sample-custom-card.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

3 30

[PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510 laptop
by Kristin Paget 15 Aug '21

15 Aug '21

The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Precision models, so requires the same quirk to enable woofer output. Tested on my own 9510. Signed-off-by: Kristin Paget <kristin(a)tombom.co.uk> --- 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 218d1aa0fd14..8b72064f8321 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8319,6 +8319,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),

2 1

[PATCH] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 15 Aug '21

15 Aug '21

The rz_ssi_dma_request function only checks the NULL condition for the value returned by the dma_request_chan function, but this function can also return an error. If it happens, the subsequent function call to rz_ssi_dma_slave_config can lead to a kernel crash. This patch fixes the issue by checking both error and NULL condition returned by dma_request_chan. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/sh/rz-ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..3867e2efd3e0 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,19 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR_OR_NULL(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR_OR_NULL(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR_OR_NULL(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.17.1

1 1

[PATCH v2] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 14 Aug '21

14 Aug '21

The pointer returned by the dma_request_chan function is an error pointer for the error cases. But on the rz_ssi_dma_request function, it checks for NULL pointer instead. This patch fixes the issue by checking the error pointer instead of NULL pointer. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- v1->v2: * Replaced IS_ERR_OR_NULL with IS_ERR for error pointer check, as the dma_request_chan function never returns NULL pointer for error case. --- sound/soc/sh/rz-ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..2f9345415a15 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,19 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.17.1

1 0

unable to play audio in a application running as root.
by GitHub issues - opened 14 Aug '21

14 Aug '21

alsa-project/alsa-lib issue #170 was opened from PeaceNIRA: ALSA lib pcm_dmix.c:1075:(snd_pcm_dmix_open) unable to open slave what ive tried: unistall and reinstalling the app adding root to the audio group application: war thunder Issue URL : https://github.com/alsa-project/alsa-lib/issues/170 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4 0/3] Add RZ/G2L Sound support
by Biju Das 14 Aug '21

14 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. v3->v4: * Updated the subject line as per style for the subsystem. * Removed select SND_SIMPLE_CARD from Kconfig * Added C++ comments for copyright and driver description. * Moved validation of channels in hw_params * removed asm issue reported by bot as well as Mark * replaced master/slave macros with provider/consumer macros * Improved locking and added more null pointer checks. v2->v3: * Fixed the dependency on KCONFIG * Merged the binding patch with dma feature added * Updated dt binding example with encoded #dma-cells value. * Improved Error handling in probe function * Removed the passing legacy channel configuration parameters from dmaengine_slave_config function * started using dma_request_chan instead of deprecated dma_request_slave_channel * Removed SoC dtsi and config patches from this series. Will send it later. v1->v2: * Rebased to latest rc kernel Biju Das (3): ASoC: sh: Add RZ/G2L SSIF-2 driver ASoC: dt-bindings: sound: renesas,rz-ssi: Update slave dma channel configuration parameters ASoC: sh: rz-ssi: Add SSI DMAC support .../bindings/sound/renesas,rz-ssi.yaml | 4 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1069 +++++++++++++++++ 4 files changed, 1081 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

4 10

[PATCH] ALSA: pcm: Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag
by Takashi Iwai 14 Aug '21

14 Aug '21

ALSA PCM core has an optimized way to communicate with user-space for its control and status data via mmap on the supported architectures like x86. Depending on the situation, however, we'd rather want to enforce user-space notifying the applptr or hwptr change explicitly via ioctl. For example, the upcoming non-contig and non-coherent buffer handling would need an explicit sync, and this needs to catch the applptr and hwptr changes. Also, ASoC SOF driver will have the SPIB support that has the similar requirement for the explicit control of the applptr and hwptr. This patch adds the new PCM hardware info flag, SNDRV_PCM_INFO_EXPLICIT_SYNC. When this flag is set, PCM core disables both the control and the status mmap, which enforces user-space to update via SYNC_PTR ioctl. In that way, drivers can catch the applptr and hwptr update and apply the sync operation if needed. Link: https://lore.kernel.org/r/20210812113818.6479-1-tiwai@suse.de Link: https://lore.kernel.org/r/20210610205326.1176400-1-pierre-louis.bossart@lin… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 9 +++++++++ 2 files changed, 10 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index d17c061950df..1d84ec9db93b 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -299,6 +299,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ #define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ #define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ +#define SNDRV_PCM_INFO_EXPLICIT_SYNC 0x10000000 /* needs explicit sync of pointers and data */ #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dc9fa312fadd..d233cb3b41d8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3621,6 +3621,12 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file static bool pcm_status_mmap_allowed(struct snd_pcm_file *pcm_file) { + /* If drivers require the explicit sync (typically for non-coherent + * pages), we have to disable the mmap of status and control data + * to enforce the control via SYNC_PTR ioctl. + */ + if (pcm_file->substream->runtime->hw.info & SNDRV_PCM_INFO_EXPLICIT_SYNC) + return false; /* See pcm_control_mmap_allowed() below. * Since older alsa-lib requires both status and control mmaps to be * coupled, we have to disable the status mmap for old alsa-lib, too. @@ -3635,6 +3641,9 @@ static bool pcm_control_mmap_allowed(struct snd_pcm_file *pcm_file) { if (pcm_file->no_compat_mmap) return false; + /* see above */ + if (pcm_file->substream->runtime->hw.info & SNDRV_PCM_INFO_EXPLICIT_SYNC) + return false; /* Disallow the control mmap when SYNC_APPLPTR flag is set; * it enforces the user-space to fall back to snd_pcm_sync_ptr(), * thus it effectively assures the manual update of appl_ptr. -- 2.26.2

1 1

[PATCH v2] usb-audio: Input source control - digidesign mbox
by Damien Zammit 14 Aug '21

14 Aug '21

This adds a second mixer control to Digidesign Mbox to select between Analog/SPDIF capture. Users will note that selecting the SPDIF input source automatically switches the clock mode to sync to SPDIF, which is a feature of the hardware. (You can change the clock source back to internal if you want to capture from spdif but not sync to its clock although this mode is probably not recommended). Unfortunately, starting the stream resets both modes to Internally clocked and Analog input mode. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: Damien Zammit <damien(a)zamaudio.com> --- sound/usb/mixer_quirks.c | 265 ++++++++++++++++++++++++++++++++++++----------- 1 file changed, 203 insertions(+), 62 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7..a66ce0375 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -594,85 +594,208 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) &snd_xonar_u1_output_switch, NULL); } +/* Digidesign Mbox 1 helper functions */ + +static int snd_mbox1_is_spdif_synced(struct snd_usb_audio *chip) +{ + unsigned char buff[3]; + int err; + int is_spdif_synced; + + /* Read clock source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + return err; + + /* spdif sync: buff is all zeroes */ + is_spdif_synced = !(buff[0] | buff[1] | buff[2]); + return is_spdif_synced; +} + +static int snd_mbox1_set_clk_source(struct snd_usb_audio *chip, int rate_or_zero) +{ + /* 2 possibilities: Internal -> expects sample rate + * S/PDIF sync -> expects rate = 0 + */ + unsigned char buff[3]; + + buff[0] = (rate_or_zero >> 0) & 0xff; + buff[1] = (rate_or_zero >> 8) & 0xff; + buff[2] = (rate_or_zero >> 16) & 0xff; + + /* Set clock source */ + return snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); +} + +static int snd_mbox1_is_spdif_input(struct snd_usb_audio *chip) +{ + /* Hardware gives 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + */ + int err; + unsigned char source[1]; + + /* Read input source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + if (err < 0) + return err; + + return (source[0] == 2); +} + +static int snd_mbox1_set_input_source(struct snd_usb_audio *chip, int is_spdif) +{ + /* NB: Setting the input source to S/PDIF resets the clock source to S/PDIF + * Hardware expects 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + */ + unsigned char buff[1]; + + buff[0] = (is_spdif & 1) + 1; + + /* Set input source */ + return snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); +} + /* Digidesign Mbox 1 clock source switch (internal/spdif) */ -static int snd_mbox1_switch_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_clk_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + goto err; + + err = snd_mbox1_is_spdif_synced(chip); + if (err < 0) + goto err; + + kctl->private_value = err; + err = 0; ucontrol->value.enumerated.item[0] = kctl->private_value; - return 0; +err: + snd_usb_unlock_shutdown(chip); + return err; } -static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) +static int snd_mbox1_clk_switch_update(struct usb_mixer_interface *mixer, int is_spdif_sync) { struct snd_usb_audio *chip = mixer->chip; int err; - unsigned char buff[3]; err = snd_usb_lock_shutdown(chip); if (err < 0) return err; - /* Prepare for magic command to toggle clock source */ - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + err = snd_mbox1_is_spdif_synced(chip); if (err < 0) goto err; - /* 2 possibilities: Internal -> send sample rate - * S/PDIF sync -> send zeroes - * NB: Sample rate locked to 48kHz on purpose to - * prevent user from resetting the sample rate - * while S/PDIF sync is enabled and confusing - * this configuration. - */ - if (val == 0) { - buff[0] = 0x80; - buff[1] = 0xbb; - buff[2] = 0x00; - } else { - buff[0] = buff[1] = buff[2] = 0x00; - } + /* FIXME: hardcoded sample rate */ + err = snd_mbox1_set_clk_source(chip, is_spdif_sync ? 0 : 48000); + if (err < 0) + goto err; - /* Send the magic command to toggle the clock source */ - err = snd_usb_ctl_msg(chip->dev, - usb_sndctrlpipe(chip->dev, 0), 0x1, - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + err = snd_mbox1_is_spdif_synced(chip); +err: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_mbox1_clk_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + int err; + bool cur_val, new_val; + + cur_val = kctl->private_value; + new_val = ucontrol->value.enumerated.item[0]; + if (cur_val == new_val) + return 0; + + kctl->private_value = new_val; + err = snd_mbox1_clk_switch_update(mixer, new_val); + return err < 0 ? err : 1; +} + +static int snd_mbox1_clk_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Internal", + "S/PDIF" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int snd_mbox1_clk_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_mbox1_clk_switch_update(list->mixer, list->kctl->private_value); +} + +/* Digidesign Mbox 1 input source switch (analog/spdif) */ + +static int snd_mbox1_src_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kctl->private_value; + return 0; +} + +static int snd_mbox1_src_switch_update(struct usb_mixer_interface *mixer, int is_spdif_input) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + err = snd_mbox1_set_input_source(chip, is_spdif_input); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x2, buff, 3); + + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; + err = snd_mbox1_is_spdif_synced(chip); err: snd_usb_unlock_shutdown(chip); return err; } -static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_src_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); struct usb_mixer_interface *mixer = list->mixer; @@ -685,42 +808,60 @@ static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, return 0; kctl->private_value = new_val; - err = snd_mbox1_switch_update(mixer, new_val); + err = snd_mbox1_src_switch_update(mixer, new_val); return err < 0 ? err : 1; } -static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int snd_mbox1_src_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static const char *const texts[2] = { - "Internal", + "Analog", "S/PDIF" }; return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } -static int snd_mbox1_switch_resume(struct usb_mixer_elem_list *list) +static int snd_mbox1_src_switch_resume(struct usb_mixer_elem_list *list) { - return snd_mbox1_switch_update(list->mixer, list->kctl->private_value); + return snd_mbox1_src_switch_update(list->mixer, list->kctl->private_value); } -static const struct snd_kcontrol_new snd_mbox1_switch = { +static const struct snd_kcontrol_new snd_mbox1_clk_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Source", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_mbox1_switch_info, - .get = snd_mbox1_switch_get, - .put = snd_mbox1_switch_put, + .info = snd_mbox1_clk_switch_info, + .get = snd_mbox1_clk_switch_get, + .put = snd_mbox1_clk_switch_put, + .private_value = 0 +}; + +static const struct snd_kcontrol_new snd_mbox1_src_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_mbox1_src_switch_info, + .get = snd_mbox1_src_switch_get, + .put = snd_mbox1_src_switch_put, .private_value = 0 }; -static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) +static int snd_mbox1_controls_create(struct usb_mixer_interface *mixer) { - return add_single_ctl_with_resume(mixer, 0, - snd_mbox1_switch_resume, - &snd_mbox1_switch, NULL); + int err; + err = add_single_ctl_with_resume(mixer, 0, + snd_mbox1_clk_switch_resume, + &snd_mbox1_clk_switch, NULL); + if (err < 0) + return err; + + return add_single_ctl_with_resume(mixer, 1, + snd_mbox1_src_switch_resume, + &snd_mbox1_src_switch, NULL); } /* Native Instruments device quirks */ @@ -3029,7 +3170,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0dba, 0x1000): /* Digidesign Mbox 1 */ - err = snd_mbox1_create_sync_switch(mixer); + err = snd_mbox1_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 2.13.1

2 1

[PATCH v2 0/4] ASoC: SOF: Intel: DMI L1 power optimization for HDaudio platforms
by Pierre-Louis Bossart 13 Aug '21

13 Aug '21

This patchset provides an optimization that result in significant power savings on Intel HDAudio platforms using SOF (Sound Open Firmware). We previously prevented the Intel DSP from enabling the DMI_L1 capability to work-around issues with pause on capture streams. It turns out that this also prevented the platform from entering high C states in full-duplex usages such as videoconferencing - a rather basic use case since the start of the pandemic. The support for pause_push/release was already a bit controversial for Intel platforms, in theory platforms should only enable PAUSE if they can resume on the same sample, which is not the case on any Intel platform. With this patchset, when the user enables DMI L1 via a kernel parameter, the PAUSE support is disabled for capture streams. A kernel parameter is far from ideal but it's a placeholder until we have an API to negotiate capabilities between applications and driver, and it's far less confusing than a Kconfig option. Changes since v1: Removal of SPIB support since it may conflict with Takashi's memalloc changes. These SPIB changes will be provided after rebase. Addition of one cleanup for cppcheck warning Move all changes to intel/ directory, no changes in shared code Flipped the logic: the selection of DMI L1 disables PAUSE Pierre-Louis Bossart (4): ASoC: SOF: Intel: Kconfig: clarify DMI L1 option description ASoC: SOF: Intel: hda-stream: remove always true condition ASoC: SOF: Intel: simplify logic for DMI_L1 handling ASoC: SOF: Intel: make DMI L1 selection more robust sound/soc/sof/intel/Kconfig | 10 ---------- sound/soc/sof/intel/hda-pcm.c | 16 ++++++++++++++-- sound/soc/sof/intel/hda-stream.c | 11 +++++------ 3 files changed, 19 insertions(+), 18 deletions(-) -- 2.25.1

2 5

[PATCH v5 0/3] Add RZ/G2L Sound support
by Biju Das 13 Aug '21

13 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. v4->v5 * Moved validation of sample bits in hw_params * Removed validation of frame bits as it is redundant * split the rz_ssi_start_stop function into rz_ssi_start and rz_ssi_stop. * remove the spin_lock around rz_ssi_stream_init. * Updated dmas description and removed fixes as it is an enhancement now. * updated ssi_start functions with setting fifo thresholds and ssi_stop function with cancel all dma txn. v3->v4: * Updated the subject line as per style for the subsystem. * Removed select SND_SIMPLE_CARD from Kconfig * Added C++ comments for copyright and driver description. * Moved validation of channels in hw_params * removed asm issue reported by bot as well as Mark * replaced master/slave macros with provider/consumer macros * Improved locking and added more null pointer checks. v2->v3: * Fixed the dependency on KCONFIG * Merged the binding patch with dma feature added * Updated dt binding example with encoded #dma-cells value. * Improved Error handling in probe function * Removed the passing legacy channel configuration parameters from dmaengine_slave_config function * started using dma_request_chan instead of deprecated dma_request_slave_channel * Removed SoC dtsi and config patches from this series. Will send it later. v1->v2: * Rebased to latest rc kernel Biju Das (3): ASoC: sh: Add RZ/G2L SSIF-2 driver ASoC: dt-bindings: renesas,rz-ssi: Update slave dma channel configuration parameters ASoC: sh: rz-ssi: Add SSI DMAC support .../bindings/sound/renesas,rz-ssi.yaml | 22 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1063 +++++++++++++++++ 4 files changed, 1093 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

2 4

[PATCH] ASoC: rt5682: Remove unused variable in rt5682_i2c_remove()
by Douglas Anderson 13 Aug '21

13 Aug '21

In commit 772d44526e20 ("ASoC: rt5682: Properly turn off regulators if wrong device ID") I deleted code but forgot to delete a variable that's now unused. Delete it. Fixes: 772d44526e20 ("ASoC: rt5682: Properly turn off regulators if wrong device ID") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index e559b965a0a6..b9d5d7a0975b 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -294,8 +294,6 @@ static void rt5682_i2c_shutdown(struct i2c_client *client) static int rt5682_i2c_remove(struct i2c_client *client) { - struct rt5682_priv *rt5682 = i2c_get_clientdata(client); - rt5682_i2c_shutdown(client); return 0; -- 2.33.0.rc1.237.g0d66db33f3-goog

3 2

USB-Audio: Device or resource busy (strace log)
by Muni Sekhar 13 Aug '21

13 Aug '21

Hi All, $ cat /proc/asound/cards 0 [USB ]: USB-Audio - Plantronics .Audio 628 USB Plantronics Plantronics .Audio 628 USB at usb-0000:00:14.0-2, full speed I am using a Plantronics USB Audio headset. $ arecord --device hw:0,0 --channels 2 --format S16_LE --rate 44100Hz x.wav arecord: main:722: audio open error: Device or resource busy 'arecord' command always fails the first time after system boot in my system. But subsequent execution of the 'arecord' command runs fine. I've attached the strace log for the "audio open error: Device or resource busy" failure. Is there any fix available for this issue? -- Thanks, Sekhar

3 6

[PATCH v2 0/8] ASoC: Intel: boards: use software node API
by Pierre-Louis Bossart 13 Aug '21

13 Aug '21

This is an update on an earlier contribution from Heikki Krogerus The function device_add_properties() is going to be removed. Replacing it with software node API equivalents. Thanks for Hans de Goede and Andy Shevchenko for their comments, suggestions and Reviewed-by tags on GitHub. The review thread can be found at https://github.com/thesofproject/linux/pull/3041) v2 changes: feedback from Andy and Hans Better error handling Codec reference is kept until the .remove callback Remove bus search to find device v1 changes from Heikki's patches: Avoid the use of devm_ routines for Baytrail machine drivers. Heikki Krogerus (1): ASoC: Intel: boards: use software node API in Atom boards Pierre-Louis Bossart (7): ASoC: Intel: boards: harden codec property handling ASoC: Intel: boards: handle errors with acpi_dev_get_first_match_dev() ASoC: Intel: boards: get codec device with ACPI instead of bus search ASoC: Intel: sof_sdw: pass card information to init/exit functions ASoC: Intel: sof_sdw_rt711*: keep codec device reference until remove ASoC: Intel: use software node API in SoundWire machines ASoC: Intel: remove device_properties for Atom boards sound/soc/intel/boards/bytcht_es8316.c | 31 ++++++++-- sound/soc/intel/boards/bytcr_rt5640.c | 57 ++++++++++++++----- sound/soc/intel/boards/bytcr_rt5651.c | 63 ++++++++++++++------- sound/soc/intel/boards/sof_sdw.c | 20 ++++--- sound/soc/intel/boards/sof_sdw_common.h | 37 +++++++----- sound/soc/intel/boards/sof_sdw_max98373.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 3 +- sound/soc/intel/boards/sof_sdw_rt1316.c | 3 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 3 +- sound/soc/intel/boards/sof_sdw_rt700.c | 3 +- sound/soc/intel/boards/sof_sdw_rt711.c | 51 +++++++++-------- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 52 +++++++++-------- sound/soc/intel/boards/sof_sdw_rt715.c | 3 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 3 +- 14 files changed, 221 insertions(+), 111 deletions(-) -- 2.25.1

2 12

[PATCH] ALSA: memalloc: Count continuous pages in vmalloc buffer handler
by Takashi Iwai 13 Aug '21

13 Aug '21

This is an enhancement for the SG-style page handling in vmalloc buffer handler to calculate the continuous pages. When snd_sgbuf_get_chunk_size() is called for a vmalloc buffer, currently we return only the size that fits into a single page. However, this API call is rather supposed for obtaining the continuous pages and most of vmalloc or noncontig buffers do have lots of continuous pages indeed. So, in this patch, the callback now calculates the possibly continuous pages up to the given size limit. Note that the end address in the function is calculated from the last byte, hence it's one byte shorter. This is because ofs + size can be above the actual buffer size boundary. Until now, this feature isn't really used, but it'll become useful in a later patch that adds the non-contiguous buffer type that shares the same callback function as vmalloc. Link: https://lore.kernel.org/r/20210812113818.6479-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 1cea8cb9668f..c7c943c661e6 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -290,11 +290,13 @@ static int snd_dma_vmalloc_mmap(struct snd_dma_buffer *dmab, return remap_vmalloc_range(area, dmab->area, 0); } +#define get_vmalloc_page_addr(dmab, offset) \ + page_to_phys(vmalloc_to_page((dmab)->area + (offset))) + static dma_addr_t snd_dma_vmalloc_get_addr(struct snd_dma_buffer *dmab, size_t offset) { - return page_to_phys(vmalloc_to_page(dmab->area + offset)) + - offset % PAGE_SIZE; + return get_vmalloc_page_addr(dmab, offset) + offset % PAGE_SIZE; } static struct page *snd_dma_vmalloc_get_page(struct snd_dma_buffer *dmab, @@ -307,11 +309,23 @@ static unsigned int snd_dma_vmalloc_get_chunk_size(struct snd_dma_buffer *dmab, unsigned int ofs, unsigned int size) { - ofs %= PAGE_SIZE; - size += ofs; - if (size > PAGE_SIZE) - size = PAGE_SIZE; - return size - ofs; + unsigned int start, end; + unsigned long addr; + + start = ALIGN_DOWN(ofs, PAGE_SIZE); + end = ofs + size - 1; /* the last byte address */ + /* check page continuity */ + addr = get_vmalloc_page_addr(dmab, start); + for (;;) { + start += PAGE_SIZE; + if (start > end) + break; + addr += PAGE_SIZE; + if (get_vmalloc_page_addr(dmab, start) != addr) + return start - ofs; + } + /* ok, all on continuous pages */ + return size; } static const struct snd_malloc_ops snd_dma_vmalloc_ops = { -- 2.26.2

1 0

[PATCH 0/3] ALSA: hda: Forcibly suspend codec at shutdown
by Takashi Iwai 13 Aug '21

13 Aug '21

Hi, this is a patch set for fixing the issue at shutdown/reboot with HD-audio. The recent regression report brought me to check the shutdown code execution and it turned out that we don't do right for the codec shutdown while powering off the controller side. As a simple but big-hammer solution, this changes the codec driver's shutdown callback to forcibly (runtime-)suspend. It assures the proper power-off and the mute, as done at the usual suspend/resume. Currently my plan is to apply this for 5.15, as the impact is significant and it's a bit too late for 5.14 shortly before the final. Takashi === Takashi Iwai (3): ALSA: hda: conexant: Turn off EAPD at suspend, too ALSA: hda: Suspend codec at shutdown ALSA: hda: Nuke unused reboot_notify callback include/sound/hda_codec.h | 1 - sound/pci/hda/hda_bind.c | 8 +------- sound/pci/hda/hda_codec.c | 14 ++++++++++++++ sound/pci/hda/hda_generic.c | 19 ------------------- sound/pci/hda/hda_generic.h | 1 - sound/pci/hda/hda_local.h | 2 ++ sound/pci/hda/patch_analog.c | 1 - sound/pci/hda/patch_ca0132.c | 6 ------ sound/pci/hda/patch_conexant.c | 15 +++++++++++---- sound/pci/hda/patch_realtek.c | 13 ------------- sound/pci/hda/patch_sigmatel.c | 1 - 11 files changed, 28 insertions(+), 53 deletions(-) -- 2.26.2

1 3

[PATCH 0/2] ALSA: trivial corrections
by Pierre-Louis Bossart 13 Aug '21

13 Aug '21

One typo and one deprecated function to be replaced. Pierre-Louis Bossart (2): ALSA: core: control_led: use strscpy instead of strlcpy ALSA: hda_audio_ext: fix kernel-doc include/sound/hdaudio_ext.h | 2 +- sound/core/control_led.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.25.1

2 3

[PATCH RFC v2 0/4] ALSA: Support for non-coherent and non-contiguous page allocation
by Takashi Iwai 13 Aug '21

13 Aug '21

Hi, this is a revised patchset to add the support for PCM buffers with non-coherent and non-contiguous pages, typically useful for non-x86 architectures. The first patch improves the SG-buffer handling, then add a new PCM info flag that disables the control and status mmap, and implements two new buffer types, SNDRV_DMA_TYPE_NONCONTIG and SNDRV_DMA_TYPE_NONCOHERENT. The former is the SG-buffer and the latter is the continuous page allocation, corresponding to SNDRV_DMA_TYPE_DEV_SG and SNDRV_DMA_TYPE_DEV on x86. Unlike other page types, those are directional (that need the DMA direction at allocation time) and require the explicit sync of buffers around the data transfer (flushing, invalidating). The sync is implemented inside ALSA PCM core and automatically applied at updating the applptr and hwsync via SYNC_PTR ioctl, which should be issued during mmap operation, so it should work transparently as long as applications are running with alsa-lib. For tinyALSA, we might need to revisit the implementation. This is currently an RFC -- more exactly, CALL FOR TESTERS, as I have no Arm machine with the sound device for now. The needed change for an existing driver is simple: * Replace the buffer type from SNDRV_DMA_TYPE_DEV to SNDRV_DMA_TYPE_NONCOHERENT (or from SNDRV_DMA_TYPE_DEV_SG to SNDRV_DMA_TYPE_NONCONTIG), which is set up typically in snd_pcm_set_managed_buffer*() call. * Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag to the PCM runtime hardware.info field It'd be greatly appreciated if anyone can test and try the changes, and help debugging. Again, the code is faily untested. The latest code is found in topic/memalloc-noncontig branch of my sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git The patch set is based on for-next branch, i.e. changes for 5.15 are included. thanks, Takashi === v1: https://lore.kernel.org/r/20210810122200.971-1-tiwai@suse.de v1->v2: * Drop the optimization patch for embedding ops that turned out to be negligible but causing a trouble with the existing sgbuf. * Minor code refactoring === Takashi Iwai (4): ALSA: memalloc: Count continuous pages in vmalloc buffer handler ALSA: pcm: Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag ALSA: memalloc: Support for non-contiguous page allocation ALSA: memalloc: Support for non-coherent page allocation include/sound/memalloc.h | 46 +++++++++- include/uapi/sound/asound.h | 1 + sound/core/memalloc.c | 172 ++++++++++++++++++++++++++++++++---- 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 | 15 ++++ 8 files changed, 236 insertions(+), 22 deletions(-) -- 2.26.2

2 6

ASoC: question about asoc and external modem used with mux
by Dev Null 12 Aug '21

12 Aug '21

Hi, On the motorola droid 4 (xt894 rch/arm/boot/dts/omap4-droid4-xt894.dts) the cpcap asoc codec is used for sound (sound/soc/codecs/cpcap.c). The codec contains a hifi and voice DAI and mixer controlls to connect the various outputs (headphones, internal speaker, earpiece etc) to these dai. The hardware setup looks like this https://lkml.org/lkml/2018/3/27/1225 with the cpcap chip acting as a mux for various other devices on the pcb. The modem is one of these devices. The problem is that the kernel has really no way of knowing if the modem is outputing anything or not and thus if the mixer is setup sutch that the modem is driving some output though dapm route (eg the earpiece), the entire dapm route chain will remain off and inactive causeing the kernel to disable the Earpiece PGA. Thus no output makes it out of the device and the voice call fails unless one messes with the registers to enable Earpiece PGA by hand. Now i know of various ways to hack this problem away and have been looking at the asoc/ alsa documentation for some time now but i really cant seam to figure out how to solve this problem correctly. Really i would need a way to pin the voice dai to be considerd allways active by the kernel so that all downsteam devices get truned on when the mixer is setup sutch that they are conected in a route. Or give userspace a mixer switch or other mechanisum to tell the kernel that the modem is currently actively playing sound. any help that can be given in this problem would me mutch appreciated. -- Dev Null <devnull(a)uvos.xyz>

1 0

pulseaudio: a52 sink doesn't stream audio from bluez source
by GitHub issues - opened 12 Aug '21

12 Aug '21

alsa-project/alsa-plugins issue #32 was opened from quequotion: Sorry to beat a dead horse, and double if this is in fact a pulseaudio issue. I recently configured bluez and pulseaudio to receive streams from devices over bluetooth (eg, to play audio from a phone over the PC's speaker system). This works when the PC's sound card is set to profiles *other than* the A52 plugin (ie, pulseaudio's provided digital stereo output and also the [DTS alsa plugin](https://github.com/darealshinji/dcaenc)). Only when set to use the AC3 output profile, audio from the bluez source is not produced. pavucontrol shows the bluez source is producing sound (the level bounces appropriately), but no audio comes out of the sink (no level bounce in pavucontrol either). Any ideas? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/32 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 1/2] Revert "ASoC: rt5682: Adjust headset volume button threshold"
by derek.fang＠realtek.com 12 Aug '21

12 Aug '21

From: Derek Fang <derek.fang(a)realtek.com> This reverts commit 6d20bf7c020f417fdef1810a22da17c126603472. Since it may risk affecting other headset Vol- button. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7dc01ae6bb66..fcf442b8a732 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,7 +44,6 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, - {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.17.1

2 4

[PATCH v2] ASoC: rt5682: Properly turn off regulators if wrong device ID
by Douglas Anderson 12 Aug '21

12 Aug '21

When I booted up on a board that had a slightly different codec stuffed on it, I got this message at bootup: rt5682 9-001a: Device with ID register 6749 is not rt5682 That's normal/expected, but what wasn't normal was the splat that I got after: WARNING: CPU: 7 PID: 176 at drivers/regulator/core.c:2151 _regulator_put+0x150/0x158 pc : _regulator_put+0x150/0x158 ... Call trace: _regulator_put+0x150/0x158 regulator_bulk_free+0x48/0x70 devm_regulator_bulk_release+0x20/0x2c release_nodes+0x1cc/0x244 devres_release_all+0x44/0x60 really_probe+0x17c/0x378 ... This is because the error paths don't turn off the regulator. Let's fix that. Fixes: 0ddce71c21f0 ("ASoC: rt5682: add rt5682 codec driver") Fixes: 87b42abae99d ("ASoC: rt5682: Implement remove callback") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> Reviewed-by: Stephen Boyd <swboyd(a)chromium.org> --- Changes in v2: - Add a blank line. sound/soc/codecs/rt5682-i2c.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 4a56a52adab5..e559b965a0a6 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -117,6 +117,13 @@ static struct snd_soc_dai_driver rt5682_dai[] = { }, }; +static void rt5682_i2c_disable_regulators(void *data) +{ + struct rt5682_priv *rt5682 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); +} + static int rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -157,6 +164,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators, + rt5682); + if (ret) + return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); if (ret) { @@ -285,7 +297,6 @@ static int rt5682_i2c_remove(struct i2c_client *client) struct rt5682_priv *rt5682 = i2c_get_clientdata(client); rt5682_i2c_shutdown(client); - regulator_bulk_disable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); return 0; } -- 2.32.0.605.g8dce9f2422-goog

2 1

[PATCH] ASoC: rt5682: Properly turn off regulators if wrong device ID
by Douglas Anderson 12 Aug '21

12 Aug '21

When I booted up on a board that had a slightly different codec stuffed on it, I got this message at bootup: rt5682 9-001a: Device with ID register 6749 is not rt5682 That's normal/expected, but what wasn't normal was the splat that I got after: WARNING: CPU: 7 PID: 176 at drivers/regulator/core.c:2151 _regulator_put+0x150/0x158 pc : _regulator_put+0x150/0x158 ... Call trace: _regulator_put+0x150/0x158 regulator_bulk_free+0x48/0x70 devm_regulator_bulk_release+0x20/0x2c release_nodes+0x1cc/0x244 devres_release_all+0x44/0x60 really_probe+0x17c/0x378 ... This is because the error paths don't turn off the regulator. Let's fix that. Fixes: 0ddce71c21f0 ("ASoC: rt5682: add rt5682 codec driver") Fixes: 87b42abae99d ("ASoC: rt5682: Implement remove callback") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 4a56a52adab5..1cc07812b5ac 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -117,6 +117,13 @@ static struct snd_soc_dai_driver rt5682_dai[] = { }, }; +static void rt5682_i2c_disable_regulators(void *data) +{ + struct rt5682_priv *rt5682 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); +} + static int rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -156,6 +163,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators, + rt5682); + if (ret) + return ret; ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); @@ -285,7 +296,6 @@ static int rt5682_i2c_remove(struct i2c_client *client) struct rt5682_priv *rt5682 = i2c_get_clientdata(client); rt5682_i2c_shutdown(client); - regulator_bulk_disable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); return 0; } -- 2.32.0.605.g8dce9f2422-goog

4 5