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[PATCH 0/6] OPE support on Tegra210 and later
by Sameer Pujar 21 May '22

21 May '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 51 + .../bindings/sound/nvidia,tegra210-ope.yaml | 91 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 52 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2920 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

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[PATCH] ALSA: usb-audio: Workaround for clock setup on TEAC devices
by Takashi Iwai 21 May '22

21 May '22

Maris reported that TEAC UD-501 (0644:8043) doesn't work with the typical "clock source 41 is not valid, cannot use" errors on the recent kernels. The currently known workaround so far is to restore (partially) what we've done unconditionally at the clock setup; namely, re-setup the USB interface immediately after the clock is changed. This patch re-introduces the behavior conditionally for TEAC devices. Further notes: - The USB interface shall be set later in snd_usb_endpoint_configure(), but this seems to be too late. - Even calling usb_set_interface() right after sne_usb_init_sample_rate() doesn't help; so this must be related with the clock validation, too. - The device may still spew the "clock source 41 is not valid" error at the first clock setup. This seems happening at the very first try of clock setup, but it disappears at later attempts. The error is likely harmless because the driver retries the clock setup (such an error is more or less expected on some devices). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-and-tested-by: Maris Abele <maris7abele(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 4dfe76416794..3c435d379306 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -572,6 +572,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, /* continue processing */ } + /* FIXME - TEAC devices require the immediate interface setup */ + if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) { + usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); + if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) + msleep(50); + } + validation: /* validate clock after rate change */ if (!uac_clock_source_is_valid(chip, fmt, clock)) -- 2.35.3

1 0

Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [x] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [x] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [x] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [x] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [ ] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [ ] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt1308-sdw: add the default value of register 0xc320
by shumingf＠realtek.com 20 May '22

20 May '22

From: Shuming Fan <shumingf(a)realtek.com> The driver missed the default value of register 0xc320. This patch adds that default value to avoid the error messages when the driver went to suspend mode already. BugLink: https://github.com/thesofproject/linux/issues/3651 Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308-sdw.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index c5ce75666dcc..6668e19d85d4 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -140,6 +140,7 @@ static const struct reg_default rt1308_reg_defaults[] = { { 0x3008, 0x02 }, { 0x300a, 0x00 }, { 0xc003 | (RT1308_DAC_SET << 4), 0x00 }, + { 0xc000 | (RT1308_POWER << 4), 0x00 }, { 0xc001 | (RT1308_POWER << 4), 0x00 }, { 0xc002 | (RT1308_POWER << 4), 0x00 }, }; -- 2.35.3

3 2

[PATCH] ASoC: max98090: Remove unneeded check in max98090_put_enab_tlv()
by Alexey Khoroshilov 20 May '22

20 May '22

Variable sel is of unsigned int type, so sel < 0 is not required. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Fixes: 2fbe467bcbfc ("ASoC: max98090: Reject invalid values in custom control put()") --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 62b41ca050a2..c535a8496bf1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,7 +413,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; - if (sel < 0 || sel > mc->max) + if (sel > mc->max) return -EINVAL; *select = sel; -- 2.7.4

3 10

[PATCH 0/8] ASoC: SOF: Introduce generic (in)firmware tracing infrastructure
by Peter Ujfalusi 20 May '22

20 May '22

Hi, SOF is using dma-trace (or dtrace) as a firmware tracing method, which is only supported with IPC3 and it is not applicable for IPC4. Currently the dtrace is 'open managed' regardless of IPC version (we do force disable it for IPC4, but the dtrace calls remain in place). >From the kernel point of view there are only few ops that needs to be exposed by the firmware tracing support and everything else is IPC private, should not be known by the core. This series converts the current dma-trace as ipc3 specific firmware tracing sub-component and moves all private data out from generic code. Regards, Peter --- Peter Ujfalusi (8): ASoC: SOF: Introduce IPC independent ops for firmware tracing support ASoC: SOF: Rename dtrace_is_supported flag to fw_trace_is_supported ASoC: SOF: Clone the trace code to ipc3-dtrace as fw_tracing implementation ASoC: SOF: Switch to IPC generic firmware tracing ASoC: SOF: ipc3-dtrace: Move host ops wrappers from generic header to private ASoC: SOF: Modify the host trace_init parameter list to include dmab ASoC: SOF: Introduce opaque storage of private data for firmware tracing ASoC: SOF: ipc3-dtrace: Move dtrace related variables local from sof_dev sound/soc/sof/Makefile | 1 + sound/soc/sof/amd/acp-trace.c | 4 +- sound/soc/sof/amd/acp.h | 2 +- sound/soc/sof/core.c | 13 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-trace.c | 4 +- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/ipc.c | 6 + sound/soc/sof/ipc3-dtrace.c | 649 ++++++++++++++++++++++++++++++++ sound/soc/sof/ipc3-priv.h | 38 ++ sound/soc/sof/ipc3.c | 3 +- sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 26 -- sound/soc/sof/pm.c | 8 +- sound/soc/sof/sof-priv.h | 53 +-- sound/soc/sof/trace.c | 621 ++---------------------------- 17 files changed, 767 insertions(+), 669 deletions(-) create mode 100644 sound/soc/sof/ipc3-dtrace.c -- 2.36.1

2 9

[PATCH 0/2] ASoC: rt9120: Fix 3byte read logic and optimize 'PWDN' control
by cy_huang 20 May '22

20 May '22

From: ChiYuan Huang <cy_huang(a)richtek.com> Hi, This patch series try to fix two things. - 3byte read logic bug - Use regcache and pm_runtime to optimize 'PWDN' control ChiYuan Huang (2): ASoC: rt9120: Fix 3byte read, valule offset typo ASoC: rt9120: Use pm_runtime and regcache to optimize 'pwdnn' logic sound/soc/codecs/rt9120.c | 113 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 106 insertions(+), 7 deletions(-) -- 2.7.4

2 3

[PATCH] ASoC: SOF: mediatek: remove duplicate include in mt8195.c
by Wan Jiabing 20 May '22

20 May '22

Fix following checkincludes.pl warning: sound/soc/sof/mediatek/mt8195/mt8195.c: linux/of_platform.h is included more than once. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index f4b24afb6f75..7278132300e0 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -12,7 +12,6 @@ #include <linux/delay.h> #include <linux/firmware.h> #include <linux/io.h> -#include <linux/of_platform.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> -- 2.36.1

2 1

[PATCH -next] ASoC: wm2000: fix missing clk_disable_unprepare() on error in wm2000_anc_transition()
by Yang Yingliang 20 May '22

20 May '22

Fix the missing clk_disable_unprepare() before return from wm2000_anc_transition() in the error handling case. Fixes: 514cfd6dd725 ("ASoC: wm2000: Integrate with clock API") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/wm2000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 72e165cc6443..97ece3114b3d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -536,7 +536,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, { struct i2c_client *i2c = wm2000->i2c; int i, j; - int ret; + int ret = 0; if (wm2000->anc_mode == mode) return 0; @@ -566,13 +566,13 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, ret = anc_transitions[i].step[j](i2c, anc_transitions[i].analogue); if (ret != 0) - return ret; + break; } if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return 0; + return ret; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) -- 2.25.1

3 2

[PATCH 0/2] ASoC: SOF: mediatek: add debug dump
by Pierre-Louis Bossart 20 May '22

20 May '22

YC Hung (2): ASoC: SOF: mediatek: Add mediatek common debug dump ASoC: SOF: mediatek: Add mt8195 debug dump sound/soc/sof/mediatek/Makefile | 1 + sound/soc/sof/mediatek/mt8195/mt8195.c | 29 ++++++++ sound/soc/sof/mediatek/mtk-adsp-common.c | 84 ++++++++++++++++++++++++ sound/soc/sof/mediatek/mtk-adsp-common.h | 10 +++ 4 files changed, 124 insertions(+) create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.c create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.h -- 2.30.2

2 3

[PATCH RESEND] ASoC: SOF: sof-client-ipc-flood-test: use pm_runtime_resume_and_get()
by Pierre-Louis Bossart 20 May '22

20 May '22

Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. No functional changes. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- This patch was sent as part of the "[PATCH 0/4] ASoC: SOF: use pm_runtime_resume_and_get()" series, but somehow not applied. The other 3 patches were selected. sound/soc/sof/sof-client-ipc-flood-test.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c index db3a052c5dd2b..4bdecd80248ac 100644 --- a/sound/soc/sof/sof-client-ipc-flood-test.c +++ b/sound/soc/sof/sof-client-ipc-flood-test.c @@ -217,10 +217,9 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf ipc_count = MAX_IPC_FLOOD_COUNT; } - ret = pm_runtime_get_sync(dev); + ret = pm_runtime_resume_and_get(dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(dev, "debugfs write failed to resume %d\n", ret); - pm_runtime_put_noidle(dev); goto out; } -- 2.30.2

2 1

[PATCH 0/2] ASoC: remove two unnecessary gpiolib dependencies
by Pierre-Louis Bossart 20 May '22

20 May '22

Remove two dependencies - issues reported by Intel kernel test bot. Pierre-Louis Bossart (2): ASoC: max98357a: remove dependency on GPIOLIB ASoC: rt1015p: remove dependency on GPIOLIB sound/soc/codecs/Kconfig | 2 -- 1 file changed, 2 deletions(-) -- 2.30.2

2 3

[PATCH v2 1/2] Revert "ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver"
by V sujith kumar Reddy 20 May '22

20 May '22

ASoC: amd : acp : Set Speaker enable/disable pin through rt1019 codec driver. RT1019 codec has two ways of controlling the en_spkr. one way is controlling through gpio pin method the another way is through codec register update through driver. Now Speaker enable/disable is controlled through codec register updated by codec driver. This patch reverts gpio logic. This reverts commit 7fa5c33d043160eba3be9fb8e21588dff2a467c7. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c855f50d6b34..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index bf61a1726f0e..8243765d490f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

2 2

[PATCH -next] ASoC: codecs: lpass: Fix passing zero to 'PTR_ERR'
by YueHaibing 20 May '22

20 May '22

sound/soc/codecs/lpass-macro-common.c:28 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:38 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:54 lpass_macro_pds_init() warn: passing zero to 'ERR_PTR' dev_pm_domain_attach_by_name() may return NULL, set 'ret' as -ENODATA to fix this warning. Fixes: 1a8ee4cf8418 ("ASoC: codecs: Fix error handling in power domain init and exit handlers") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/lpass-macro-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 3c661fd61173..1b9082d237c1 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -25,7 +25,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); if (IS_ERR_OR_NULL(l_pds->macro_pd)) { - ret = PTR_ERR(l_pds->macro_pd); + ret = l_pds->macro_pd ? PTR_ERR(l_pds->macro_pd) : -ENODATA; goto macro_err; } @@ -35,7 +35,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) { - ret = PTR_ERR(l_pds->dcodec_pd); + ret = l_pds->dcodec_pd ? PTR_ERR(l_pds->dcodec_pd) : -ENODATA; goto dcodec_err; } -- 2.17.1

2 1

topology: plugins: nhlt: fix intel dmic mode calculation
by GitHub pull_request - opened 20 May '22

20 May '22

alsa-project/alsa-utils pull request #148 was opened from juimonen: Dmic find_modes function was used with same sampling rate for both fifo a and b parameter calculations. This bug was found in testing two simultaneous dmic dais in topology. Fix this by introducing a sampling rate argument for find_modes. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/148 Patch URL : https://github.com/alsa-project/alsa-utils/pull/148.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

6 14

[PATCH v1 0/2] ASOC : amd: acp :Add Machine driver changes for
by V sujith kumar Reddy 19 May '22

19 May '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 18 +++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ 5 files changed, 263 insertions(+), 17 deletions(-) -- 2.25.1

1 2

[v7] FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 18 May '22

18 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 86 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 89 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..56bae2185904b 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,50 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +576,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +605,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +630,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +643,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +688,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; 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 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Fix wrong kfree issue in snd_usb_endpoint_free_all
by Wan Jiabing 18 May '22

18 May '22

Fix following coccicheck error: ./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671. Here should be 'cp' rather than 'ip'. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index df5a70013a85..f9c921683948 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1668,7 +1668,7 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) kfree(ip); list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) - kfree(ip); + kfree(cp); } /* -- 2.36.1

2 1

[PATCH v2 00/15] ASoC: Intel: avs: Driver core and PCM operations
by Cezary Rojewski 17 May '22

17 May '22

Part three of main AVS driver series. This series was originally part of the initial series which was later divided [1] into smaller, easier-to-review chunks. Thus, many patches found here were already present on the list. This series consists of code typical to many drivers - PCI driver operations, trace ability, PM operations - as well as PCM handlers for all standard audio interfaces, that is, HDA, I2S and DMIC are found here. Series starts with updating firmware boot flow - libraries are no longer ignored. This change is dependent on already merged topology code [2] and because of that could not be part of the initial series [1]. PCM operations are split into four changes. First component operations alone i.e. operations which are usually agnostic towards path position (FE/BE). Then it continues with "generic" FE operations - there is no interface split here as from Intel ADSP point of view, FE, or HOST side as it's called in the specs, involves HD-Audio operations only. BE (also known as LINK) side on the other hand is divided into "non-HD-Audio" and HD-Audio part. The former represents transfer over DMIC and I2S interfaces both. While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: - event tracing - preparation for firmware tracing (debugability) - coredump (debugability) - recovery flow (attempt recovery after IPC timeout or exception) - D0ix (D0 device substate, complements standard power management) Series is finalized by actual addition of supported platforms: SKL and APL-based. Platform-specific files are limited to firmware-specific bits, that is, bits that are specific to given firmware generation. Everything else is shared and is part of already upstream messaging code found in ipc.c, messages.c and messages.h files. Changes in v2: - usage of avs_releast_last_firmware() dropped in error path for library loading procedure as suggested by Pierre - 'link_mask' usage replaced with 'i2s_link_mask' as requested by Pierre. Existing code addressed with new patch: "ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask" - fixed possible race during recovery flow (->recovering flag is now atomic and tested in single location only). - dropped platform prefixes for basically all i2s board descriptors - 'ssp_test' renamed to 'i2s_test' to match naming convention of other boards - simplified PM implementation for current series, 'low_power' bits moved to future series - replaced SND_INTEL_DSP_DRIVER_SST with _AVS as suggested by Mark. Required changes to intel-dspcfg will be added in future series - number of typos across commit messages addressed [1]: https://lore.kernel.org/all/20220311153544.136854-1-cezary.rojewski@intel.c… [2]: https://lore.kernel.org/all/20220331135246.993089-1-cezary.rojewski@intel.c… Cezary Rojewski (14): ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support Piotr Maziarz (1): ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask include/sound/intel-dsp-config.h | 3 +- include/sound/soc-acpi.h | 2 + sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 7 +- sound/soc/intel/avs/apl.c | 250 ++++++ sound/soc/intel/avs/avs.h | 79 ++ sound/soc/intel/avs/board_selection.c | 501 +++++++++++ sound/soc/intel/avs/core.c | 631 +++++++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 253 +++++- sound/soc/intel/avs/loader.c | 84 ++ sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 ++ sound/soc/intel/avs/pcm.c | 1182 +++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 +++ sound/soc/intel/avs/topology.c | 14 +- sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 158 ++++ sound/soc/intel/avs/utils.c | 23 + 20 files changed, 3447 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h -- 2.25.1

5 21

Re: Re: [PATCH] conf: vc4-hdmi: use a proper hdmi pcm, fix broken default pcm
by popcorn mix 17 May '22

17 May '22

Matthias Reichl wrote: > Userspace expects to see a HDMI pcm, so remove the somewhat broken > front and iec958 pcms and add a proper hdmi pcm instead. > > The hdmi pcm supports HD/HBR audio passthrough and relays the IEC958 > status bits on to the kernel driver so it can switch to HBR audio > packets if needed. The control hook is marked optional because kernels > before 5.14 didn't expose the IEC958 controls. > > The default pcm never worked as dmix doesn't support the iec958 format. > So drop dmix, only use plug and softvol, and use the hdmi pcm for > iec958 formatting. I can confirm the upstream cards/vc4-hdmi.conf is not usable, and RPiOS ships a patched version (which was an earlier version of this patch). I've tested this version with RPiOS bullseye and all seems good. I could play stereo and multi-channel PCM from VLC. I could also play DTS, DTS-HD and TrueHD formats as passthrough using kodi. It would be good if we can switch from a custom version of vc4-hdmi.conf to an upstream version in RPiOS when this is merged. Tested-by: Dom Cobley <popcornmix(a)gmail.com>

1 0

[PATCH] ASoC: ux500: mop500: Check before clk_put() not needed
by Yihao Han 17 May '22

17 May '22

clk_put() already checks the clk ptr using !clk and IS_ERR() so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/ux500/mop500_ab8500.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 3e654e708f78..1ea1729984a9 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -433,12 +433,9 @@ void mop500_ab8500_remove(struct snd_soc_card *card) { struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); - if (drvdata->clk_ptr_sysclk != NULL) - clk_put(drvdata->clk_ptr_sysclk); - if (drvdata->clk_ptr_ulpclk != NULL) - clk_put(drvdata->clk_ptr_ulpclk); - if (drvdata->clk_ptr_intclk != NULL) - clk_put(drvdata->clk_ptr_intclk); + clk_put(drvdata->clk_ptr_sysclk); + clk_put(drvdata->clk_ptr_ulpclk); + clk_put(drvdata->clk_ptr_intclk); snd_soc_card_set_drvdata(card, drvdata); } -- 2.17.1

2 1

[PATCH] ASoC: SOF: ipc-msg-injector: Cap the rmaining to count in IPC4 mode
by Peter Ujfalusi 17 May '22

17 May '22

If user space provides smaller buffer than the IPC4 reply then it is possible that we corrupt user space memory since the IPC4 dfs_read function is not using the count directly in copy_to_user() due to the nature of an IPC4 message. Cap the remaining counter to make sure that we are not writing too much to the user space provided buffer. Add a check also to make sure that the buffer is at least the size of the IPC4 header. Fixes: 066c67624d8c: "ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages" Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-client-ipc-msg-injector.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index c2480317730c..03490a4d4ae7 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -76,12 +76,17 @@ static ssize_t sof_msg_inject_ipc4_dfs_read(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->rx_buffer; + size_t header_size = sizeof(ipc4_msg->header_u64); size_t remaining; if (!ipc4_msg->header_u64 || !count || *ppos) return 0; - remaining = sizeof(ipc4_msg->header_u64); + /* we need space for the header at minimum (u64) */ + if (count < header_size) + return -ENOSPC; + + remaining = header_size; /* Only get large config have payload */ if (SOF_IPC4_MSG_IS_MODULE_MSG(ipc4_msg->primary) && @@ -90,13 +95,15 @@ static ssize_t sof_msg_inject_ipc4_dfs_read(struct file *file, if (count > remaining) count = remaining; + else if (count < remaining) + remaining = count; /* copy the header first */ - if (copy_to_user(buffer, &ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64))) + if (copy_to_user(buffer, &ipc4_msg->header_u64, header_size)) return -EFAULT; - *ppos += sizeof(ipc4_msg->header_u64); - remaining -= sizeof(ipc4_msg->header_u64); + *ppos += header_size; + remaining -= header_size; if (!remaining) return count; -- 2.36.1

2 1

[PATCH -next] ASoC: amd: vangogh: Remove duplicate include files
by YueHaibing 17 May '22

17 May '22

Remove duplicated includes. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/vangogh/acp5x-mach.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index d8b25622f911..727de46860b1 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -17,10 +17,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> -#include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> -#include <linux/io.h> #include <linux/acpi.h> #include <linux/dmi.h> -- 2.17.1

2 1

[PATCH 0/3] ASoC: fsl_sai: Add support for i.MX8MM, MP, ULP
by Shengjiu Wang 17 May '22

17 May '22

ASoC: fsl_sai: Add support for i.MX8MM, MP, ULP platforms Shengjiu Wang (3): ASoC: fsl_sai: Add support for i.MX8MM ASoC: fsl_sai: Add support for i.MX8M Plus ASoC: fsl_sai: Add support for i.MX8ULP sound/soc/fsl/fsl_sai.c | 43 ++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 43 insertions(+), 1 deletion(-) -- 2.17.1

3 5

[PATCH v0] ASoC: rt5645: Fix errorenous cleanup order
by Lin Ma 17 May '22

17 May '22

There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 197c56047947..71a8990ba548 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

2 1

[PATCH 1/3] ASoC: core: Correct spelling fliped -> flipped
by Charles Keepax 17 May '22

17 May '22

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc.h | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/soc-core.c | 6 +++--- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f906e5a708308..f20f5f890794a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1259,7 +1259,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname); -unsigned int snd_soc_daifmt_clock_provider_fliped(unsigned int dai_fmt); +unsigned int snd_soc_daifmt_clock_provider_flipped(unsigned int dai_fmt); unsigned int snd_soc_daifmt_clock_provider_from_bitmap(unsigned int bit_frame); unsigned int snd_soc_daifmt_parse_format(struct device_node *np, const char *prefix); diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 2b5d20f02f8f8..77ac4051b8276 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -711,7 +711,7 @@ static void graph_link_init(struct asoc_simple_priv *priv, */ daiclk = snd_soc_daifmt_clock_provider_from_bitmap(bit_frame); if (is_cpu_node) - daiclk = snd_soc_daifmt_clock_provider_fliped(daiclk); + daiclk = snd_soc_daifmt_clock_provider_flipped(daiclk); dai_link->dai_fmt = daifmt | daiclk; dai_link->init = asoc_simple_dai_init; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d68e64b73eea7..32267a38130b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,7 +1230,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, /* * Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ - inv_dai_fmt = snd_soc_daifmt_clock_provider_fliped(dai_fmt); + inv_dai_fmt = snd_soc_daifmt_clock_provider_flipped(dai_fmt); for_each_rtd_cpu_dais(rtd, i, cpu_dai) { unsigned int fmt = dai_fmt; @@ -3035,7 +3035,7 @@ int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname) } EXPORT_SYMBOL_GPL(snd_soc_of_parse_aux_devs); -unsigned int snd_soc_daifmt_clock_provider_fliped(unsigned int dai_fmt) +unsigned int snd_soc_daifmt_clock_provider_flipped(unsigned int dai_fmt) { unsigned int inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; @@ -3056,7 +3056,7 @@ unsigned int snd_soc_daifmt_clock_provider_fliped(unsigned int dai_fmt) return inv_dai_fmt; } -EXPORT_SYMBOL_GPL(snd_soc_daifmt_clock_provider_fliped); +EXPORT_SYMBOL_GPL(snd_soc_daifmt_clock_provider_flipped); unsigned int snd_soc_daifmt_clock_provider_from_bitmap(unsigned int bit_frame) { -- 2.30.2

3 4

[PATCH] ASoC: ti: j721e-evm: Fix refcount leak in j721e_soc_probe_*
by Miaoqian Lin 17 May '22

17 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not needed anymore. Add missing of_node_put() to avoid refcount leak. Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/ti/j721e-evm.c | 44 ++++++++++++++++++++++++++++++---------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 4077e15ec48b..6a969874c927 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -630,17 +630,18 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, codec_node = of_parse_phandle(node, "ti,cpb-codec", 0); if (!codec_node) { dev_err(priv->dev, "CPB codec node is not provided\n"); - return -EINVAL; + ret = -EINVAL; + goto put_dai_node; } domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB]; ret = j721e_get_clocks(priv->dev, &domain->codec, "cpb-codec-scki"); if (ret) - return ret; + goto put_codec_node; ret = j721e_get_clocks(priv->dev, &domain->mcasp, "cpb-mcasp-auxclk"); if (ret) - return ret; + goto put_codec_node; /* * Common Processor Board, two links @@ -650,8 +651,10 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, comp_count = 6; compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), GFP_KERNEL); - if (!compnent) - return -ENOMEM; + if (!compnent) { + ret = -ENOMEM; + goto put_codec_node; + } comp_idx = 0; priv->dai_links[*link_idx].cpus = &compnent[comp_idx++]; @@ -702,6 +705,12 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, (*conf_idx)++; return 0; + +put_codec_node: + of_node_put(codec_node); +put_dai_node: + of_node_put(dai_node); + return ret; } static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, @@ -726,23 +735,25 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, codeca_node = of_parse_phandle(node, "ti,ivi-codec-a", 0); if (!codeca_node) { dev_err(priv->dev, "IVI codec-a node is not provided\n"); - return -EINVAL; + ret = -EINVAL; + goto put_dai_node; } codecb_node = of_parse_phandle(node, "ti,ivi-codec-b", 0); if (!codecb_node) { dev_warn(priv->dev, "IVI codec-b node is not provided\n"); - return 0; + ret = 0; + goto put_codeca_node; } domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_IVI]; ret = j721e_get_clocks(priv->dev, &domain->codec, "ivi-codec-scki"); if (ret) - return ret; + goto put_codecb_node; ret = j721e_get_clocks(priv->dev, &domain->mcasp, "ivi-mcasp-auxclk"); if (ret) - return ret; + goto put_codecb_node; /* * IVI extension, two links @@ -754,8 +765,10 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, comp_count = 8; compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), GFP_KERNEL); - if (!compnent) - return -ENOMEM; + if (!compnent) { + ret = -ENOMEM; + goto put_codecb_node; + } comp_idx = 0; priv->dai_links[*link_idx].cpus = &compnent[comp_idx++]; @@ -816,6 +829,15 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, (*conf_idx)++; return 0; + + +put_codecb_node: + of_node_put(codecb_node); +put_codeca_node: + of_node_put(codeca_node); +put_dai_node: + of_node_put(dai_node); + return ret; } static int j721e_soc_probe(struct platform_device *pdev) -- 2.25.1

2 1

[PATCH v9 0/3] firmware: mtk: add adsp ipc protocol for SOF
by Tinghan Shen 17 May '22

17 May '22

This patch provides mediatek adsp ipc support for SOF. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This patch was tested and confirmed to work with SOF fw on MT8195 cherry board and MT8186 krabby board. changes since v8: - fix patchset 2 and 3. move "depends on MTK_ADSP_IPC" from SND_SOC_SOF_MTK_COMMON to SND_SOC_SOF_MT8195/MT8186 to prevent generating wrong config. changes since v7: - rebase to linux-next/next-22020504 - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL in mtk-adsp-ipc.c - move mtk-adsp-ipc.c out from driver/firmware/mediatek - add user of mtk-adsp-ipc.h in patchset 2 and 3. changes since v6: - rebase to matthias.bgg/linux.git, v5.18-next/soc - Prefer "GPL" over "GPL v2" for MODULE_LICENSE changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): ASoC: SOF: mediatek: Add ipc support for mt8195 TingHan Shen (1): firmware: mediatek: add adsp ipc protocol interface Tinghan Shen (1): ASoC: SOF: mediatek: Add mt8186 ipc support drivers/firmware/Kconfig | 9 + drivers/firmware/Makefile | 1 + drivers/firmware/mtk-adsp-ipc.c | 157 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 ++++++++ sound/soc/sof/mediatek/Kconfig | 2 + sound/soc/sof/mediatek/adsp_helper.h | 12 +- sound/soc/sof/mediatek/mt8186/mt8186-loader.c | 5 + sound/soc/sof/mediatek/mt8186/mt8186.c | 141 ++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 138 ++++++++++++++- 9 files changed, 519 insertions(+), 11 deletions(-) create mode 100644 drivers/firmware/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 4

vc4-hdmi.conf: use a proper hdmi pcm, fix broken default pcm
by GitHub issues - opened 17 May '22

17 May '22

alsa-project/alsa-lib issue #229 was opened from macmpi: referring https://github.com/raspberrypi/linux/issues/4662#issuecomment-1114000673 and assorted [patch](https://mailman.alsa-project.org/pipermail/alsa-devel/2022-April/200… from @HiassofT Thanks for consideration. Issue URL : https://github.com/alsa-project/alsa-lib/issues/229 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2 v3] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 17 May '22

17 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- Changes since v2: Fixes commit description length sound/soc/codecs/adau7118.c | 62 ++++++++++++++++++++++++++++++++++--- 1 file changed, 58 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

2 4

UCM2 configuration guide
by Peter Wedder 17 May '22

17 May '22

Hello, I have been working on an article to describe the process of writing UCM2 configuration to split multi-channel devices into smaller devices, particularly in the context of USB audio interfaces. This is a common problem with enthusiast/studio USB audio interfaces and it happens to be a pet peeve of mine. I recently contributed a profile for the Behringer UMC204HD (https://github.com/alsa-project/alsa-ucm-conf/pull/128) and found the process to be a little intimidating, but with a lot of help from Perexg I managed to get it working. I would like to help other potential contributors out by explaining the process step by step. I was considering posting the article to the Fedora Magazine (https://fedoramagazine.org/), since that's the distro I use. However, I wanted to check with you, the ALSA maintainers, first, because the intent of the article is to encourage readers to contribute UCM configurations for their own cards, and I thought perhaps a potential surge of contributions is not something you want at the moment. Let me know what you think -- is this a welcome idea or should I forget it? I have a draft of the article available here: https://gist.github.com/brndd/ccec98a575f7c0422d50402937439227 Thanks, Peter Wedder

2 3

[PATCH] ALSA: seq: replace ternary operator with max()
by Guo Zhengkui 17 May '22

17 May '22

Fix the following coccicheck warning: sound/core/seq/seq_ports.c:142:12-14: WARNING opportunity for max() Signed-off-by: Guo Zhengkui <guozhengkui(a)vivo.com> --- sound/core/seq/seq_ports.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 84d78630463e..25fcf5a2c71c 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -139,7 +139,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, port_subs_info_init(&new_port->c_dest); snd_use_lock_use(&new_port->use_lock); - num = port >= 0 ? port : 0; + num = max(port, 0); mutex_lock(&client->ports_mutex); write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { -- 2.20.1

2 1

[PATCH] selftests: alsa: Better error messages
by Siddh Raman Pant 16 May '22

16 May '22

This allows for potentially better machine-parsing due to an expected / fixed format. Also because of eyecandy reasons. Whenever possible, errors will start with the name of the offending card/control, separated by the error with colons. Whenever there is a description of an error given, the generated error string from snd_strerror will be (mostly) enclosed in parentheses at the end. Clarity of error messages have been (hopefully) improved. Signed-off-by: Siddh Raman Pant <siddhpant.gh(a)gmail.com> --- tools/testing/selftests/alsa/mixer-test.c | 94 +++++++++++------------ 1 file changed, 47 insertions(+), 47 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index a38b89c28..4719b12a5 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -114,7 +114,7 @@ static void find_controls(void) err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); if (err < 0) { - ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", + ksft_print_msg("Unable to parse custom alsa-lib configuration (%s)\n", snd_strerror(err)); ksft_exit_fail(); } @@ -128,7 +128,7 @@ static void find_controls(void) err = snd_ctl_open_lconf(&card_data->handle, name, 0, config); if (err < 0) { - ksft_print_msg("Failed to get hctl for card %d: %s\n", + ksft_print_msg("Card %d: Failed to get hctl (%s)\n", card, snd_strerror(err)); goto next_card; } @@ -177,9 +177,8 @@ static void find_controls(void) err = snd_ctl_elem_info(card_data->handle, ctl_data->info); if (err < 0) { - ksft_print_msg("%s getting info for %d\n", - snd_strerror(err), - ctl_data->name); + ksft_print_msg("%s : %s while getting info\n", + ctl_data->name, snd_strerror(err)); } snd_ctl_elem_value_set_id(ctl_data->def_val, @@ -192,20 +191,20 @@ static void find_controls(void) /* Set up for events */ err = snd_ctl_subscribe_events(card_data->handle, true); if (err < 0) { - ksft_exit_fail_msg("snd_ctl_subscribe_events() failed for card %d: %d\n", + ksft_exit_fail_msg("Card %d : snd_ctl_subscribe_events() failed (%d)\n", card, err); } err = snd_ctl_poll_descriptors_count(card_data->handle); if (err != 1) { - ksft_exit_fail_msg("Unexpected descriptor count %d for card %d\n", - err, card); + ksft_exit_fail_msg("Card %d : Unexpected descriptor count %d\n", + card, err); } err = snd_ctl_poll_descriptors(card_data->handle, &card_data->pollfd, 1); if (err != 1) { - ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", + ksft_exit_fail_msg("Card %d : snd_ctl_poll_descriptors() failed (%d)\n", card, err); } @@ -236,8 +235,8 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) do { err = poll(&(ctl->card->pollfd), 1, timeout); if (err < 0) { - ksft_print_msg("poll() failed for %s: %s (%d)\n", - ctl->name, strerror(errno), errno); + ksft_print_msg("%s : poll() failed (%d - %s)\n", + ctl->name, errno, strerror(errno)); return -1; } /* Timeout */ @@ -248,12 +247,12 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) &(ctl->card->pollfd), 1, &revents); if (err < 0) { - ksft_print_msg("snd_ctl_poll_descriptors_revents() failed for %s: %d\n", + ksft_print_msg("%s : snd_ctl_poll_descriptors_revents() failed (%d)\n", ctl->name, err); return err; } if (revents & POLLERR) { - ksft_print_msg("snd_ctl_poll_descriptors_revents() reported POLLERR for %s\n", + ksft_print_msg("%s : snd_ctl_poll_descriptors_revents() reported POLLERR\n", ctl->name); return -1; } @@ -265,7 +264,7 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) err = snd_ctl_read(ctl->card->handle, event); if (err < 0) { - ksft_print_msg("snd_ctl_read() failed for %s: %d\n", + ksft_print_msg("%s : snd_ctl_read() failed (%d)\n", ctl->name, err); return err; } @@ -283,7 +282,7 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) } if ((mask & SND_CTL_EVENT_MASK_REMOVE) == SND_CTL_EVENT_MASK_REMOVE) { - ksft_print_msg("Removal event for %s\n", + ksft_print_msg("%s : Removal event\n", ctl->name); return -1; } @@ -301,7 +300,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, switch (snd_ctl_elem_info_get_type(ctl->info)) { case SND_CTL_ELEM_TYPE_NONE: - ksft_print_msg("%s.%d Invalid control type NONE\n", + ksft_print_msg("%s.%d : Invalid control type NONE\n", ctl->name, index); return false; @@ -312,7 +311,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, case 1: break; default: - ksft_print_msg("%s.%d Invalid boolean value %ld\n", + ksft_print_msg("%s.%d : Invalid boolean value %ld\n", ctl->name, index, int_val); return false; } @@ -322,14 +321,14 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, int_val = snd_ctl_elem_value_get_integer(val, index); if (int_val < snd_ctl_elem_info_get_min(ctl->info)) { - ksft_print_msg("%s.%d value %ld less than minimum %ld\n", + ksft_print_msg("%s.%d : Value %ld is less than the minimum (%ld)\n", ctl->name, index, int_val, snd_ctl_elem_info_get_min(ctl->info)); return false; } if (int_val > snd_ctl_elem_info_get_max(ctl->info)) { - ksft_print_msg("%s.%d value %ld more than maximum %ld\n", + ksft_print_msg("%s.%d : Value %ld is more than the maximum (%ld)\n", ctl->name, index, int_val, snd_ctl_elem_info_get_max(ctl->info)); return false; @@ -339,7 +338,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, if (snd_ctl_elem_info_get_step(ctl->info) && (int_val - snd_ctl_elem_info_get_min(ctl->info) % snd_ctl_elem_info_get_step(ctl->info))) { - ksft_print_msg("%s.%d value %ld invalid for step %ld minimum %ld\n", + ksft_print_msg("%s.%d : Value %ld is invalid for step %ld, minimum %ld\n", ctl->name, index, int_val, snd_ctl_elem_info_get_step(ctl->info), snd_ctl_elem_info_get_min(ctl->info)); @@ -351,14 +350,14 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, int64_val = snd_ctl_elem_value_get_integer64(val, index); if (int64_val < snd_ctl_elem_info_get_min64(ctl->info)) { - ksft_print_msg("%s.%d value %lld less than minimum %lld\n", + ksft_print_msg("%s.%d : Value %lld is less than the minimum (%lld)\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_min64(ctl->info)); return false; } if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { - ksft_print_msg("%s.%d value %lld more than maximum %lld\n", + ksft_print_msg("%s.%d : Value %lld is more than the maximum (%lld)\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_max(ctl->info)); return false; @@ -368,7 +367,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, if (snd_ctl_elem_info_get_step64(ctl->info) && (int64_val - snd_ctl_elem_info_get_min64(ctl->info)) % snd_ctl_elem_info_get_step64(ctl->info)) { - ksft_print_msg("%s.%d value %lld invalid for step %lld minimum %lld\n", + ksft_print_msg("%s.%d : Value %lld is invalid for step %lld, minimum %lld\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_step64(ctl->info), snd_ctl_elem_info_get_min64(ctl->info)); @@ -380,13 +379,13 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, int_val = snd_ctl_elem_value_get_enumerated(val, index); if (int_val < 0) { - ksft_print_msg("%s.%d negative value %ld for enumeration\n", + ksft_print_msg("%s.%d : Negative value %ld for enumeration\n", ctl->name, index, int_val); return false; } if (int_val >= snd_ctl_elem_info_get_items(ctl->info)) { - ksft_print_msg("%s.%d value %ld more than item count %ld\n", + ksft_print_msg("%s.%d : Value %ld is more than item count (%ld)\n", ctl->name, index, int_val, snd_ctl_elem_info_get_items(ctl->info)); return false; @@ -427,7 +426,7 @@ static void test_ctl_get_value(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("get_value.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -435,7 +434,7 @@ static void test_ctl_get_value(struct ctl_data *ctl) /* Can't test reading on an unreadable control */ if (!snd_ctl_elem_info_is_readable(ctl->info)) { - ksft_print_msg("%s is not readable\n", ctl->name); + ksft_print_msg("%s : Not readable\n", ctl->name); ksft_test_result_skip("get_value.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -443,7 +442,7 @@ static void test_ctl_get_value(struct ctl_data *ctl) err = snd_ctl_elem_read(ctl->card->handle, ctl->def_val); if (err < 0) { - ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + ksft_print_msg("snd_ctl_elem_read() failed (%s)\n", snd_strerror(err)); goto out; } @@ -474,7 +473,7 @@ static void test_ctl_name(struct ctl_data *ctl) /* Only boolean controls should end in Switch */ if (strend(ctl->name, " Switch")) { if (snd_ctl_elem_info_get_type(ctl->info) != SND_CTL_ELEM_TYPE_BOOLEAN) { - ksft_print_msg("%d.%d %s ends in Switch but is not boolean\n", + ksft_print_msg("%d.%d %s : Not a boolean, but name ends in Switch\n", ctl->card->card, ctl->elem, ctl->name); name_ok = false; } @@ -484,7 +483,7 @@ static void test_ctl_name(struct ctl_data *ctl) if (snd_ctl_elem_info_get_type(ctl->info) == SND_CTL_ELEM_TYPE_BOOLEAN && snd_ctl_elem_info_is_writable(ctl->info)) { if (!strend(ctl->name, " Switch")) { - ksft_print_msg("%d.%d %s is a writeable boolean but not a Switch\n", + ksft_print_msg("%d.%d %s : Not a Switch despite being a writeable boolean\n", ctl->card->card, ctl->elem, ctl->name); name_ok = false; } @@ -542,11 +541,12 @@ static bool show_mismatch(struct ctl_data *ctl, int index, /* * NOTE: The volatile attribute means that the hardware * can voluntarily change the state of control element - * independent of any operation by software. + * independent of any operation by software. */ bool is_volatile = snd_ctl_elem_info_is_volatile(ctl->info); - ksft_print_msg("%s.%d expected %lld but read %lld, is_volatile %d\n", - ctl->name, index, expected_int, read_int, is_volatile); + ksft_print_msg("%s.%d : Expected %lld, but read %lld (%s)\n", + ctl->name, index, expected_int, read_int, + (is_volatile ? "Volatile" : "Non-volatile")); return !is_volatile; } else { return false; @@ -590,7 +590,7 @@ static int write_and_verify(struct ctl_data *ctl, err = snd_ctl_elem_read(ctl->card->handle, initial_val); if (err < 0) { - ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + ksft_print_msg("snd_ctl_elem_read() failed (%s)\n", snd_strerror(err)); return err; } @@ -602,7 +602,7 @@ static int write_and_verify(struct ctl_data *ctl, */ err = snd_ctl_elem_write(ctl->card->handle, w_val); if (err < 0 && !error_expected) { - ksft_print_msg("snd_ctl_elem_write() failed: %s\n", + ksft_print_msg("snd_ctl_elem_write() failed (%s)\n", snd_strerror(err)); return err; } @@ -615,7 +615,7 @@ static int write_and_verify(struct ctl_data *ctl, err = snd_ctl_elem_read(ctl->card->handle, read_val); if (err < 0) { - ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + ksft_print_msg("snd_ctl_elem_read() failed (%s)\n", snd_strerror(err)); return err; } @@ -631,13 +631,13 @@ static int write_and_verify(struct ctl_data *ctl, err = wait_for_event(ctl, 0); if (snd_ctl_elem_value_compare(initial_val, read_val)) { if (err < 1) { - ksft_print_msg("No event generated for %s\n", + ksft_print_msg("%s : No event generated\n", ctl->name); ctl->event_missing++; } } else { if (err != 0) { - ksft_print_msg("Spurious event generated for %s\n", + ksft_print_msg("%s : Spurious event generated\n", ctl->name); ctl->event_spurious++; } @@ -658,7 +658,7 @@ static int write_and_verify(struct ctl_data *ctl, mismatch_shown = true; if (!mismatch_shown) - ksft_print_msg("%s read and written values differ\n", + ksft_print_msg("%s : Read and written values differ\n", ctl->name); return -1; @@ -674,14 +674,14 @@ static void test_ctl_write_default(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("write_default.%d.%d\n", ctl->card->card, ctl->elem); return; } if (!snd_ctl_elem_info_is_writable(ctl->info)) { - ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_print_msg("%s : Not writeable\n", ctl->name); ksft_test_result_skip("write_default.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -689,7 +689,7 @@ static void test_ctl_write_default(struct ctl_data *ctl) /* No idea what the default was for unreadable controls */ if (!snd_ctl_elem_info_is_readable(ctl->info)) { - ksft_print_msg("%s couldn't read default\n", ctl->name); + ksft_print_msg("%s : Couldn't read default\n", ctl->name); ksft_test_result_skip("write_default.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -808,14 +808,14 @@ static void test_ctl_write_valid(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("write_valid.%d.%d\n", ctl->card->card, ctl->elem); return; } if (!snd_ctl_elem_info_is_writable(ctl->info)) { - ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_print_msg("%s : Not writeable\n", ctl->name); ksft_test_result_skip("write_valid.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -868,7 +868,7 @@ static bool test_ctl_write_invalid_value(struct ctl_data *ctl, /* ...but some devices will clamp to an in range value */ err = snd_ctl_elem_read(ctl->card->handle, val); if (err < 0) { - ksft_print_msg("%s failed to read: %s\n", + ksft_print_msg("%s : Failed to read (%s)\n", ctl->name, snd_strerror(err)); return true; } @@ -1026,14 +1026,14 @@ static void test_ctl_write_invalid(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("write_invalid.%d.%d\n", ctl->card->card, ctl->elem); return; } if (!snd_ctl_elem_info_is_writable(ctl->info)) { - ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_print_msg("%s : Not writeable\n", ctl->name); ksft_test_result_skip("write_invalid.%d.%d\n", ctl->card->card, ctl->elem); return; -- 2.35.1

3 4

[PATCH] kselftest: alsa: handle cross compilation
by Antonio Terceiro 16 May '22

16 May '22

Calling just `pkg-config` is adequate for native builds, but finding the foreign libraries with pkg-config needs pkg-config to be called via its architecture-specific wrapper. This works in Debian, where there is a corresponding *-pkg-config wrapper script for each enabled foreign architecture, just like there are *-gcc, *-ld. Signed-off-by: Antonio Terceiro <antonio.terceiro(a)linaro.org> --- tools/testing/selftests/alsa/Makefile | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index f64d9090426d..f5675b3c929d 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -1,8 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # -CFLAGS += $(shell pkg-config --cflags alsa) -LDLIBS += $(shell pkg-config --libs alsa) +PKG_CONFIG ?= $(CROSS_COMPILE)pkg-config + +CFLAGS += $(shell $(PKG_CONFIG) --cflags alsa) +LDLIBS += $(shell $(PKG_CONFIG) --libs alsa) TEST_GEN_PROGS := mixer-test -- 2.35.1

2 1

Complete support for "Pro" USB multichannel cards
by GitHub issues - opened 16 May '22

16 May '22

alsa-project/alsa-ucm-conf issue #154 was opened from perexg: - [x] channel split using alsa-lib - [ ] channel split inside app - [ ] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [ ] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/2] Revert "ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver"
by V sujith kumar Reddy 16 May '22

16 May '22

RT1019 codec has two ways of controlling the en_spkr. one way is controlling through gpio pin method the another way is through codec register update. Now Speaker enable/disable is controlled through register update in BIOS. So this patch reverse gpio logic, which is no longer in use. This reverts commit 7fa5c33d043160eba3be9fb8e21588dff2a467c7. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c855f50d6b34..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index bf61a1726f0e..8243765d490f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

3 3

[PATCH] ASoC: max98088: add support for reg_4a_cfg_bypass reg
by Tommaso Merciai 16 May '22

16 May '22

Add mixer controls support for M98088_REG_4A_CFG_BYPASS register References: - https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf p71, p113 Signed-off-by: Tommaso Merciai <tommaso.merciai(a)amarulasolutions.com> --- sound/soc/codecs/max98088.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 429717d4ac5a..f8ec2f164e08 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -486,6 +486,11 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_SINGLE("SPK Bypass Switch", M98088_REG_4A_CFG_BYPASS, 0, 1, 0), + SOC_SINGLE("REC Bypass Switch", M98088_REG_4A_CFG_BYPASS, 1, 1, 0), + SOC_SINGLE("MIC2 Bypass Switch", M98088_REG_4A_CFG_BYPASS, 4, 1, 0), + SOC_SINGLE("INA Bypass Switch", M98088_REG_4A_CFG_BYPASS, 7, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), -- 2.25.1

2 9

[PATCH] ALSA: usb-audio: Restore Rane SL-1 quirk
by Takashi Iwai 16 May '22

16 May '22

At cleaning up and moving the device rename from the quirk table to its own table, we removed the entry for Rane SL-1 as we thought it's only for renaming. It turned out, however, that the quirk is required for matching with the device that declares itself as no standard audio but only as vendor-specific. Restore the quirk entry for Rane SL-1 to fix the regression. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215887 Fixes: 5436f59bc5bc ("ALSA: usb-audio: Move device rename and profile quirks to an internal table") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..40a5e3eb4ef2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3235,6 +3235,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Rane SL-1 */ +{ + USB_DEVICE(0x13e5, 0x0001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + } +}, + /* disabled due to regression for other devices; * see https://bugzilla.kernel.org/show_bug.cgi?id=199905 */ -- 2.35.3

1 0

[bug report] ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages
by Dan Carpenter 16 May '22

16 May '22

Hello Peter Ujfalusi, The patch 066c67624d8c: "ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages" from May 6, 2022, leads to the following Smatch static checker warning: sound/soc/sof/sof-client-ipc-msg-injector.c:95 sof_msg_inject_ipc4_dfs_read() warn: userbuf overflow? is '8' <= 'count' sound/soc/sof/sof-client-ipc-msg-injector.c 72 static ssize_t sof_msg_inject_ipc4_dfs_read(struct file *file, 73 char __user *buffer, 74 size_t count, loff_t *ppos) 75 { 76 struct sof_client_dev *cdev = file->private_data; 77 struct sof_msg_inject_priv *priv = cdev->data; 78 struct sof_ipc4_msg *ipc4_msg = priv->rx_buffer; 79 size_t remaining; 80 81 if (!ipc4_msg->header_u64 || !count || *ppos) 82 return 0; 83 84 remaining = sizeof(ipc4_msg->header_u64); 85 86 /* Only get large config have payload */ 87 if (SOF_IPC4_MSG_IS_MODULE_MSG(ipc4_msg->primary) && 88 (SOF_IPC4_MSG_TYPE_GET(ipc4_msg->primary) == SOF_IPC4_MOD_LARGE_CONFIG_GET)) 89 remaining += ipc4_msg->data_size; 90 91 if (count > remaining) 92 count = remaining; 93 94 /* copy the header first */ --> 95 if (copy_to_user(buffer, &ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64))) 96 return -EFAULT; 97 98 *ppos += sizeof(ipc4_msg->header_u64); 99 remaining -= sizeof(ipc4_msg->header_u64); 100 101 if (!remaining) 102 return count; 103 104 if (remaining > ipc4_msg->data_size) 105 remaining = ipc4_msg->data_size; 106 107 /* Copy the payload */ 108 if (copy_to_user(buffer + *ppos, ipc4_msg->data_ptr, remaining)) ^^^^^^^^^^^^^^^ Potentially writing more than count bytes resulting in corrupting the user space memory. 109 return -EFAULT; 110 111 *ppos += remaining; 112 return count; 113 } regards, dan carpenter

2 1

snd_pcm_wait_nocheck - use more finer timeout for the the poll wait
by GitHub issues - edited 16 May '22

16 May '22

alsa-project/alsa-lib issue #228 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-lib/issues/228 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

axfer: call of polling system call never return when driver fails to queue no period-wakeup event due to the lack of hw IRQ context
by GitHub issues - edited 16 May '22

16 May '22

alsa-project/alsa-lib issue #228 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-lib/issues/228 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machine
by Andy Chi 16 May '22

16 May '22

* The HP EliteBook 630 is using ALC236 codec which used 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- 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 33c439578a61..3a1500e487eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9091,6 +9091,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8995, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x89a4, "HP ProBook 440 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89a6, "HP ProBook 450 G9", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), -- 2.25.1

2 1

[PATCH net-next] ALSA: ice1724: return void from snd_vt1724_chip_init()
by Lu Wei 16 May '22

16 May '22

Return int in snd_vt1724_chip_init() is useless since it always returns 0. Signed-off-by: Lu Wei <luwei32(a)huawei.com> --- sound/pci/ice1712/ice1724.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6fab2ad85bbe..dfef468ad6dc 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2347,7 +2347,7 @@ static void snd_vt1724_chip_reset(struct snd_ice1712 *ice) msleep(10); } -static int snd_vt1724_chip_init(struct snd_ice1712 *ice) +static void snd_vt1724_chip_init(struct snd_ice1712 *ice) { outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG)); outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); @@ -2369,8 +2369,6 @@ static int snd_vt1724_chip_init(struct snd_ice1712 *ice) * since they cause machine lockups */ outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); - - return 0; } static int snd_vt1724_spdif_build_controls(struct snd_ice1712 *ice) @@ -2506,8 +2504,8 @@ static int snd_vt1724_create(struct snd_card *card, snd_vt1724_chip_reset(ice); if (snd_vt1724_read_eeprom(ice, modelname) < 0) return -EIO; - if (snd_vt1724_chip_init(ice) < 0) - return -EIO; + + snd_vt1724_chip_init(ice); return 0; } @@ -2702,11 +2700,7 @@ static int snd_vt1724_resume(struct device *dev) return 0; snd_vt1724_chip_reset(ice); - - if (snd_vt1724_chip_init(ice) < 0) { - snd_card_disconnect(card); - return -EIO; - } + snd_vt1724_chip_init(ice); if (ice->pm_resume) ice->pm_resume(ice); -- 2.17.1

1 0

Weaknesses in alsa_info on new Ubuntu systems
by GitHub issues - opened 15 May '22

15 May '22

alsa-project/alsa-utils issue #146 was opened from mhampl: There are some weaknesses on current Ubuntu systems (e.g. release 22.04) where the alsa_info scripts does not deliver the desired information: The command "tree" is no more provided by default (but can be installed in addition) This results in the error message "./alsa-info.sh: line 661: tree: command not found" and causes missing output in "Sysfs card info" The command "dmesg" requires administrative authorizations (e.g. "sudo dmesg | grep ...") This results in the error message "dmesg: read kernel buffer failed: Operation not permitted" and causes missing output in "ALSA/HDA dmesg" Issue URL : https://github.com/alsa-project/alsa-utils/issues/146 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Missing documentation
by GitHub issues - opened 15 May '22

15 May '22

alsa-project/alsa-lib issue #227 was opened from fedeb95: Hi, this probably isn't the right place, but I couldn't find anything specific for documentation bugs. The links under https://www.alsa-project.org/wiki/ALSA_Driver_Documentation currently give a 404. Issue URL : https://github.com/alsa-project/alsa-lib/issues/227 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

chore: Unify Travis CI process to GitHub Actions
by GitHub pull_request - opened 15 May '22

15 May '22

alsa-project/alsa-lib pull request #226 was opened from Korilakkuma: [Travis are now recommending removing the sudo tag](https://blog.travis-ci.com/2018-11-19-required-linux-infrastructure-mi…. Request URL : https://github.com/alsa-project/alsa-lib/pull/226 Patch URL : https://github.com/alsa-project/alsa-lib/pull/226.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

chore: Add link to GitHub Actions
by GitHub pull_request - opened 15 May '22

15 May '22

alsa-project/alsa-lib pull request #225 was opened from Korilakkuma: This PR enables to transition GitHub Actions by clicking badge. Request URL : https://github.com/alsa-project/alsa-lib/pull/225 Patch URL : https://github.com/alsa-project/alsa-lib/pull/225.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: hdmi-codec: Add option for ELD bypass
by Sugar Zhang 14 May '22

14 May '22

This patch allow users to enable "ELD Bypass" who don't care anything from EDID Link Data. Currently, this driver gets ELD(from EDID) to constraint channels and rates. Unfortunately, EDID is not always valid, maybe caused by the fragile HDMI port or cable, in this situation, the max features are limited to 48kHz stereo. So, add this option to allow user to select the manual way to output audio as expected. such as multi-channels LPCM(7.1), or HBR bitstream for these sink devices. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - Use MACRO SOC_SINGLE_BOOL_EXT to simplify code. Fix event_missing checked by mixer-test. Add suffix "Switch" for "ELD Bypass". sound/soc/codecs/hdmi-codec.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b07607a..be46fbd 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -275,6 +275,7 @@ struct hdmi_codec_priv { unsigned int chmap_idx; struct mutex lock; bool busy; + bool eld_bypass; struct snd_soc_jack *jack; unsigned int jack_status; u8 iec_status[AES_IEC958_STATUS_SIZE]; @@ -427,6 +428,31 @@ static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } +static int hdmi_codec_eld_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = hcp->eld_bypass; + + return 0; +} + +static int hdmi_codec_eld_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + if (hcp->eld_bypass == ucontrol->value.integer.value[0]) + return 0; + + hcp->eld_bypass = ucontrol->value.integer.value[0]; + + return 1; +} + static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -447,7 +473,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, goto err; } - if (tx && hcp->hcd.ops->get_eld) { + if (tx && !hcp->eld_bypass && hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); if (ret) @@ -770,6 +796,8 @@ static struct snd_kcontrol_new hdmi_codec_controls[] = { .info = hdmi_eld_ctl_info, .get = hdmi_eld_ctl_get, }, + SOC_SINGLE_BOOL_EXT("ELD Bypass Switch", 0, + hdmi_codec_eld_bypass_get, hdmi_codec_eld_bypass_put), }; static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, @@ -854,7 +882,7 @@ static void plugged_cb(struct device *dev, bool plugged) struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); if (plugged) { - if (hcp->hcd.ops->get_eld) { + if (!hcp->eld_bypass && hcp->hcd.ops->get_eld) { hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); } -- 2.7.4

3 3

Add "Pro Audio" profile for UMC204HD
by GitHub pull_request - opened 14 May '22

14 May '22

alsa-project/alsa-ucm-conf pull request #153 was opened from brndd: Follow-up to #128 I figured that some users might prefer to keep the channel mapping untouched from what the card defaults to when unconfigured. Providing an alternative profile like this (implementation adapted from #145) for these users seems like a good idea. I believe PipeWire autogenerates profiles like these for unconfigured devices, so that's where the "Pro Audio" nomenclature comes from. Likely a good idea to match that since PipeWire is increasingly the default on most distributions. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/153 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/153.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] sound: sound_core: changed spaces to tabs
by Paul Lemmermann 14 May '22

14 May '22

changed spaces to tabs in sound_core.c according to kernel styling guidelines. found using checkpatch.pl. Signed-off-by: Paul Lemmermann <thepaulodoom(a)thepaulodoom.com> --- sound/sound_core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 3332fe321..28bc038dc 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -290,7 +290,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -395,12 +395,12 @@ int register_sound_special_device(const struct file_operations *fops, int unit, name = "admmidi"; break; default: - { + { __unknown: sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; } break; } -- 2.35.1

1 0

Re: [PATCH 19/32] afs: Use mem_to_flex_dup() with struct afs_acl
by David Howells 13 May '22

13 May '22

Kees Cook <keescook(a)chromium.org> wrote: > struct afs_acl { > - u32 size; > - u8 data[]; > + DECLARE_FLEX_ARRAY_ELEMENTS_COUNT(u32, size); > + DECLARE_FLEX_ARRAY_ELEMENTS(u8, data); > }; Oof... That's really quite unpleasant syntax. Is it not possible to have mem_to_flex_dup() and friends work without that? You are telling them the fields they have to fill in. > + struct afs_acl *acl = NULL; > > - acl = kmalloc(sizeof(*acl) + size, GFP_KERNEL); > - if (!acl) { > + if (mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL)) { Please don't do that. Either do: acl = mem_to_flex_dup(buffer, size, GFP_KERNEL); if (!acl) or: acl = mem_to_flex_dup(buffer, size, GFP_KERNEL); if (IS_ERR(acl)) Please especially don't make it that an apparent 'true' return indicates an error. If you absolutely must return the acl pointer through the argument list (presumably because it's actually a macro), make it return false on failure: if (!mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL)) or return and explicitly check for an error code: if (mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL) < 0) or: ret = mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL); if (ret < 0) (or use != 0 rather than < 0) David

2 1

[PATCH 1/2 v2] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 13 May '22

13 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- Changes since v1: split new bindings and documentation into separate patches sound/soc/codecs/adau7118.c | 62 ++++++++++++++++++++++++++++++++++--- 1 file changed, 58 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

2 5

[PATCH v2] ALSA: hda/realtek: Add quirk for TongFang devices with pop noise
by Werner Sembach 13 May '22

13 May '22

When audio stops playing there is an audible "pop"-noise when using headphones on the TongFang GMxMRxx, GKxNRxx, GMxZGxx, GMxTGxx and GMxAGxx. This quirk fixes this mostly. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cf531c1efa132..8003bf7441d6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9290,6 +9290,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.25.1

2 1

Re: [PATCH 00/32] Introduce flexible array struct memcpy() helpers
by David Howells 13 May '22

13 May '22

Kees Cook <keescook(a)chromium.org> wrote: > I'm happy to also point out that the conversions (patches 5+) are actually > a net reduction in lines of code: > 49 files changed, 154 insertions(+), 244 deletions(-) That doesn't mean that it's actually code that's clearer to read. I would say that it's actually less clear. In a bunch of places, you've done something like: - e = kmalloc(...); - if (!e) + if (__mem_to_flex_dup(&e, ...)) The problem is that, to me at least, it looks like: - e = kmalloc(...); - if (kmalloc failed) + if (__mem_to_flex_dup(&e, ...) succeeded) David

1 0

[PATCH] ALSA: hda/intel: Add quirk for TongFang devices with pop noise
by Werner Sembach 12 May '22

12 May '22

When audio stops playing and sometimes when it starts playing, there is an audible "pop" noise when using headphones on most Tongfang GMxMxxx, GKxNxxx, GMxZxxx, GMxTxxx, and GMxAxxx devices. Disabling power saving for the Realtek codec fixes this noise. Presumably it is triggered on some power event in the audio circuit. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1f..8a1088e057ec0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2201,6 +2201,13 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Several TongFang barebones making popping sounds */ + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAxxx", 0), {} }; #endif /* CONFIG_PM */ -- 2.25.1

2 4

[PATCH -next] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Zheng Bin 12 May '22

12 May '22

acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index b8910bb7f27c..d5d9bcc2c997 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -101,6 +101,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.31.1

3 2

[PATCH] ASoC: samsung: Fix refcount leak in aries_audio_probe
by Miaoqian Lin 12 May '22

12 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when done. If extcon_find_edev_by_node() fails, it doesn't call of_node_put() Calling of_node_put() after extcon_find_edev_by_node() to fix this. Fixes: 7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index 5265e546b124..83acbe57b248 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -585,10 +585,10 @@ static int aries_audio_probe(struct platform_device *pdev) extcon_np = of_parse_phandle(np, "extcon", 0); priv->usb_extcon = extcon_find_edev_by_node(extcon_np); + of_node_put(extcon_np); if (IS_ERR(priv->usb_extcon)) return dev_err_probe(dev, PTR_ERR(priv->usb_extcon), "Failed to get extcon device"); - of_node_put(extcon_np); priv->adc = devm_iio_channel_get(dev, "headset-detect"); if (IS_ERR(priv->adc)) -- 2.25.1

4 3

[PATCH 0/5] ASoC: SOF: Add IPC4 FW loader support
by Ranjani Sridharan 12 May '22

12 May '22

The patches in this series add support for FW loading for IPC4 in the SOF driver. Ranjani Sridharan (5): ASoC: SOF: Intel: HDA: Set IPC4-specific DSP ops for CaVS platforms ASoC: SOF: Add IPC4 private header ASoC: SOF: Add header for IPC4 manifest ASoC: SOF: IPC4: Add FW loader ops ASoC: SOF: Intel: HDA: Set up sof_ipc4_fw_data for IPC4 include/sound/sof/ext_manifest4.h | 119 +++++++++++++++++ sound/soc/sof/Makefile | 2 +- sound/soc/sof/intel/apl.c | 29 ++++- sound/soc/sof/intel/cnl.c | 109 +++++++++++++++- sound/soc/sof/intel/hda-ipc.c | 82 ++++++++++++ sound/soc/sof/intel/hda.h | 6 + sound/soc/sof/intel/tgl.c | 29 ++++- sound/soc/sof/ipc4-loader.c | 210 ++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 44 +++++++ sound/soc/sof/ipc4.c | 2 + 10 files changed, 619 insertions(+), 13 deletions(-) create mode 100644 include/sound/sof/ext_manifest4.h create mode 100644 sound/soc/sof/ipc4-loader.c create mode 100644 sound/soc/sof/ipc4-priv.h -- 2.25.1

2 6

[PATCH -next v2 0/2] ASoC: codecs: Fix build error
by Hui Tang 12 May '22

12 May '22

Fix two build error, as follows: Changes in v2: * Add missing header files instead of adding dependencies. Hui Tang (2): ASoC: max98396: Fix build error for implicit function declaration ASoC: tlv320adc3xxx: Fix build error for implicit function declaration sound/soc/codecs/max98396.c | 1 + sound/soc/codecs/tlv320adc3xxx.c | 3 +++ 2 files changed, 4 insertions(+) -- 2.17.1

2 3

[PATCH] ASoC: imx-hdmi: Fix refcount leak in imx_hdmi_probe
by Miaoqian Lin 12 May '22

12 May '22

of_find_device_by_node() takes reference, we should use put_device() to release it. when devm_kzalloc() fails, it doesn't have a put_device(), it will cause refcount leak. Add missing put_device() to fix this. Fixes: 6a5f850aa83a ("ASoC: fsl: Add imx-hdmi machine driver") Fixes: f670b274f7f6 ("ASoC: imx-hdmi: add put_device() after of_find_device_by_node()") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 929f69b758af..ec149dc73938 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -126,6 +126,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; + put_device(&cpu_pdev->dev); goto fail; } -- 2.25.1

2 1

[PATCH] ASoC: mxs-saif: Fix refcount leak in mxs_saif_probe
by Miaoqian Lin 12 May '22

12 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when done. Fixes: 08641c7c74dd ("ASoC: mxs: add device tree support for mxs-saif") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mxs/mxs-saif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 879c1221a809..7afe1a1acc56 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -754,6 +754,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->master_id = saif->id; } else { ret = of_alias_get_id(master, "saif"); + of_node_put(master); if (ret < 0) return ret; else -- 2.25.1

2 1

[PATCH 1/2] ASoC: ops: Fix bounds check for _sx controls
by Mark Brown 12 May '22

12 May '22

For _sx controls the semantics of the max field is not the usual one, max is the number of steps rather than the maximum value. This means that our check in snd_soc_put_volsw_sx() needs to just check against the maximum value. Fixes: 4f1e50d6a9cf9c1b ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx()") Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..1ac7e2ce31a1 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -435,7 +435,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; - if (val > max - min) + if (val > max) return -EINVAL; val_mask = mask << shift; val = (val + min) & mask; -- 2.30.2

1 2

[PATCH] ASoC: fsl: Fix refcount leak in imx_sgtl5000_probe
by Miaoqian Lin 12 May '22

12 May '22

of_find_i2c_device_by_node() takes a reference, In error paths, we should call put_device() to drop the reference to aviod refount leak. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/fsl/imx-sgtl5000.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 8daced42d55e..580a0d963f0e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -120,19 +120,19 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; - goto fail; + goto put_device; } comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL); if (!comp) { ret = -ENOMEM; - goto fail; + goto put_device; } data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); - goto fail; + goto put_device; } data->clk_frequency = clk_get_rate(data->codec_clk); @@ -158,10 +158,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto fail; + goto put_device; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto fail; + goto put_device; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -174,7 +174,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); - goto fail; + goto put_device; } of_node_put(ssi_np); @@ -182,6 +182,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; +put_device: + put_device(&codec_dev->dev); fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); -- 2.25.1

3 2

[PATCH 0/6] ASOC: Fix the error handling code of the probe
by Zheyu Ma 12 May '22

12 May '22

These drivers mishandle the regulator resource in the probe function, failing to disable the regulator for probing failure. Zheyu Ma (6): ASoC: cs42l52: Fix the error handling of cs42l56_i2c_probe() ASoC: cs35l36: Fix the error handling of cs35l36_i2c_probe() ASoC: rt5645: Fix the error handling of rt5645_i2c_probe() ASoC: tas571x: Fix the error handling of tas571x_i2c_probe() ASoC: tas6424: Fix the error handling of tas6424_i2c_probe() ASoC: wm8903: Fix the error handling of wm8903_i2c_probe() sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/tas571x.c | 6 ++++-- sound/soc/codecs/tas6424.c | 8 ++++++-- sound/soc/codecs/wm8903.c | 2 +- 6 files changed, 15 insertions(+), 9 deletions(-) -- 2.25.1

3 12

[PATCH] ASoC: fsl_ssi: Add support multi fifo script
by Shengjiu Wang 12 May '22

12 May '22

With dual fifo enabled, the case that recording mono sound in the background, playback mono sound twice in parallal, at second time playback sound may distort, the possible reason is using dual fifo to playback mono sound is not recommended. This patch is to provide a option to use multi fifo script, which can be dynamically configured as one fifo or two fifo mode. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 34 ++++++++++++++++++++++++++++++++-- 1 file changed, 32 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ca30a4ede076..84cb36d9dfea 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -40,6 +40,7 @@ #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/dma/imx-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -214,6 +215,7 @@ struct fsl_ssi_soc_data { * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for dual FIFO mode + * @use_dyna_fifo: DMA with support for multi FIFO script * @has_ipg_clk_name: If "ipg" is in the clock name list of device tree * @fifo_depth: Depth of the SSI FIFOs * @slot_width: Width of each DAI slot @@ -243,6 +245,7 @@ struct fsl_ssi_soc_data { * @dma_maxburst: Max number of words to transfer in one go. So far, * this is always the same as fifo_watermark. * @ac97_reg_lock: Mutex lock to serialize AC97 register access operations + * @audio_config: configure for dma multi fifo script */ struct fsl_ssi { struct regmap *regs; @@ -255,6 +258,7 @@ struct fsl_ssi { bool synchronous; bool use_dma; bool use_dual_fifo; + bool use_dyna_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; unsigned int slot_width; @@ -287,6 +291,7 @@ struct fsl_ssi { u32 dma_maxburst; struct mutex ac97_reg_lock; + struct sdma_peripheral_config audio_config[2]; }; /* @@ -643,7 +648,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * task from fifo0, fifo1 would be neglected at the end of each * period. But SSI would still access fifo1 with an invalid data. */ - if (ssi->use_dual_fifo) + if (ssi->use_dual_fifo || ssi->use_dyna_fifo) snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2); @@ -802,6 +807,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { bool tx2, tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(dai); + struct fsl_ssi_regvals *vals = ssi->regvals; struct regmap *regs = ssi->regs; unsigned int channels = params_channels(hw_params); unsigned int sample_size = params_width(hw_params); @@ -856,6 +862,28 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, tx2 = tx || ssi->synchronous; regmap_update_bits(regs, REG_SSI_SxCCR(tx2), SSI_SxCCR_WL_MASK, wl); + if (ssi->use_dyna_fifo) { + if (channels == 1) { + ssi->audio_config[0].n_fifos_dst = 1; + ssi->audio_config[1].n_fifos_src = 1; + vals[RX].srcr &= ~SSI_SRCR_RFEN1; + vals[TX].stcr &= ~SSI_STCR_TFEN1; + vals[RX].scr &= ~SSI_SCR_TCH_EN; + vals[TX].scr &= ~SSI_SCR_TCH_EN; + } else { + ssi->audio_config[0].n_fifos_dst = 2; + ssi->audio_config[1].n_fifos_src = 2; + vals[RX].srcr |= SSI_SRCR_RFEN1; + vals[TX].stcr |= SSI_STCR_TFEN1; + vals[RX].scr |= SSI_SCR_TCH_EN; + vals[TX].scr |= SSI_SCR_TCH_EN; + } + ssi->dma_params_tx.peripheral_config = &ssi->audio_config[0]; + ssi->dma_params_tx.peripheral_size = sizeof(ssi->audio_config[0]); + ssi->dma_params_rx.peripheral_config = &ssi->audio_config[1]; + ssi->dma_params_rx.peripheral_size = sizeof(ssi->audio_config[1]); + } + return 0; } @@ -1353,7 +1381,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi->dma_params_rx.addr = ssi->ssi_phys + REG_SSI_SRX0; /* Use even numbers to avoid channel swap due to SDMA script design */ - if (ssi->use_dual_fifo) { + if (ssi->use_dual_fifo || ssi->use_dyna_fifo) { ssi->dma_params_tx.maxburst &= ~0x1; ssi->dma_params_rx.maxburst &= ~0x1; } @@ -1446,6 +1474,8 @@ static int fsl_ssi_probe_from_dt(struct fsl_ssi *ssi) if (ssi->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) ssi->use_dual_fifo = true; + if (ssi->use_dma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + ssi->use_dyna_fifo = true; /* * Backward compatible for older bindings by manually triggering the * machine driver's probe(). Use /compatible property, including the -- 2.17.1

2 1

[PATCH 1/2] ASoC: wm2000: Remove redundant endianness flag
by Charles Keepax 12 May '22

12 May '22

The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device has no DAI links and as such the flag would have no effect, remove the redundant flag. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm2000.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 075ee852eb724..6a86ce3fb182d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -803,7 +803,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm2000 = { .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), .idle_bias_on = 1, .use_pmdown_time = 1, - .endianness = 1, .non_legacy_dai_naming = 1, }; -- 2.30.2

2 2

[PATCH 0/3] firewire: fix minor issues
by Takashi Sakamoto 12 May '22

12 May '22

Hi, This patchset includes below patches posted by several developers. * [PATCH v2] firewire: convert sysfs sprintf/snprintf family to sysfs_emit * https://lore.kernel.org/lkml/1612424420-96871-1-git-send-email-jiapeng.chon… * [PATCH V2] drivers/firewire: use struct_size over open coded arithmetic * https://lore.kernel.org/lkml/20220210060805.1608198-1-chi.minghao@zte.com.c… * [PATCH] firewire: Fix using uninitialized value * https://lore.kernel.org/lkml/20220411105205.2520784-1-lv.ruyi@zte.com.cn/ They are for some minor issues, while preferable. I expect the patches are going to be sent to Linus for next merge window by maintainer of either Linux FireWire subsystem or sound subsystem. Regards Jiapeng Chong (1): firewire: convert sysfs sprintf/snprintf family to sysfs_emit Lv Ruyi (1): firewire: Fix using uninitialized value Minghao Chi (CGEL ZTE) (1): firewire: use struct_size over open coded arithmetic drivers/firewire/core-device.c | 6 ++---- drivers/firewire/core-transaction.c | 2 +- 2 files changed, 3 insertions(+), 5 deletions(-) -- 2.34.1

1 3

[PATCH v7 0/2] Add generic serial MIDI driver using serial bus API
by Daniel Kaehn 12 May '22

12 May '22

Generic serial MIDI driver adding support for using serial devices compatible with the serial bus as raw MIDI devices, allowing using additional serial devices not compatible with the existing serial-u16550 driver. Supports only setting standard serial baudrates on the underlying serial device; however, the underlying serial device can be configured so that a requested 38.4 kBaud is actually the standard MIDI 31.25 kBaud. Supports DeviceTree configuration. Changes in v7: - Separate examples in dt-binding to remove need to specify unit name (fixing dt_binding_check error) Changes in v6: - Change compatible "serialmidi" -> serial-midi" - Default current-speed to 38400 (closest baud to MIDI standard speed) - Appropriately stop reading or writing MIDI if input or output _trigger() is called with a parameter of zero, respectively - Zero out corresponding triggered state on close to ensure input and output closing results in the serial port being closed - Fix order of operations in _probe() - Remove "DEBUG" literal from debug messages - Remove unused dt-parsing patch checking for existence of node - Whitespace / tabbing fixes / improvements Changes in v5: - Reword description in dt-binding for clarity - Change 'speed' dt property to standard 'current-speed' - Move MIDI output loop onto workqueue (since this could loop quite a while, if ALSA provides a continuous stream of bytes) - Add tx_state bit flags to snd_serial_generic struct - Safegard critical section in tx_work with atomic bit ops on tx_state - Switch operations on filemode to use atomic bit ops Changes in v4: - Fix regressed typo - Correct 3.84 kBaud -> 38.4 kBaud in DT & Kconfig (sorry about spam - noticed after sending v3 and didn't want to let the error sit around for too long) Changes in v3: - Replace use of snd_printk() with dev_* alternatives - Removed unnecessary initialization of err variables - Replaced instances of `== SERIAL_MODE_NOT_OPENED` with zero check - Loop on output_write to completely fill output buffer if data available - Depend on CONFIG_OF in Kconfig - Replace use of devm_kzalloc() with extra_size allocation in snd_devm_card_new() - Use module_serdev_device_driver() instead of module_init() and module_exit(0) Changes in v2: - Fix 'snd_serial_generic_write_wakeup' missing static keyword - Correct 3.125 kBaud > 31.25 kBaud in documentation for MIDI The need for this driver arose from a project using a Raspberry Pi4 which needed to receive and send raw MIDI with low latency. The pl011 UART used is not compatible with the existing serial MIDI driver made for u16550-style devices. Using a userspace program such as ttymidi to feed input from the TTY device to a virtual ALSA MIDI device was functional, but not ideal. I am not sure if a MIDI driver needing the mentioned 'hack' to clock 38.4 kBaud down to the standard MIDI baud is permissible in the mainline kernel, but am submitting nevertheless in case it is useful. To my knowledge, it doesn't seem that there would be any way for this driver to manually configure a serial port to 31.25 kBaud using the serial bus API (please correct me f I'm wrong). In my use case, I am actually configuring one port to run at 115.2 kBaud for faster communication with a custom onboard MIDI controller. Daniel Kaehn (2): dt-bindings: sound: Add generic serial MIDI device Add generic serial MIDI driver using serial bus API .../bindings/sound/serial-midi.yaml | 51 +++ sound/drivers/Kconfig | 18 + sound/drivers/Makefile | 2 + sound/drivers/serial-generic.c | 374 ++++++++++++++++++ 4 files changed, 445 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/serial-midi.yaml create mode 100644 sound/drivers/serial-generic.c base-commit: c5eb0a61238dd6faf37f58c9ce61c9980aaffd7a -- 2.33.0

3 5

[PATCH v2 0/3] ALSA: hda/cs8409: Add support for Odin Laptop Variants
by Stefan Binding 12 May '22

12 May '22

Add support for new Odin Laptop Variants into CS8409 HDA Driver. Since these laptops require Speaker Playback Switch, support must be added into CS8409 HDA driver. Since CS8409 does not have amplifier parameters for the NID associated with the speaker, the HDA driver does not add the switch automatically, so the driver needs to add this support manually. changes since v1: - add missing break in switch statement Stefan Binding (3): ALSA: hda/cs8409: Support new Odin Variants ALSA: hda/cs8409: Add Speaker Playback Switch for Cyborg ALSA: hda/cs8409: Add Speaker Playback Switch for Warlock sound/pci/hda/patch_cs8409-tables.c | 19 +++++++ sound/pci/hda/patch_cs8409.c | 86 ++++++++++++++++++++++++----- sound/pci/hda/patch_cs8409.h | 5 ++ 3 files changed, 96 insertions(+), 14 deletions(-) -- 2.25.1

2 4

[PATCH -next] ALSA: ppc: tumbler: use for_each_child_of_node() macro
by Yang Yingliang 12 May '22

12 May '22

Use for_each_child_of_node() macro instead of open coding it. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/ppc/tumbler.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index c65e74d7cd0a..c7e150f82bcc 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1060,8 +1060,7 @@ static struct device_node *find_audio_device(const char *name) if (! gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { const char *property = of_get_property(np, "audio-gpio", NULL); if (property && strcmp(property, name) == 0) break; -- 2.25.1

1 0

[PATCH] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 12 May '22

12 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Update yaml documentation accordingly. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- .../bindings/sound/adi,adau7118.yaml | 24 +++++++ sound/soc/codecs/adau7118.c | 62 +++++++++++++++++-- 2 files changed, 82 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index fb78967ee17b..71e8a9ff2edf 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -51,6 +51,27 @@ properties: maximum: 1 default: [0, 0, 1, 1] + adi,pdm-clk0-ds: + description: | + This property set the drive strength of PDM CLK0 output pad. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 2, 1, 0] + default: 2 + + adi,pdm-clk1-ds: + description: | + This property set the drive strength of PDM CLK1 output pad. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 2, 1, 0] + default: 2 + + adi,sdata-ds: + description: | + This property set the drive strength of SDATA output pad. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 2, 1, 0] + default: 2 + required: - "#sound-dai-cells" - compatible @@ -73,6 +94,9 @@ examples: dvdd-supply = <&supply>; adi,pdm-clk-map = <1 1 0 0>; adi,decimation-ratio = <16>; + adi,pdm-clk0-ds = <3>; + adi,pdm-clk1-ds = <3>; + adi,sdata-ds = <3>; }; }; diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

2 1

[PATCH -next 0/2] ASoC: codecs: Fix build error
by Hui Tang 12 May '22

12 May '22

Fix two build error, as follows: tanghui (2): ASoC: max98396: Fix build error for implicit function declaration ASoC: tlv320adc3xxx: Fix build error for implicit function declaration sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) -- 2.17.1

2 4

Re: kismet: WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A when selected by SND_SOC_INTEL_SOF_CS42L42_MACH
by Pierre-Louis Bossart 11 May '22

11 May '22

On 5/10/22 22:36, kernel test robot wrote: > tree: > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: feb9c5e19e913b53cb536a7aa7c9f20107bb51ec > commit: 5a7f27a624d9e33262767b328aa7a4baf7846c14 ASoC: Intel: add > sof-cs42l42 machine driver > date: 1 year ago > config: (attached as .config) > reproduce: > # > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… > > git remote add linus > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > git fetch --no-tags linus master > git checkout 5a7f27a624d9e33262767b328aa7a4baf7846c14 > # 1. reproduce by kismet > # install kmax per https://github.com/paulgazz/kmax/README.md > kismet --linux-ksrc=linux --selectees > CONFIG_SND_SOC_MAX98357A --selectors > CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH -a=x86_64 > # 2. reproduce by make > # save the config file to linux source tree > cd linux > make ARCH=x86_64 olddefconfig > > If you fix the issue, kindly add following tag as appropriate > Reported-by: kernel test robot <yujie.liu(a)intel.com> > > > kismet warnings: (new ones prefixed by >>) >>> kismet: WARNING: unmet direct dependencies detected for > SND_SOC_MAX98357A when selected by SND_SOC_INTEL_SOF_CS42L42_MACH > > WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A > Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && > GPIOLIB [=n] > Selected by [y]: > - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND > [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK > [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && > SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && > (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) I can reproduce this, but I question the hard dependency on GPIOLIB in the amplifier Kconfig: max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev, "sdmode", GPIOD_OUT_LOW); If the gpio is optional, max98357a.c should build and compile without GPIOLIB, no? > WARNING: unmet direct dependencies detected for SND_SOC_DMIC > Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && > GPIOLIB [=n] > Selected by [y]: > - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND > [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK > [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && > SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && > (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) I can't reproduce this one with Mark's for-next branch, this was fixed I believe.

1 1

[PATCH 1/2] ASoc: fsl_micfil: explicitly clear software reset bit
by Shengjiu Wang 11 May '22

11 May '22

SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined as non volatile register, it still remain in regmap cache after set, then every update of REG_MICFIL_CTRL1, software reset happens. to avoid this, clear it explicitly. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index cd85c8714f97..2149fac0dcc6 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -179,6 +179,17 @@ static int fsl_micfil_reset(struct device *dev) if (ret) return ret; + /* + * SRES is self-cleared bit, but REG_MICFIL_CTRL1 is defined + * as non-volatile register, so SRES still remain in regmap + * cache after set, that every update of REG_MICFIL_CTRL1, + * software reset happens. so clear it explicitly. + */ + ret = regmap_clear_bits(micfil->regmap, REG_MICFIL_CTRL1, + MICFIL_CTRL1_SRES); + if (ret) + return ret; + return 0; } -- 2.17.1

2 2

[PATCH 0/3] ASoC: Intel: boards: Chromebook configuration updates
by Pierre-Louis Bossart 11 May '22

11 May '22

two minor changes to enable DMIC and capture for CS35L41, and one new configuration for AlderLake hardware. Brent Lu (2): ASoC: Intel: sof_ssp_amp: fix no DMIC BE Link on Chromebooks ASoC: Intel: sof_ssp_amp: enable capture stream from cs35l41 Vamshi Krishna (1): ASoC: Intel: sof_rt5682: Add support for adl_rt1019_rt5682s sound/soc/intel/boards/sof_realtek_common.c | 39 +++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 5 +++ sound/soc/intel/boards/sof_rt5682.c | 13 +++++++ sound/soc/intel/boards/sof_ssp_amp.c | 19 ++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++++ 5 files changed, 81 insertions(+), 2 deletions(-) -- 2.30.2

2 4

[PATCH] [v2] ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 11 May '22

11 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 88 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 8 ++ 2 files changed, 92 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..4ac4f51c7c115 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,52 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return ret; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +578,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +607,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +632,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +645,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +690,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; 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 7c8cd00457f81..3f40519250a90 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.31.0

3 4

[v5] ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 11 May '22

11 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 92 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 95 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..442f13de6cd55 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,56 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) { + ret = -ENOMEM; + goto devm_err; + } + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) { + ret = -ENOMEM; + goto devm_err; + } + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +582,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +611,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +636,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +649,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +694,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; 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 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

2 1

[v4] ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 11 May '22

11 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 88 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 8 ++ 2 files changed, 92 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..4ac4f51c7c115 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,52 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return ret; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +578,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +607,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +632,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +645,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +690,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; 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 7c8cd00457f81..3f40519250a90 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

1 0

[PATCH] [v3]FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 11 May '22

11 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 92 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 8 ++ 2 files changed, 96 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..e7711480922cd 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,56 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) { + ret = -ENOMEM; + goto devm_err; + } + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) { + ret = -ENOMEM; + goto devm_err; + } + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +582,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +611,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +636,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +649,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +694,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; 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 7c8cd00457f81..3f40519250a90 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

2 1

[PATCH v2] ALSA: hda/realtek: Add quirk for the Framework Laptop
by Dustin L. Howett 11 May '22

11 May '22

Some board revisions of the Framework Laptop have an ALC295 with a disconnected or faulty headset mic presence detect. The "dell-headset-multi" fixup addresses this issue, but also enables an inoperative "Headphone Mic" input device whenever a headset is connected. Adding a new quirk chain specific to the Framework Laptop resolves this issue. The one introduced here is based on the System76 "no headphone mic" quirk chain. The VID:PID f111:0001 have been allocated to Framework Computer for this board revision. Revision history: - v2: Moved to a custom quirk chain to suppress the "Headphone Mic" pincfg. Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c0c593f3c0a..e00e6d2038a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7012,6 +7012,7 @@ enum { ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, + ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -8806,6 +8807,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_MUTE_LED, }, + [ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1112c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9294,6 +9304,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), + SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. -- 2.36.0

2 1

[PATCH v8 0/3] firmware: mtk: add adsp ipc protocol for SOF
by Tinghan Shen 11 May '22

11 May '22

This patch provides mediatek adsp ipc support for SOF. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This patch was tested and confirmed to work with SOF fw on MT8195 cherry board and MT8186 krabby board. changes since v7: - rebase to linux-next/next-22020504 - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL in mtk-adsp-ipc.c - move mtk-adsp-ipc.c out from driver/firmware/mediatek - add user of mtk-adsp-ipc.h in patchset 2 and 3. changes since v6: - rebase to matthias.bgg/linux.git, v5.18-next/soc - Prefer "GPL" over "GPL v2" for MODULE_LICENSE changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): ASoC: SOF: mediatek: Add ipc support for mt8195 TingHan Shen (1): firmware: mediatek: add adsp ipc protocol interface Tinghan Shen (1): ASoC: SOF: mediatek: Add mt8186 ipc support drivers/firmware/Kconfig | 9 + drivers/firmware/Makefile | 1 + drivers/firmware/mtk-adsp-ipc.c | 157 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 ++++++++ sound/soc/sof/mediatek/Kconfig | 1 + sound/soc/sof/mediatek/adsp_helper.h | 12 +- sound/soc/sof/mediatek/mt8186/mt8186-loader.c | 5 + sound/soc/sof/mediatek/mt8186/mt8186.c | 141 ++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 138 ++++++++++++++- 9 files changed, 518 insertions(+), 11 deletions(-) create mode 100644 drivers/firmware/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

1 3

[PATCH v1 0/3] ALSA: hda/cs8409: Add support for Odin Laptop Variants
by Stefan Binding 11 May '22

11 May '22

Add support for new Odin Laptop Variants into CS8409 HDA Driver. Since these laptops require Speaker Playback Switch, support must be added into CS8409 HDA driver. Since CS8409 does not have amplifier parameters for the NID associated with the speaker, the HDA driver does not add the switch automatically, so the driver needs to add this support manually. Stefan Binding (3): ALSA: hda/cs8409: Support new Odin Variants ALSA: hda/cs8409: Add Speaker Playback Switch for Cyborg ALSA: hda/cs8409: Add Speaker Playback Switch for Warlock sound/pci/hda/patch_cs8409-tables.c | 19 +++++++ sound/pci/hda/patch_cs8409.c | 85 ++++++++++++++++++++++++----- sound/pci/hda/patch_cs8409.h | 5 ++ 3 files changed, 95 insertions(+), 14 deletions(-) -- 2.25.1

2 4

[PATCH AUTOSEL 4.9 4/4] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 74968ddee49f..90ba5521c189 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -528,7 +528,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -543,6 +551,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 4.9 3/4] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 4aefb13900c2..1a55f6aecdfe 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -436,7 +436,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 4.9 2/4] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index e7aef841f87d..4aefb13900c2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -419,6 +419,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 4/5] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index d6d72595fbd0..0848aec1bd24 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -528,7 +528,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -543,6 +551,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 3/5] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index a25183f6fba6..362d1fb2aae9 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -436,7 +436,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 2/5] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 3fe09828745a..a25183f6fba6 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -419,6 +419,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 4/6] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index c88bc6bb41cf..7a37312c8e0c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -523,7 +523,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -538,6 +546,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 3/6] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6e5a4e757bf1..b9f15a260c78 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -436,7 +436,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 2/6] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index a5b0c40ee545..6e5a4e757bf1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -419,6 +419,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 4/8] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index c88bc6bb41cf..7a37312c8e0c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -523,7 +523,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -538,6 +546,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 3/8] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 207cdcfb6ebb..ce9f99dd3e87 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -430,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 2/8] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6b9d326e11b0..207cdcfb6ebb 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 4/9] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2bc9fa6a34b8..15bfcdbdfaa4 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -510,7 +510,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -525,6 +533,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 3/9] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 779845e3a9e3..5b6405392f08 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -430,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 2/9] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 945a79e4f3eb..779845e3a9e3 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 08/19] ASoC: SOF: Fix NULL pointer exception in sof_pci_probe callback
by Sasha Levin 10 May '22

10 May '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> [ Upstream commit c61711c1c95791850be48dd65a1d72eb34ba719f ] We are accessing "desc->ops" in sof_pci_probe without checking "desc" pointer. This results in NULL pointer exception if pci_id->driver_data i.e desc pointer isn't defined in sof device probe: BUG: kernel NULL pointer dereference, address: 0000000000000060 PGD 0 P4D 0 Oops: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:sof_pci_probe+0x1e/0x17f [snd_sof_pci] Code: Unable to access opcode bytes at RIP 0xffffffffc043dff4. RSP: 0018:ffffac4b03b9b8d8 EFLAGS: 00010246 Add NULL pointer check for sof_dev_desc pointer to avoid such exception. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220426183357.102155-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-pci-dev.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..b773289c928d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -129,6 +129,11 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) dev_dbg(&pci->dev, "PCI DSP detected"); + if (!desc) { + dev_err(dev, "error: no matching PCI descriptor\n"); + return -ENODEV; + } + if (!desc->ops) { dev_err(dev, "error: no matching PCI descriptor ops\n"); return -ENODEV; -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 06/19] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index ee3782ecd7e3..bd88e56cacac 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -519,7 +519,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -534,6 +542,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 05/19] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6d9261346842..62b41ca050a2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -430,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 04/19] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b45ec35cd63c..6d9261346842 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 09/21] ASoC: SOF: Fix NULL pointer exception in sof_pci_probe callback
by Sasha Levin 10 May '22

10 May '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> [ Upstream commit c61711c1c95791850be48dd65a1d72eb34ba719f ] We are accessing "desc->ops" in sof_pci_probe without checking "desc" pointer. This results in NULL pointer exception if pci_id->driver_data i.e desc pointer isn't defined in sof device probe: BUG: kernel NULL pointer dereference, address: 0000000000000060 PGD 0 P4D 0 Oops: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:sof_pci_probe+0x1e/0x17f [snd_sof_pci] Code: Unable to access opcode bytes at RIP 0xffffffffc043dff4. RSP: 0018:ffffac4b03b9b8d8 EFLAGS: 00010246 Add NULL pointer check for sof_dev_desc pointer to avoid such exception. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220426183357.102155-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-pci-dev.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 20c6ca37dbc4..53e97abbe6e3 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -130,6 +130,11 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) dev_dbg(&pci->dev, "PCI DSP detected"); + if (!desc) { + dev_err(dev, "error: no matching PCI descriptor\n"); + return -ENODEV; + } + if (!desc->ops) { dev_err(dev, "error: no matching PCI descriptor ops\n"); return -ENODEV; -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 07/21] ASoC: ops: Validate input values in snd_soc_put_volsw_range()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 ] Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a0ca58ba1627..884c8fd48dab 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -519,7 +519,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -534,6 +542,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 06/21] ASoC: max98090: Generate notifications on changes for custom control
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 13fcf676d9e102594effc686d98521ff5c90b925 ] The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6d9261346842..62b41ca050a2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -430,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 05/21] ASoC: max98090: Reject invalid values in custom control put()
by Sasha Levin 10 May '22

10 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2fbe467bcbfc760a08f08475eea6bbd4c2874319 ] The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b45ec35cd63c..6d9261346842 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- 2.35.1

1 0

[PATCH] ASoC: SOF: ipc: introduce cont_update_posn in sof_ipc_stream_params struct
by Pierre-Louis Bossart 10 May '22

10 May '22

From: YC Hung <yc.hung(a)mediatek.com> The host stream position is updated when no_stream_position is set as 0. However current implementation updates host stream position only when report data is larger than or equal to host period size which is decided by the period size of host side. It maybe cause host stream position update not in time. Therefore this patch introduces another field "cont_update_posn", a boolean value aimed to update host stream position continuously and based on period size of pipeline. It can get better precise when need to update host stream position from firmware. Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: YC Hung <yc.hung(a)mediatek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/sof/stream.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 58a0d49977d63..1db3bbc3e65d4 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -85,8 +85,9 @@ struct sof_ipc_stream_params { uint32_t host_period_bytes; uint16_t no_stream_position; /**< 1 means don't send stream position */ + uint8_t cont_update_posn; /**< 1 means continuous update stream position */ - uint16_t reserved[3]; + uint8_t reserved[5]; uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ } __packed; diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h index b87e46684623b..0e7dccdc25fdd 100644 --- a/include/uapi/sound/sof/abi.h +++ b/include/uapi/sound/sof/abi.h @@ -26,7 +26,7 @@ /* SOF ABI version major, minor and patch numbers */ #define SOF_ABI_MAJOR 3 -#define SOF_ABI_MINOR 20 +#define SOF_ABI_MINOR 21 #define SOF_ABI_PATCH 0 /* SOF ABI version number. Format within 32bit word is MMmmmppp */ -- 2.30.2

2 1

[PATCH 1/2] ASoC: fsl_micfil: Add support for i.MX8MPlus
by Shengjiu Wang 10 May '22

10 May '22

On i.MX8Plus there are two updates for micfil module. One is that the output format is S32_LE, only the 24 more significative bits have information, the other bits are always zero. Add 'formats' variable in soc data to distinguish the format on different platform. Another is that the fifo depth is 32 entries. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index e4d1da55293e..25c647ae080a 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -55,6 +55,7 @@ struct fsl_micfil_soc_data { unsigned int fifo_depth; unsigned int dataline; bool imx; + u64 formats; }; static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { @@ -62,10 +63,20 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { .fifos = 8, .fifo_depth = 8, .dataline = 0xf, + .formats = SNDRV_PCM_FMTBIT_S16_LE, +}; + +static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { + .imx = true, + .fifos = 8, + .fifo_depth = 32, + .dataline = 0xf, + .formats = SNDRV_PCM_FMTBIT_S32_LE, }; static const struct of_device_id fsl_micfil_dt_ids[] = { { .compatible = "fsl,imx8mm-micfil", .data = &fsl_micfil_imx8mm }, + { .compatible = "fsl,imx8mp-micfil", .data = &fsl_micfil_imx8mp }, {} }; MODULE_DEVICE_TABLE(of, fsl_micfil_dt_ids); @@ -659,6 +670,8 @@ static int fsl_micfil_probe(struct platform_device *pdev) return ret; } + fsl_micfil_dai.capture.formats = micfil->soc->formats; + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component, &fsl_micfil_dai, 1); if (ret) { -- 2.17.1

2 3

[PATCH -next] ASoC: samsung: spdif: remove unnecessary check of mem_res
by Yang Yingliang 10 May '22

10 May '22

The resource is checked in probe function, so there is no need do this check in remove function. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/samsung/spdif.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 226c359892e9..47b6d19e43ff 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -467,8 +467,7 @@ static int spdif_remove(struct platform_device *pdev) iounmap(spdif->regs); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); + release_mem_region(mem_res->start, resource_size(mem_res)); clk_disable_unprepare(spdif->sclk); clk_disable_unprepare(spdif->pclk); -- 2.25.1

2 1

[RFC PATCH 0/2] Make qcom-ngd-ctrl not wait indefinitely for already booted ADSP
by Yassine Oudjana 10 May '22

10 May '22

Commit a899d324863a3 ("slimbus: qcom-ngd-ctrl: add Sub System Restart support") made qcom-ngd-ctrl wait for ADSP to become ready before starting to do its work. Due to how the SSR notifications currently work though, if qcom-ngd-ctrl probes after ADSP boots and becomes ready, it never receives a QCOM_SSR_AFTER_POWERUP event notification and keeps waiting indefinitely, making SLIMbus never come up. This series makes qcom_register_ssr_notifier call the notifier_call of the newly registered notifier block with the last SSR event received from the remoteproc, basically reporting the event that qcom-ngd-ctrl missed by registering late, stopping it from waiting for an event that has already happened. I'm not sure if this approach would have any unwanted consequences in other drivers relying on SSR events however, hence I'm sending this as a RFC. This can also be considered a bug report, so if anyone has a better fix then I'd appreciate it getting applied instead of this one. The second patch is a general fix that became necessary after the first patch, and should likely be applied anyway. Yassine Oudjana (2): remoteproc: qcom: Report last event on SSR notifier registration slimbus: qcom-ngd-ctrl: Initialize ngd_up_work before it can be scheduled drivers/remoteproc/qcom_common.c | 17 +++++++++++++---- drivers/slimbus/qcom-ngd-ctrl.c | 3 ++- 2 files changed, 15 insertions(+), 5 deletions(-) -- 2.36.0

1 2

[PATCH v2] ASoC: tlv320adcx140: Register a callback to disable the regulator_disable
by Zheyu Ma 10 May '22

10 May '22

The driver should register a callback that will deal with the disabling when it fails to probe. Signed-off-by: Zheyu Ma <zheyuma97(a)gmail.com> --- Changes in v2: - Generate the patch based on the latest source tree --- sound/soc/codecs/tlv320adcx140.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 0b06fbb14171..b55f0b836932 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -1083,6 +1083,13 @@ static const struct of_device_id tlv320adcx140_of_match[] = { MODULE_DEVICE_TABLE(of, tlv320adcx140_of_match); #endif +static void adcx140_disable_regulator(void *arg) +{ + struct adcx140_priv *adcx140 = arg; + + regulator_disable(adcx140->supply_areg); +} + static int adcx140_i2c_probe(struct i2c_client *i2c) { struct adcx140_priv *adcx140; @@ -1112,6 +1119,10 @@ static int adcx140_i2c_probe(struct i2c_client *i2c) dev_err(adcx140->dev, "Failed to enable areg\n"); return ret; } + + ret = devm_add_action_or_reset(&i2c->dev, adcx140_disable_regulator, adcx140); + if (ret) + return ret; } adcx140->regmap = devm_regmap_init_i2c(i2c, &adcx140_i2c_regmap); -- 2.25.1

2 1

[PATCH] ASoC: codecs: rt715-sdca: remove useless assignment of ops
by Pierre-Louis Bossart 10 May '22

10 May '22

The ops are already part of the 'struct sdw_driver', it's unclear why this was copied into the 'slave' structure - no other driver does so. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt715-sdca-sdw.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index a5c673f43d824..0ecd2948f7aa7 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -181,8 +181,6 @@ static int rt715_sdca_sdw_probe(struct sdw_slave *slave, { struct regmap *mbq_regmap, *regmap; - slave->ops = &rt715_sdca_slave_ops; - /* Regmap Initialization */ mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt715_sdca_mbq_regmap); if (IS_ERR(mbq_regmap)) -- 2.30.2

2 1

[PATCH] ASoC: tlv320adcx140: Register a callback to disable the regulator_disable
by Zheyu Ma 10 May '22

10 May '22

The driver should register a callback that will deal with the disabling when it fails to probe. Signed-off-by: Zheyu Ma <zheyuma97(a)gmail.com> --- sound/soc/codecs/tlv320adcx140.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 32b120d624b2..f70ea59259f1 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -1083,6 +1083,13 @@ static const struct of_device_id tlv320adcx140_of_match[] = { MODULE_DEVICE_TABLE(of, tlv320adcx140_of_match); #endif +static void adcx140_disable_regulator(void *arg) +{ + struct adcx140_priv *adcx140 = arg; + + regulator_disable(adcx140->supply_areg); +} + static int adcx140_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1113,6 +1120,10 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, dev_err(adcx140->dev, "Failed to enable areg\n"); return ret; } + + ret = devm_add_action_or_reset(&i2c->dev, adcx140_disable_regulator, adcx140); + if (ret) + return ret; } adcx140->regmap = devm_regmap_init_i2c(i2c, &adcx140_i2c_regmap); -- 2.25.1

2 2

[PATCH 1/2] ASoC: wm8940: add devicetree support
by Lukasz Majewski 10 May '22

10 May '22

This adds devicetree support to the wm8940 codec driver. With a DT-based kernel, there is no board-specific setting to select the driver so allow it to be manually chosen. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8940.c | 7 +++++++ 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f46a22660103..5120b15139bc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1811,7 +1811,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate "Wolfson Microelectronics WM8940 codec" depends on I2C config SND_SOC_WM8955 diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 440d048ef0c0..8615a55f84dd 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -779,9 +779,16 @@ static const struct i2c_device_id wm8940_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); +static const struct of_device_id wm8940_of_match[] = { + { .compatible = "wlf,wm8940", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8940_of_match); + static struct i2c_driver wm8940_i2c_driver = { .driver = { .name = "wm8940", + .of_match_table = wm8940_of_match, }, .probe = wm8940_i2c_probe, .id_table = wm8940_i2c_id, -- 2.35.1

3 6

[PATCH 00/38] Clean up usage of the endianness flag
by Charles Keepax 10 May '22

10 May '22

Before componentisation any part registered as a CODEC would have automatically supported both little and big endian, ie. the core would duplicate any supported LE or BE PCM format to support the other endian as well. As componentisation removed the distinction between CODEC drivers and platform drivers, a flag was added to specify if this behaviour is required for a particular component. However, as most systems tend to use little endian the absence of the flag is rarely noticed. Also the naming of the flag "endianness" is a little unobvious as to if it should be applied to a particular component. This series adds a comment to better explain the meaning of the flag and then tidys up the usage of the flag. A couple of uses of the flag are removed where is has been used inappropriately on the CPU side of the DAI link, this is clearly not valid in the cases it has been used, and I suspect never would be valid. Then some redundant formats are removed, since they would be covered by existing endianness flags. And finally a bunch of devices that are missing the flag have it added. It is worth noting that since componenisation there are now a couple of cases where it is not entire clear to me that the flag should be applied to all CODECs as it was before. In those cases I haven't updated the driver to add the flag and they are outlined here: 1) Build into the AP CODECs, these are actual silicon inside the main processor and they typically receive audio directly from an internal bus. It is not obvious to me that these can happily ignore endian. On the CODEC side these include: jz4725b.c, jz4760.c, jz4770.c, rk3328_codec.c, lpass-va-macro.c, lpass-rx-macro.c, lpass-tx-macro.c, lpass-wsa-macro.c. There are also some examples of this scattered around the various platform support directories in sound/soc. 2) Devices behind non-audio buses, SPI just moves bits and doesn't really define an endian for audio data on the bus. Thus it seems the CODEC probably can care about the endian. The only devices that fall into this group (mostly for AoV) are: rt5514-spi.c, rt5677-spi.c, cros_ec_codec.c (only the AoV). 3) CODECs with no DAIs, these could specify the flag and plenty of them do; CODECs from the initial conversion to componentisation. But the flag makes no difference here since there is nothing for it to apply to. This includes purely analogue CODECs: aw8738.c, ssm2305.c, tpa6130a2.c, tda7419.c, max9759.c, max9768.c, max9877.c, lm4857.c, simple-mux.c, simple-amplifier.c. And devices that only do jack detection: ts3a227e.c, mt6359-accdet.c. If there are any opinions on adding the flag to any of those three groups they would be greatfully received. But I am leaning towards leaving 1,2 without endianness flags since it feels inappropriate, and removing the endian flag from devices in catagory 3 that already have it. Assuming no one objects to that I will do a follow up series for that. Thanks, Charles Charles Keepax (38): ASoC: soc-component: Add comment for the endianness flag ASoC: atmel-pdmic: Remove endianness flag on pdmic component ASoC: atmel-classd: Remove endianness flag on class d component ASoC: cs4270: Remove redundant big endian formats ASoC: cs42l51: Remove redundant big endian formats ASoC: cs4349: Remove redundant big endian formats ASoC: hdmi-codec: Remove redundant big endian formats ASoC: sta32x: Remove redundant big endian formats ASoC: sta350: Remove redundant big endian formats ASoC: hdac_hda: Add endianness flag in snd_soc_component_driver ASoC: max98504: Add endianness flag in snd_soc_component_driver ASoC: adau1372: Add endianness flag in snd_soc_component_driver ASoC: cs4234: Add endianness flag in snd_soc_component_driver ASoC: cs35l41: Add endianness flag in snd_soc_component_driver ASoC: cx2072x: Add endianness flag in snd_soc_component_driver ASoC: lochnagar: Add endianness flag in snd_soc_component_driver ASoC: mt6351: Add endianness flag in snd_soc_component_driver ASoC: mt6358: Add endianness flag in snd_soc_component_driver ASoC: mt6359: Add endianness flag in snd_soc_component_driver ASoC: mt6660: Add endianness flag in snd_soc_component_driver ASoC: pcm3060: Add endianness flag in snd_soc_component_driver ASoC: rt1019: Add endianness flag in snd_soc_component_driver ASoC: rt9120: Add endianness flag in snd_soc_component_driver ASoC: tlv320adc3xxx: Add endianness flag in snd_soc_component_driver ASoC: tscs454: Add endianness flag in snd_soc_component_driver ASoC: cros_ec_codec: Add endianness flag in i2s_rx_component_driver ASoC: wcd934x: Add endianness flag in snd_soc_component_driver ASoC: wcd9335: Add endianness flag in snd_soc_component_driver ASoC: rt700: Add endianness flag in snd_soc_component_driver ASoC: rt711: Add endianness flag in snd_soc_component_driver ASoC: rt711-sdca: Add endianness flag in snd_soc_component_driver ASoC: rt715: Add endianness flag in snd_soc_component_driver ASoC: rt715-sdca: Add endianness flag in snd_soc_component_driver ASoC: rt1308-sdw: Add endianness flag in snd_soc_component_driver ASoC: rt1316-sdw: Add endianness flag in snd_soc_component_driver ASoC: wcd938x: Add endianness flag in snd_soc_component_driver ASoC: wsa881x: Add endianness flag in snd_soc_component_driver ASoC: sdw-mockup: Add endianness flag in snd_soc_component_driver include/sound/soc-component.h | 9 +++++++++ sound/soc/atmel/atmel-classd.c | 1 - sound/soc/atmel/atmel-pdmic.c | 1 - sound/soc/codecs/adau1372.c | 1 + sound/soc/codecs/cros_ec_codec.c | 7 ++++--- sound/soc/codecs/cs35l41.c | 2 ++ sound/soc/codecs/cs4234.c | 1 + sound/soc/codecs/cs4270.c | 15 +++------------ sound/soc/codecs/cs42l51.c | 7 ++----- sound/soc/codecs/cs4349.c | 9 +++------ sound/soc/codecs/cx2072x.c | 1 + sound/soc/codecs/hdac_hda.c | 15 ++++++++------- sound/soc/codecs/hdmi-codec.c | 15 +++++---------- sound/soc/codecs/lochnagar-sc.c | 5 +++-- sound/soc/codecs/max98504.c | 1 + sound/soc/codecs/mt6351.c | 10 ++++------ sound/soc/codecs/mt6358.c | 10 ++++------ sound/soc/codecs/mt6359.c | 10 ++++------ sound/soc/codecs/mt6660.c | 1 + sound/soc/codecs/pcm3060.c | 1 + sound/soc/codecs/rt1019.c | 3 ++- sound/soc/codecs/rt1308-sdw.c | 1 + sound/soc/codecs/rt1316-sdw.c | 1 + sound/soc/codecs/rt700.c | 1 + sound/soc/codecs/rt711-sdca.c | 1 + sound/soc/codecs/rt711.c | 1 + sound/soc/codecs/rt715-sdca.c | 1 + sound/soc/codecs/rt715.c | 1 + sound/soc/codecs/rt9120.c | 1 + sound/soc/codecs/sdw-mockup.c | 1 + sound/soc/codecs/sta32x.c | 9 +++------ sound/soc/codecs/sta350.c | 9 +++------ sound/soc/codecs/tlv320adc3xxx.c | 11 ++++++----- sound/soc/codecs/tscs454.c | 12 ++++++------ sound/soc/codecs/wcd9335.c | 1 + sound/soc/codecs/wcd934x.c | 1 + sound/soc/codecs/wcd938x.c | 1 + sound/soc/codecs/wsa881x.c | 1 + 38 files changed, 90 insertions(+), 89 deletions(-) -- 2.30.2

3 46

[PATCH 00/14] ASoC: Intel: avs: Machine boards and HDA codec support
by Cezary Rojewski 10 May '22

10 May '22

This series focuses on populating boards/ subdirectory with supported configurations by the avs-driver. Note: it is independent of recently provided "Driver code and PCM operations" series [1], that is, code found here should not collide with it. Series starts with a small change that adds a helper to sound pcm header, allowing for retrieving string naming a direction without the need of substream pointer. Said helper is used by codec driver code that follows it but I believe it's generic and helpful enough that it can be called an independent addition to the sound core. Code for generic HD-Audio codec driver follows. It is a ASoC wrapper for existing HD-Audio codec code found in sound/pci/hda/. There is basically no custom logic involved up to the point that driver follows HDA_DEV_LEGACY convention, rather than the HDA_DEV_ASOC one. Commit message for the given patch iterates on this and explains crucial parts of the implementation. >From there on is a range of boards appended. All of them follow the same scheme: - define avs_create_dai_link() so DAI-LINKs can be created dynamically, based on the link_mask (I2S) or the number of entries in the ->pcm_list_head list (HDA) - define avs_create_dapm_routes() so DAPM routes can be created dynamically, same rules as above apply - define probe() function that creates new ASoC card, assign all required operations and resources along with calling the two above [1]: https://lore.kernel.org/all/20220426172346.3508411-1-cezary.rojewski@intel.… Amadeusz Sławiński (1): ASoC: Intel: avs: Add max98373 machine board Cezary Rojewski (13): ALSA: Add snd_pcm_direction_name() helper ASoC: codecs: Add HD-Audio codec driver ASoC: Intel: avs: Add HDAudio machine board ASoC: Intel: avs: Add DMIC machine board ASoC: Intel: avs: Add ssp-test machine board ASoC: Intel: avs: Add rt274 machine board ASoC: Intel: avs: Add rt286 machine board ASoC: Intel: avs: Add rt298 machine board ASoC: Intel: avs: Add rt5682 machine board ASoC: Intel: avs: Add nau8825 machine board ASoC: Intel: avs: Add ssm4567 machine board ASoC: Intel: avs: Add max98357a machine board ASoC: Intel: avs: Add da7219 machine board include/sound/pcm.h | 19 +- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hda-dai.c | 102 +++++++ sound/soc/codecs/hda.c | 395 +++++++++++++++++++++++++ sound/soc/codecs/hda.h | 19 ++ sound/soc/intel/Kconfig | 3 + sound/soc/intel/avs/Makefile | 3 + sound/soc/intel/avs/boards/Kconfig | 121 ++++++++ sound/soc/intel/avs/boards/Makefile | 27 ++ sound/soc/intel/avs/boards/da7219.c | 281 ++++++++++++++++++ sound/soc/intel/avs/boards/dmic.c | 93 ++++++ sound/soc/intel/avs/boards/hdaudio.c | 294 ++++++++++++++++++ sound/soc/intel/avs/boards/max98357a.c | 153 ++++++++++ sound/soc/intel/avs/boards/max98373.c | 238 +++++++++++++++ sound/soc/intel/avs/boards/nau8825.c | 352 ++++++++++++++++++++++ sound/soc/intel/avs/boards/rt274.c | 309 +++++++++++++++++++ sound/soc/intel/avs/boards/rt286.c | 280 ++++++++++++++++++ sound/soc/intel/avs/boards/rt298.c | 280 ++++++++++++++++++ sound/soc/intel/avs/boards/rt5682.c | 339 +++++++++++++++++++++ sound/soc/intel/avs/boards/ssm4567.c | 270 +++++++++++++++++ sound/soc/intel/avs/boards/ssp_test.c | 178 +++++++++++ 22 files changed, 3759 insertions(+), 4 deletions(-) create mode 100644 sound/soc/codecs/hda-dai.c create mode 100644 sound/soc/codecs/hda.c create mode 100644 sound/soc/codecs/hda.h create mode 100644 sound/soc/intel/avs/boards/Kconfig create mode 100644 sound/soc/intel/avs/boards/Makefile create mode 100644 sound/soc/intel/avs/boards/da7219.c create mode 100644 sound/soc/intel/avs/boards/dmic.c create mode 100644 sound/soc/intel/avs/boards/hdaudio.c create mode 100644 sound/soc/intel/avs/boards/max98357a.c create mode 100644 sound/soc/intel/avs/boards/max98373.c create mode 100644 sound/soc/intel/avs/boards/nau8825.c create mode 100644 sound/soc/intel/avs/boards/rt274.c create mode 100644 sound/soc/intel/avs/boards/rt286.c create mode 100644 sound/soc/intel/avs/boards/rt298.c create mode 100644 sound/soc/intel/avs/boards/rt5682.c create mode 100644 sound/soc/intel/avs/boards/ssm4567.c create mode 100644 sound/soc/intel/avs/boards/ssp_test.c base-commit: fa96ec9fa9f8f0c80e8a0129a819f592a39b1044 -- 2.25.1

4 24

[PATCH] ALSA: wavefront: Proper check of get_user() error
by Takashi Iwai 10 May '22

10 May '22

The antient ISA wavefront driver reads its sample patch data (uploaded over an ioctl) via __get_user() with no good reason; likely just for some performance optimizations in the past. Let's change this to the standard get_user() and the error check for handling the fault case properly. Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/wavefront/wavefront_synth.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69cbc79fbb71..2aaaa6807174 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1094,7 +1094,8 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dataptr < data_end) { - __get_user (sample_short, dataptr); + if (get_user(sample_short, dataptr)) + return -EFAULT; dataptr += skip; if (data_is_unsigned) { /* GUS ? */ -- 2.31.1

1 0

Parallel build failure in rate/ (probably rate-lav/, pph/ too)
by GitHub issues - edited 10 May '22

10 May '22

alsa-project/alsa-plugins issue #44 was edited from thesamesam: Originally reported downstream in [Gentoo](https://bugs.gentoo.org/835920) by a user. It looks like in `rate/Makefile.am` (and I think ditto for `rate-lav/Makefile.am`, `pph/Makefile.am`), with high parallelism (`make -jN install`, for large N), `$(DESTDIR)@ALSA_PLUGIN_DIR@` may not have been created yet. All of these dirs override `install-exec-hook`. For other subdirs, like `jack/Makefile.am`, there's no problem, because the implementation of `install-exec-hook` for those is set by `install-hooks.am` which handles `mkdir -p` before symlinking/installing. Example of failure: ``` >>> Install media-plugins/alsa-plugins-1.2.2 into /var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image * abi_x86_64.amd64: running multilib-minimal_abi_src_install make -j2 DESTDIR=/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image install Making install in doc make[1]: Entering directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/doc' make[2]: Entering directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/doc' make[2]: Nothing to be done for 'install-exec-am'. make[2]: Nothing to be done for 'install-data-am'. make[2]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/doc' make[1]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/doc' Making install in rate make[1]: Entering directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/rate' make[2]: Entering directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/rate' make install-exec-hook /bin/mkdir -p '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/share/alsa/alsa.conf.d' make[3]: Entering directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/rate' rm -f /var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib/libasound_module_rate_samplerate_*.so ln -s libasound_module_rate_samplerate.so /var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib/libasound_module_rate_samplerate_best.so /usr/bin/install -c -m 644 /var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2/rate/10-samplerate.conf '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/share/alsa/alsa.conf.d' ln: failed to create symbolic link '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib/libasound_module_rate_samplerate_best.so': No such file or directory make[3]: *** [Makefile:738: install-exec-hook] Error 1 make[3]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/rate' make[2]: *** [Makefile:651: install-exec-am] Error 2 make[2]: *** Waiting for unfinished jobs.... /bin/mkdir -p '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib' /bin/sh ../libtool --mode=install /usr/bin/install -c libasound_module_rate_samplerate.la '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib' libtool: install: /usr/bin/install -c .libs/libasound_module_rate_samplerate.so /var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib/libasound_module_rate_samplerate.so libtool: install: /usr/bin/install -c .libs/libasound_module_rate_samplerate.lai /var/tmp/portage/media-plugins/alsa-plugins-1.2.2/image/usr/lib64/alsa-lib/libasound_module_rate_samplerate.la libtool: warning: remember to run 'libtool --finish /usr/lib64/alsa-lib' make[2]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/rate' make[1]: *** [Makefile:594: install-am] Error 2 make[1]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.2/work/alsa-plugins-1.2.2-abi_x86_64.amd64/rate' make: *** [Makefile:444: install-recursive] Error 1 * ERROR: media-plugins/alsa-plugins-1.2.2::gentoo failed (install phase): * emake failed ``` Reported to happen with 1.2.6 too, which makes sense, as nothing changed there. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/44 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

Parallel build failure in rate/ (probably rate-lav/, pph/ too)
by GitHub issues - opened 10 May '22

10 May '22

alsa-project/alsa-plugins issue #44 was opened from thesamesam: Originally reported downstream in [Gentoo](https://bugs.gentoo.org/835920) by a user. It looks like in `rate/Makefile.am` (and I think ditto for `rate-lav/Makefile.am`, `pph/Makefile.am`), with high parallelism (`make -jN install`, for large N), `$(DESTDIR)@ALSA_PLUGIN_DIR@` may not have been created yet. All of these dirs override `install-exec-hook`. For other subdirs, like `jack/Makefile.am`, there's no problem, because the implementation of `install-exec-hook` for those is set by `install-hooks.am` which handles `mkdir -p` before symlinking/installing. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/44 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/8] ASoC: SOF: sof-client: Update for different IPC versions
by Peter Ujfalusi 10 May '22

10 May '22

Hi, The current IPC client infrastructure can only be used with IPC3. This series carries updates for the core side of the client support to handle IPC4 messages and updates the ipc message injector to be usable with IPC4. The IPC flood test is only supported by SOF_IPC (IPC3), we are not going to create the aux device for it at all if the firmware is using IPC4. Regards, Peter --- Peter Ujfalusi (8): ASoC: SOF: sof-client: Add API to get the maximum IPC payload size ASoC: SOF: ipc-msg-injector: Query the maximum IPC payload size ASoC: SOF: sof-client-probes: Query the maximum IPC payload size ASoC: SOF: sof-client: Add API to get the ipc_type ASoC: SOF: sof-client: Add support IPC4 message sending ASoC: SOF: ipc-msg-injector: Separate the message sending ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages ASoC: SOF: sof-client: IPC flood test can only work with SOF_IPC sound/soc/sof/sof-client-ipc-msg-injector.c | 181 ++++++++++++++++++-- sound/soc/sof/sof-client-probes.c | 5 +- sound/soc/sof/sof-client.c | 66 ++++++- sound/soc/sof/sof-client.h | 2 + 4 files changed, 227 insertions(+), 27 deletions(-) -- 2.36.0

2 9

[PATCH] Revert "ASoC: dt-bindings: mediatek: mt8192: Add i2s-share properties"
by Nícolas F. R. A. Prado 09 May '22

09 May '22

This reverts commit e056cf4341ae3f856f1e38da02b27cb04de4c69b. The commit was merged while the property name and definition were still being discussed. Revert the commit for now and a follow up commit will re-add the property after it is further discussed and reviewed. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml | 5 ----- 1 file changed, 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml index 5b03c8dbf318..7a25bc9b8060 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -54,11 +54,6 @@ properties: - const: aud_infra_clk - const: aud_infra_26m_clk -patternProperties: - "^i2s[0-35-9]-share$": - description: Name of the I2S bus that is shared with this bus - pattern: "^I2S[0-35-9]$" - required: - compatible - interrupts -- 2.36.0

2 2

[PATCH] ASoC: sam9g20_wm8731: Use dev_err_probe() for snd_soc_register_card()
by Mark Brown 09 May '22

09 May '22

Log the error code when snd_soc_regster_card() fails, but fold in the silencing of deferred probe errors. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/sam9g20_wm8731.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 0365b583ba70..4d25fb61c652 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -171,7 +171,8 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed\n"); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card() failed: %d\n", ret); goto err; } -- 2.30.2

2 2

[PATCH] ASoC: SOF: trace: The dtrace is only available with SOF_IPC
by Peter Ujfalusi 09 May '22

09 May '22

Currently the dtrace only supported with SOF_IPC. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Hi, I have pending series to rework the whole tracing support to allow different tracing solutions to be plugged in for different IPC versions. It is going to be sent at later time, until then we need to make sure that the current dtrace is not used with IPC4. Regards, Peter sound/soc/sof/trace.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index ba6361b5d64b..5d171bf8a5ea 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -457,6 +457,10 @@ int snd_sof_init_trace(struct snd_sof_dev *sdev) { int ret; + /* dtrace is only supported with SOF_IPC */ + if (sdev->pdata->ipc_type != SOF_IPC) + sdev->dtrace_is_supported = false; + if (!sdev->dtrace_is_supported) return 0; -- 2.36.0

2 1

[PATCH 0/3] Add missing dt-binding properties for audio components on mt8192-asurada
by Nícolas F. R. A. Prado 09 May '22

09 May '22

These patches add properties that were missing on the dt-bindings of the audio components used by mt8192-asurada. Namely the i2s-share properties for the sound platform and the #sound-dai-cells on the rt1015p and rt5682 codecs when they're referenced by the machine sound node. Nícolas F. R. A. Prado (3): ASoC: dt-bindings: mediatek: mt8192: Add i2s-share properties ASoC: dt-bindings: rt1015p: Add #sound-dai-cells ASoC: dt-bindings: rt5682: Add #sound-dai-cells Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml | 5 +++++ Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 3 +++ Documentation/devicetree/bindings/sound/rt5682.txt | 2 ++ 3 files changed, 10 insertions(+) -- 2.36.0

5 17

[PATCH 2/2] doc: dts: Add device tree information regarding wm8940 codec
by Lukasz Majewski 09 May '22

09 May '22

This commit provides documentation entry for wm8940 codec. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> --- .../devicetree/bindings/sound/wlf,wm8940.yaml | 57 +++++++++++++++++++ 1 file changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8940.yaml diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml new file mode 100644 index 000000000000..8aadcbeed502 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8940.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8940.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8940 Codec Device Tree Bindings + +maintainers: + - patches(a)opensource.cirrus.com + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8940 + + 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,wm8940"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "wlf,wm8940"; + reg = <0x1a>; + }; + }; + +... -- 2.35.1

3 2

[PATCH v3 -next] ASoC: mediatek: mt8195: Fix build warning without CONFIG_OF
by YueHaibing 09 May '22

09 May '22

sound/soc/mediatek/mt8195/mt8195-mt6359.c:1639:32: warning: ‘mt8195_mt6359_max98390_rt5682_card’ defined but not used [-Wunused-variable] 1639 | static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/mt8195/mt8195-mt6359.c:1634:32: warning: ‘mt8195_mt6359_rt1011_rt5682_card’ defined but not used [-Wunused-variable] 1634 | static struct mt8195_card_data mt8195_mt6359_rt1011_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/mt8195/mt8195-mt6359.c:1629:32: warning: ‘mt8195_mt6359_rt1019_rt5682_card’ defined but not used [-Wunused-variable] 1629 | static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Since all users of this driver do need CONFIG_OF anyway, there is no need to save a few bytes on kernel builds while CONFIG_OF disabled, so just remove the #ifdef to fix this warning. Fixes: 86a6b9c9dfff ("ASoC: mediatek: mt8195: add machine support for max98390 and rt5682") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Arnd Bergmann <arnd(a)arndb.de> --- v3: rebase on next-20220509 v2: remove #ifdef block as Arnd suggested --- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 3e32fe801b3c..54a00b0699b1 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -1641,7 +1641,6 @@ static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = { .quirk = MAX98390_SPEAKER_AMP_PRESENT, }; -#ifdef CONFIG_OF static const struct of_device_id mt8195_mt6359_dt_match[] = { { .compatible = "mediatek,mt8195_mt6359_rt1019_rt5682", @@ -1657,7 +1656,6 @@ static const struct of_device_id mt8195_mt6359_dt_match[] = { }, {}, }; -#endif static const struct dev_pm_ops mt8195_mt6359_pm_ops = { .poweroff = snd_soc_poweroff, @@ -1667,9 +1665,7 @@ static const struct dev_pm_ops mt8195_mt6359_pm_ops = { static struct platform_driver mt8195_mt6359_driver = { .driver = { .name = "mt8195_mt6359", -#ifdef CONFIG_OF .of_match_table = mt8195_mt6359_dt_match, -#endif .pm = &mt8195_mt6359_pm_ops, }, .probe = mt8195_mt6359_dev_probe, -- 2.17.1

2 1

[PATCH v2 -next] ASoC: mediatek: mt8195: Fix build warning without CONFIG_OF
by YueHaibing 09 May '22

09 May '22

sound/soc/mediatek/mt8195/mt8195-mt6359.c:1639:32: warning: ‘mt8195_mt6359_max98390_rt5682_card’ defined but not used [-Wunused-variable] 1639 | static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/mt8195/mt8195-mt6359.c:1634:32: warning: ‘mt8195_mt6359_rt1011_rt5682_card’ defined but not used [-Wunused-variable] 1634 | static struct mt8195_card_data mt8195_mt6359_rt1011_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/mt8195/mt8195-mt6359.c:1629:32: warning: ‘mt8195_mt6359_rt1019_rt5682_card’ defined but not used [-Wunused-variable] 1629 | static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Since all users of this driver do need CONFIG_OF anyway, there is no need to save a few bytes on kernel builds while CONFIG_OF disabled, so just remove the #ifdef to fix this warning. Fixes: 86a6b9c9dfff ("ASoC: mediatek: mt8195: add machine support for max98390 and rt5682") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- v2: remove #ifdef block as Arnd suggested --- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index f90675f14d60..54a00b0699b1 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -1626,7 +1626,6 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) return ret; } -#ifdef CONFIG_OF static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = { .name = "mt8195_r1019_5682", .quirk = RT1019_SPEAKER_AMP_PRESENT, @@ -1657,7 +1656,6 @@ static const struct of_device_id mt8195_mt6359_dt_match[] = { }, {}, }; -#endif static const struct dev_pm_ops mt8195_mt6359_pm_ops = { .poweroff = snd_soc_poweroff, @@ -1667,9 +1665,7 @@ static const struct dev_pm_ops mt8195_mt6359_pm_ops = { static struct platform_driver mt8195_mt6359_driver = { .driver = { .name = "mt8195_mt6359", -#ifdef CONFIG_OF .of_match_table = mt8195_mt6359_dt_match, -#endif .pm = &mt8195_mt6359_pm_ops, }, .probe = mt8195_mt6359_dev_probe, -- 2.17.1

3 4

Re: [Intel-gfx] [PATCH v5 1/2] module: update dependencies at try_module_get()
by Mauro Carvalho Chehab 09 May '22

09 May '22

Em Thu, 5 May 2022 23:35:29 +0200 Andi Shyti <andi.shyti(a)linux.intel.com> escreveu: > Hi Mauro, > > [...] > > > +static int ref_module_dependency(struct module *mod, struct module *this) > > +{ > > + int ret; > > + > > + if (!this || !this->name) > > + return -EINVAL; > > + > > + if (mod == this) > > + return 0; > > + > > + mutex_lock(&module_mutex); > > + > > + ret = ref_module(this, mod); > > + > > +#ifdef CONFIG_MODULE_UNLOAD > > + if (ret) > > + goto ret; > > + > > + ret = sysfs_create_link(mod->holders_dir, > > + &this->mkobj.kobj, this->name); > > +#endif > > + > > +ret: > > + mutex_unlock(&module_mutex); > > + return ret; > > +} > > + > > /* Clear the unload stuff of the module. */ > > static void module_unload_free(struct module *mod) > > { > > @@ -841,24 +886,16 @@ void __module_get(struct module *module) > > } > > EXPORT_SYMBOL(__module_get); > > > > -bool try_module_get(struct module *module) > > +bool try_module_get_owner(struct module *module, struct module *this) > > { > > - bool ret = true; > > + int ret = __try_module_get(module); > > > > - if (module) { > > - preempt_disable(); > > - /* Note: here, we can fail to get a reference */ > > - if (likely(module_is_live(module) && > > - atomic_inc_not_zero(&module->refcnt) != 0)) > > - trace_module_get(module, _RET_IP_); > > - else > > - ret = false; > > + if (ret) > > + ref_module_dependency(module, this); > > do we care about the return value here? I don't think it should care about the return value, as a failure to create a sysfs node for the holder or to add it to the holders list is not fatal: modules can still continue working without that. Also, I opted to be conservative here: currently, not creating these doesn't cause try_module_get() to fail. I'm not sure what would be the impact if this starts to fail. So, right now, I'm opting to just ignore the return value. Perhaps in the future this could a warning (similarly to what sysfs create link does). Regards, Mauro > > Andi > > > > > - preempt_enable(); > > - } > > return ret; > > } > > -EXPORT_SYMBOL(try_module_get); > > +EXPORT_SYMBOL(try_module_get_owner); > > > > void module_put(struct module *module) > > { > > -- > > 2.35.1 Thanks, Mauro

1 0

[PATCH v1] ALSA: hda/realtek: Add quirk for the Framework Laptop
by Dustin L. Howett 09 May '22

09 May '22

Some board revisions of the Framework Laptop have an ALC295 and have an issue detecting headset microphones. The "dell-headset-multi" fixup addresses this issue, but its application requires an end-user or distributor to opt in. f111:0001 is Framework's PCI VID:PID for this board. Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- 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 4c0c593f3c0a..a1b45c72e9b1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9294,6 +9294,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), + SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. -- 2.36.0

2 1

[PATCH] ASoC: pcm186x: simplify the return expression of pcm186x_power_off()
by cgel.zte＠gmail.com 09 May '22

09 May '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Simplify the return expression. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/codecs/pcm186x.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 2c78dccb3f62..fda9d7ee3fe6 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -534,19 +534,14 @@ static int pcm186x_power_on(struct snd_soc_component *component) static int pcm186x_power_off(struct snd_soc_component *component) { struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component); - int ret; snd_soc_component_update_bits(component, PCM186X_POWER_CTRL, PCM186X_PWR_CTRL_PWRDN, PCM186X_PWR_CTRL_PWRDN); regcache_cache_only(priv->regmap, true); - ret = regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + return regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); - if (ret) - return ret; - - return 0; } static int pcm186x_set_bias_level(struct snd_soc_component *component, -- 2.25.1

2 1

[PATCH] ASoC: mediatek: simplify the return expression of mtk_dai_pcm_prepare()
by cgel.zte＠gmail.com 09 May '22

09 May '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Simplify the return expression. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c index 151914c873ac..12644ded83d5 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c @@ -213,8 +213,6 @@ static int mtk_dai_pcm_configure(struct snd_pcm_substream *substream, static int mtk_dai_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - int ret; - dev_dbg(dai->dev, "%s(), id %d, stream %d, widget active p %d, c %d\n", __func__, dai->id, substream->stream, dai->playback_widget->active, dai->capture_widget->active); @@ -222,11 +220,7 @@ static int mtk_dai_pcm_prepare(struct snd_pcm_substream *substream, if (dai->playback_widget->active || dai->capture_widget->active) return 0; - ret = mtk_dai_pcm_configure(substream, dai); - if (ret) - return ret; - - return 0; + return mtk_dai_pcm_configure(substream, dai); } static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) -- 2.25.1

2 1

[PATCH] ASoC: uniphier: simplify the return expression of uniphier_aio_compr_set_params()
by cgel.zte＠gmail.com 09 May '22

09 May '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Simplify the return expression. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/uniphier/aio-compress.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c index 0f76bc601ca9..7d1492c15b57 100644 --- a/sound/soc/uniphier/aio-compress.c +++ b/sound/soc/uniphier/aio-compress.c @@ -139,7 +139,6 @@ static int uniphier_aio_compr_set_params(struct snd_soc_component *component, struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; struct device *dev = &aio->chip->pdev->dev; - int ret; if (params->codec.id != SND_AUDIOCODEC_IEC61937) { dev_err(dev, "Codec ID is not supported(%d)\n", @@ -161,11 +160,7 @@ static int uniphier_aio_compr_set_params(struct snd_soc_component *component, aio_port_reset(sub); aio_src_reset(sub); - ret = uniphier_aio_compr_prepare(component, cstream); - if (ret) - return ret; - - return 0; + return uniphier_aio_compr_prepare(component, cstream); } static int uniphier_aio_compr_hw_free(struct snd_soc_component *component, -- 2.25.1

2 1

[PATCH v2] soundwire: qcom: adjust autoenumeration timeout
by Srinivas Kandagatla 09 May '22

09 May '22

Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: replaced HZ/10 with 100 as suggested by Pierre drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 7367aa88b8ac..d6111f69d320 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -105,7 +105,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.21.0

3 9

Re: [PATCH v3 00/21] xen: simplify frontend side ring setup
by Oleksandr 09 May '22

09 May '22

On 05.05.22 11:16, Juergen Gross wrote: Hello Juergen. > Many Xen PV frontends share similar code for setting up a ring page > (allocating and granting access for the backend) and for tearing it > down. > > Create new service functions doing all needed steps in one go. > > This requires all frontends to use a common value for an invalid > grant reference in order to make the functions idempotent. > > Changes in V3: > - new patches 1 and 2, comments addressed > > Changes in V2: > - new patch 9 and related changes in patches 10-18 > > Juergen Gross (21): > xen: update grant_table.h > xen/grant-table: never put a reserved grant on the free list > xen/blkfront: switch blkfront to use INVALID_GRANT_REF > xen/netfront: switch netfront to use INVALID_GRANT_REF > xen/scsifront: remove unused GRANT_INVALID_REF definition > xen/usb: switch xen-hcd to use INVALID_GRANT_REF > xen/drm: switch xen_drm_front to use INVALID_GRANT_REF > xen/sound: switch xen_snd_front to use INVALID_GRANT_REF > xen/dmabuf: switch gntdev-dmabuf to use INVALID_GRANT_REF > xen/shbuf: switch xen-front-pgdir-shbuf to use INVALID_GRANT_REF > xen: update ring.h > xen/xenbus: add xenbus_setup_ring() service function > xen/blkfront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/netfront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/tpmfront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/drmfront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/pcifront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/scsifront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/usbfront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/sndfront: use xenbus_setup_ring() and xenbus_teardown_ring() > xen/xenbus: eliminate xenbus_grant_ring() For the patches that touch PV display (#07, #16), PV sound (#08, #20) and shared buffer framework used by both frontends (#10): Reviewed-by: Oleksandr Tyshchenko <oleksandr_tyshchenko(a)epam.com> Also I didn't see any issues with these frontends while testing on Arm64 based board. So, you can also add: [Arm64 only] Tested-by: Oleksandr Tyshchenko <oleksandr_tyshchenko(a)epam.com> Thanks! > > drivers/block/xen-blkfront.c | 57 +++---- > drivers/char/tpm/xen-tpmfront.c | 18 +-- > drivers/gpu/drm/xen/xen_drm_front.h | 9 -- > drivers/gpu/drm/xen/xen_drm_front_evtchnl.c | 43 ++---- > drivers/net/xen-netfront.c | 85 ++++------- > drivers/pci/xen-pcifront.c | 19 +-- > drivers/scsi/xen-scsifront.c | 31 +--- > drivers/usb/host/xen-hcd.c | 65 ++------ > drivers/xen/gntdev-dmabuf.c | 13 +- > drivers/xen/grant-table.c | 12 +- > drivers/xen/xen-front-pgdir-shbuf.c | 18 +-- > drivers/xen/xenbus/xenbus_client.c | 82 +++++++--- > include/xen/grant_table.h | 2 - > include/xen/interface/grant_table.h | 161 ++++++++++++-------- > include/xen/interface/io/ring.h | 19 ++- > include/xen/xenbus.h | 4 +- > sound/xen/xen_snd_front_evtchnl.c | 44 ++---- > sound/xen/xen_snd_front_evtchnl.h | 9 -- > 18 files changed, 287 insertions(+), 404 deletions(-) > -- Regards, Oleksandr Tyshchenko

1 0

[PATCH v6 0/2] Add generic serial MIDI driver using serial bus API
by Daniel Kaehn 09 May '22

09 May '22

Generic serial MIDI driver adding support for using serial devices compatible with the serial bus as raw MIDI devices, allowing using additional serial devices not compatible with the existing serial-u16550 driver. Supports only setting standard serial baudrates on the underlying serial device; however, the underlying serial device can be configured so that a requested 38.4 kBaud is actually the standard MIDI 31.25 kBaud. Supports DeviceTree configuration. Changes in v6: - Change compatible "serialmidi" -> serial-midi" - Default current-speed to 38400 (closest baud to MIDI standard speed) - Appropriately stop reading or writing MIDI if input or output _trigger() is called with a parameter of zero, respectively - Zero out corresponding triggered state on close to ensure input and output closing results in the serial port being closed - Fix order of operations in _probe() - Remove "DEBUG" literal from debug messages - Remove unused dt-parsing patch checking for existence of node - Whitespace / tabbing fixes / improvements Changes in v5: - Reword description in dt-binding for clarity - Change 'speed' dt property to standard 'current-speed' - Move MIDI output loop onto workqueue (since this could loop quite a while, if ALSA provides a continuous stream of bytes) - Add tx_state bit flags to snd_serial_generic struct - Safegard critical section in tx_work with atomic bit ops on tx_state - Switch operations on filemode to use atomic bit ops Changes in v4: - Fix regressed typo - Correct 3.84 kBaud -> 38.4 kBaud in DT & Kconfig (sorry about spam - noticed after sending v3 and didn't want to let the error sit around for too long) Changes in v3: - Replace use of snd_printk() with dev_* alternatives - Removed unnecessary initialization of err variables - Replaced instances of `== SERIAL_MODE_NOT_OPENED` with zero check - Loop on output_write to completely fill output buffer if data available - Depend on CONFIG_OF in Kconfig - Replace use of devm_kzalloc() with extra_size allocation in snd_devm_card_new() - Use module_serdev_device_driver() instead of module_init() and module_exit(0) Changes in v2: - Fix 'snd_serial_generic_write_wakeup' missing static keyword - Correct 3.125 kBaud > 31.25 kBaud in documentation for MIDI The need for this driver arose from a project using a Raspberry Pi4 which needed to receive and send raw MIDI with low latency. The pl011 UART used is not compatible with the existing serial MIDI driver made for u16550-style devices. Using a userspace program such as ttymidi to feed input from the TTY device to a virtual ALSA MIDI device was functional, but not ideal. I am not sure if a MIDI driver needing the mentioned 'hack' to clock 38.4 kBaud down to the standard MIDI baud is permissible in the mainline kernel, but am submitting nevertheless in case it is useful. To my knowledge, it doesn't seem that there would be any way for this driver to manually configure a serial port to 31.25 kBaud using the serial bus API (please correct me f I'm wrong). In my use case, I am actually configuring one port to run at 115.2 kBaud for faster communication with a custom onboard MIDI controller. Daniel Kaehn (2): dt-bindings: sound: Add generic serial MIDI device Add generic serial MIDI driver using serial bus API .../bindings/sound/serial-midi.yaml | 50 +++ sound/drivers/Kconfig | 18 + sound/drivers/Makefile | 2 + sound/drivers/serial-generic.c | 374 ++++++++++++++++++ 4 files changed, 444 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/serial-midi.yaml create mode 100644 sound/drivers/serial-generic.c base-commit: c5eb0a61238dd6faf37f58c9ce61c9980aaffd7a -- 2.33.0

2 3

[RFC v2 31/39] sound: add HAS_IOPORT dependencies
by Niklas Schnelle 09 May '22

09 May '22

In a future patch HAS_IOPORT=n will result in inb()/outb() and friends not being declared. We thus need to add HAS_IOPORT as dependency for those drivers using them. For SND_OPL3_LIB this adds its first dependency so drivers currently selecting it unconditionally need to depend on it instead. Co-developed-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Niklas Schnelle <schnelle(a)linux.ibm.com> --- sound/drivers/Kconfig | 5 ++++ sound/isa/Kconfig | 44 ++++++++++++++--------------- sound/pci/Kconfig | 64 +++++++++++++++++++++++++++++-------------- 3 files changed, 70 insertions(+), 43 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index ca4cdf666f82..4d250e619786 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,10 +1,12 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART tristate + depends on HAS_IOPORT select SND_RAWMIDI config SND_OPL3_LIB tristate + depends on HAS_IOPPORT select SND_TIMER select SND_HWDEP select SND_SEQ_DEVICE if SND_SEQUENCER != n @@ -128,6 +130,7 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" + depends on HAS_IOPORT select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -152,6 +155,7 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" + depends on HAS_IOPORT select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -167,6 +171,7 @@ config SND_SERIAL_U16550 config SND_MPU401 tristate "Generic MPU-401 UART driver" + depends on HAS_IOPORT select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 570b88e0b201..072265429f39 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -31,7 +31,7 @@ if SND_ISA config SND_ADLIB tristate "AdLib FM card" - select SND_OPL3_LIB + depends on SND_OPL3_LIB help Say Y here to include support for AdLib FM cards. @@ -42,7 +42,7 @@ config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" depends on PNP select ISAPNP - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM select SND_TIMER @@ -70,7 +70,7 @@ config SND_ALS100 tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" depends on PNP select ISAPNP - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP help @@ -86,7 +86,7 @@ config SND_AZT1605 depends on SND select SND_WSS_LIB select SND_MPU401_UART - select SND_OPL3_LIB + depends on SND_OPL3_LIB help Say Y here to include support for Aztech Sound Galaxy cards based on the AZT1605 chipset. @@ -99,7 +99,7 @@ config SND_AZT2316 depends on SND select SND_WSS_LIB select SND_MPU401_UART - select SND_OPL3_LIB + depends on SND_OPL3_LIB help Say Y here to include support for Aztech Sound Galaxy cards based on the AZT2316 chipset. @@ -111,7 +111,7 @@ config SND_AZT2320 tristate "Aztech Systems AZT2320" depends on PNP select ISAPNP - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help @@ -124,7 +124,7 @@ config SND_AZT2320 config SND_CMI8328 tristate "C-Media CMI8328" select SND_WSS_LIB - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART help Say Y here to include support for soundcards based on the @@ -137,7 +137,7 @@ config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB select SND_SB16_DSP - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART help Say Y here to include support for soundcards based on the @@ -159,7 +159,7 @@ config SND_CS4231 config SND_CS4236 tristate "Generic Cirrus Logic CS4232/CS4236+ driver" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help @@ -172,7 +172,7 @@ config SND_CS4236 config SND_ES1688 tristate "Generic ESS ES688/ES1688 and ES968 PnP driver" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help @@ -184,7 +184,7 @@ config SND_ES1688 config SND_ES18XX tristate "Generic ESS ES18xx driver" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help @@ -197,7 +197,7 @@ config SND_SC6000 tristate "Gallant SC-6000/6600/7000 and Audio Excel DSP 16" depends on HAS_IOPORT_MAP select SND_WSS_LIB - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART help Say Y here to include support for Gallant SC-6000, SC-6600, SC-7000 @@ -223,7 +223,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM select SND_TIMER @@ -273,7 +273,7 @@ config SND_INTERWAVE_STB config SND_JAZZ16 tristate "Media Vision Jazz16 card and compatibles" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_SB8_DSP help @@ -289,7 +289,7 @@ config SND_JAZZ16 config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help @@ -301,7 +301,7 @@ config SND_OPL3SA2 config SND_OPTI92X_AD1848 tristate "OPTi 82C92x - AD1848" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART select SND_WSS_LIB @@ -314,7 +314,7 @@ config SND_OPTI92X_AD1848 config SND_OPTI92X_CS4231 tristate "OPTi 82C92x - CS4231" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART select SND_WSS_LIB @@ -327,7 +327,7 @@ config SND_OPTI92X_CS4231 config SND_OPTI93X tristate "OPTi 82C93x" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help @@ -352,7 +352,7 @@ config SND_MIRO config SND_SB8 tristate "Sound Blaster 1.0/2.0/Pro (8-bit)" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_RAWMIDI select SND_SB8_DSP help @@ -364,7 +364,7 @@ config SND_SB8 config SND_SB16 tristate "Sound Blaster 16 (PnP)" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP help @@ -376,7 +376,7 @@ config SND_SB16 config SND_SBAWE tristate "Sound Blaster AWE (32,64) (PnP)" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP select SND_SEQ_DEVICE if SND_SEQUENCER != n @@ -427,7 +427,7 @@ config SND_SSCAPE config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" select FW_LOADER - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index a55836225401..c9fd973a7893 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -25,8 +25,8 @@ config SND_ALS300 tristate "Avance Logic ALS300/ALS300+" select SND_PCM select SND_AC97_CODEC - select SND_OPL3_LIB - depends on ZONE_DMA + depends on SND_OPL3_LIB + depends on ZONE_DMA && HAS_IOPORT help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -36,7 +36,7 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" depends on ISA_DMA_API - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM select SND_SB_COMMON @@ -51,7 +51,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -96,6 +96,7 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -110,6 +111,7 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -123,6 +125,7 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -151,13 +154,13 @@ config SND_AW2 config SND_AZT3328 tristate "Aztech AZF3328 / PCI168" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM select SND_RAWMIDI select SND_AC97_CODEC select SND_TIMER - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -193,6 +196,7 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" + depends on HAS_IOPORT select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -205,7 +209,8 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" - select SND_OPL3_LIB + depends on HAS_IOPORT + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help @@ -221,6 +226,7 @@ config SND_OXYGEN_LIB config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -246,7 +252,8 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" - select SND_OPL3_LIB + depends on HAS_IOPORT + depends on SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC help @@ -257,6 +264,7 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select FW_LOADER @@ -290,6 +298,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on X86_32 || MIPS || COMPILE_TEST + depends on HAS_IOPORT select SND_PCM select SND_AC97_CODEC help @@ -307,6 +316,7 @@ config SND_CS5535AUDIO config SND_CTXFI tristate "Creative Sound Blaster X-Fi" + depends on HAS_IOPORT select SND_PCM help If you want to use soundcards based on Creative Sound Blastr X-Fi @@ -468,7 +478,7 @@ config SND_EMU10K1 select SND_AC97_CODEC select SND_TIMER select SND_SEQ_DEVICE if SND_SEQUENCER != n - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -491,7 +501,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -501,6 +511,7 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" + depends on HAS_IOPORT select SND_RAWMIDI select SND_PCM help @@ -511,6 +522,7 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC help @@ -522,10 +534,10 @@ config SND_ENS1371 config SND_ES1938 tristate "ESS ES1938/1946/1969 (Solo-1)" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -537,7 +549,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -569,7 +581,8 @@ config SND_ES1968_RADIO config SND_FM801 tristate "ForteMedia FM801" - select SND_OPL3_LIB + depends on HAS_IOPORT + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC help @@ -624,7 +637,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -640,6 +653,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER @@ -712,7 +726,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -753,6 +767,7 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" + depends on HAS_IOPORT select FW_LOADER select SND_PCM select SND_HWDEP @@ -764,8 +779,9 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" + depends on HAS_IOPORT select FW_LOADER - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC help @@ -808,6 +824,7 @@ config SND_RME9652 config SND_SE6X tristate "Studio Evolution SE6X" depends on SND_OXYGEN=n && SND_VIRTUOSO=n # PCI ID conflict + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -827,10 +844,10 @@ config SND_SIS7019 config SND_SONICVIBES tristate "S3 SonicVibes" - select SND_OPL3_LIB + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -842,7 +859,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. @@ -852,6 +869,7 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -863,6 +881,7 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" + depends on HAS_IOPORT select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -873,6 +892,7 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 66/100/200 (Xonar)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -889,6 +909,7 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" + depends on HAS_IOPORT select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -898,7 +919,8 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" - select SND_OPL3_LIB + depends on HAS_IOPORT + depends on SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC select SND_TIMER -- 2.32.0

3 4

Re: [PATCH] kernel/drivers: Remove redundant driver match function
by Tony Krowiak 09 May '22

09 May '22

For the drivers/s390/crypto/vfio_ap_drv.c device driver: Reviewed-by: Tony Krowiak <akrowiak(a)stny.rr.com> On 5/6/22 12:59 AM, lizhe wrote: > If there is no driver match function, the driver core assumes that each > candidate pair (driver, device) matches, see driver_match_device(). > > Signed-off-by: lizhe <sensor1010(a)163.com> > --- > drivers/mfd/mcp-core.c | 6 ------ > drivers/nubus/bus.c | 6 ------ > drivers/s390/crypto/vfio_ap_drv.c | 6 ------ > drivers/scsi/scsi_debug.c | 7 ------- > drivers/target/loopback/tcm_loop.c | 7 ------- > drivers/w1/w1.c | 6 ------ > sound/ac97_bus.c | 11 ----------- > 7 files changed, 49 deletions(-) > > diff --git a/drivers/mfd/mcp-core.c b/drivers/mfd/mcp-core.c > index 2fa592c37c6f..281a9369f2b3 100644 > --- a/drivers/mfd/mcp-core.c > +++ b/drivers/mfd/mcp-core.c > @@ -20,11 +20,6 @@ > #define to_mcp(d) container_of(d, struct mcp, attached_device) > #define to_mcp_driver(d) container_of(d, struct mcp_driver, drv) > > -static int mcp_bus_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > static int mcp_bus_probe(struct device *dev) > { > struct mcp *mcp = to_mcp(dev); > @@ -43,7 +38,6 @@ static void mcp_bus_remove(struct device *dev) > > static struct bus_type mcp_bus_type = { > .name = "mcp", > - .match = mcp_bus_match, > .probe = mcp_bus_probe, > .remove = mcp_bus_remove, > }; > diff --git a/drivers/nubus/bus.c b/drivers/nubus/bus.c > index 17fad660032c..72921e4f35f6 100644 > --- a/drivers/nubus/bus.c > +++ b/drivers/nubus/bus.c > @@ -14,11 +14,6 @@ > #define to_nubus_board(d) container_of(d, struct nubus_board, dev) > #define to_nubus_driver(d) container_of(d, struct nubus_driver, driver) > > -static int nubus_bus_match(struct device *dev, struct device_driver *driver) > -{ > - return 1; > -} > - > static int nubus_device_probe(struct device *dev) > { > struct nubus_driver *ndrv = to_nubus_driver(dev->driver); > @@ -39,7 +34,6 @@ static void nubus_device_remove(struct device *dev) > > struct bus_type nubus_bus_type = { > .name = "nubus", > - .match = nubus_bus_match, > .probe = nubus_device_probe, > .remove = nubus_device_remove, > }; > diff --git a/drivers/s390/crypto/vfio_ap_drv.c b/drivers/s390/crypto/vfio_ap_drv.c > index 29ebd54f8919..0a662c451f2a 100644 > --- a/drivers/s390/crypto/vfio_ap_drv.c > +++ b/drivers/s390/crypto/vfio_ap_drv.c > @@ -172,14 +172,8 @@ static void vfio_ap_matrix_dev_release(struct device *dev) > kfree(matrix_dev); > } > > -static int matrix_bus_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > static struct bus_type matrix_bus = { > .name = "matrix", > - .match = &matrix_bus_match, > }; > > static struct device_driver matrix_driver = { > diff --git a/drivers/scsi/scsi_debug.c b/drivers/scsi/scsi_debug.c > index 592a290e6cfa..8107489b36e8 100644 > --- a/drivers/scsi/scsi_debug.c > +++ b/drivers/scsi/scsi_debug.c > @@ -7844,15 +7844,8 @@ static void sdebug_driver_remove(struct device *dev) > scsi_host_put(sdbg_host->shost); > } > > -static int pseudo_lld_bus_match(struct device *dev, > - struct device_driver *dev_driver) > -{ > - return 1; > -} > - > static struct bus_type pseudo_lld_bus = { > .name = "pseudo", > - .match = pseudo_lld_bus_match, > .probe = sdebug_driver_probe, > .remove = sdebug_driver_remove, > .drv_groups = sdebug_drv_groups, > diff --git a/drivers/target/loopback/tcm_loop.c b/drivers/target/loopback/tcm_loop.c > index 4407b56aa6d1..eeb63deff94f 100644 > --- a/drivers/target/loopback/tcm_loop.c > +++ b/drivers/target/loopback/tcm_loop.c > @@ -83,15 +83,8 @@ static int tcm_loop_show_info(struct seq_file *m, struct Scsi_Host *host) > static int tcm_loop_driver_probe(struct device *); > static void tcm_loop_driver_remove(struct device *); > > -static int pseudo_lld_bus_match(struct device *dev, > - struct device_driver *dev_driver) > -{ > - return 1; > -} > - > static struct bus_type tcm_loop_lld_bus = { > .name = "tcm_loop_bus", > - .match = pseudo_lld_bus_match, > .probe = tcm_loop_driver_probe, > .remove = tcm_loop_driver_remove, > }; > diff --git a/drivers/w1/w1.c b/drivers/w1/w1.c > index f2ae2e563dc5..a6ecfa1b3417 100644 > --- a/drivers/w1/w1.c > +++ b/drivers/w1/w1.c > @@ -58,11 +58,6 @@ MODULE_PARM_DESC(slave_ttl, > DEFINE_MUTEX(w1_mlock); > LIST_HEAD(w1_masters); > > -static int w1_master_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > static int w1_master_probe(struct device *dev) > { > return -ENODEV; > @@ -174,7 +169,6 @@ static int w1_uevent(struct device *dev, struct kobj_uevent_env *env); > > static struct bus_type w1_bus_type = { > .name = "w1", > - .match = w1_master_match, > .uevent = w1_uevent, > }; > > diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c > index b4685c53ff11..c7aee8c42c55 100644 > --- a/sound/ac97_bus.c > +++ b/sound/ac97_bus.c > @@ -75,19 +75,8 @@ int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, > } > EXPORT_SYMBOL_GPL(snd_ac97_reset); > > -/* > - * Let drivers decide whether they want to support given codec from their > - * probe method. Drivers have direct access to the struct snd_ac97 > - * structure and may decide based on the id field amongst other things. > - */ > -static int ac97_bus_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > struct bus_type ac97_bus_type = { > .name = "ac97", > - .match = ac97_bus_match, > }; > > static int __init ac97_bus_init(void)

1 0

[PATCH v5 0/2] Let userspace know when snd-hda-intel needs i915
by Mauro Carvalho Chehab 09 May '22

09 May '22

Currently, kernel/module annotates module dependencies when request_symbol is used, but it doesn't cover more complex inter-driver dependencies that are subsystem and/or driver-specific. In the case of hdmi sound, depending on the CPU/GPU, sometimes the snd_hda_driver can talk directly with the hardware, but sometimes, it uses the i915 driver. When the snd_hda_driver uses i915, it should first be unbind/rmmod, as otherwise trying to unbind/rmmod the i915 driver cause driver issues, as as reported by CI tools with different GPU models: https://intel-gfx-ci.01.org/tree/drm-tip/IGT_6415/fi-tgl-1115g4/igt@core_ho… https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_11495/bat-adlm-1/igt@i915_m… In the past, just a few CPUs were doing such bindings, but this issue now applies to all "modern" Intel CPUs that have onboard graphics, as well as to the newer discrete GPUs. With the discrete GPU case, the HDA controller is physically separate and requires i915 to power on the hardware for all hardware access. In this case, the issue is hit basicly 100% of the time. With on-board graphics, i915 driver is needed only when the display codec is accessed. If i915 is unbind during runtime suspend, while snd-hda-intel is still bound, nothing bad happens, but unbinding i915 on other situations may also cause issues. So, add support at kernel/modules to allow snd-hda drivers to properly annotate when a dependency on a DRM driver dependencies exists, and add a call to such new function at the snd-hda driver when it successfully binds into the DRM driver. This would allow userspace tools to check and properly remove the audio driver before trying to remove or unbind the GPU driver. It should be noticed that this series conveys the hidden module dependencies. Other changes are needed in order to allow removing or unbinding the i915 driver while keeping the snd-hda-intel driver loaded/bound. With that regards, there are some discussions on how to improve this at alsa-devel a while back: https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/190099… So, future improvements on both in i915 and the audio drivers could be made. E.g. with discrete GPUs, it's the only codec of the card, so it seems feasible to detach the ALSA card if i915 is bound (using infra made for VGA switcheroo), but, until these improvements are done and land in upstream, audio drivers needs to be unbound if i915 driver goes unbind. Yet, even if such fixes got merged, this series is still needed, as it makes such dependencies more explicit and easier to debug. PS.: This series was generated against next-20220428. --- v5: - while v4 works fine, it ends calling try_module_format() recursively, which is not what it it was supposed to do. So, change the logic to avoid such recursion, by adding a static __try_module_format() and renaming the new version that takes two arguments as try_module_format_owner(). v4: - fix a compilation warning reported by Intel's Kernel robot when !CONFIG_MODULE_UNLOAD or !CONFIG_MODULE. v3: minor fixes: - fixed a checkpatch warning; - use a single line for the new function prototype. v2: - the dependencies are now handled directly at try_module_get(). Mauro Carvalho Chehab (2): module: update dependencies at try_module_get() ALSA: hda - identify when audio is provided by a video driver include/linux/module.h | 8 +++-- kernel/module/main.c | 65 ++++++++++++++++++++++++++++++-------- sound/hda/hdac_component.c | 2 +- 3 files changed, 57 insertions(+), 18 deletions(-) -- 2.35.1

3 4

[PATCH v1 0/3] Add Manual Mode Jack Detection for CS42L42 in HDA
by Stefan Binding 09 May '22

09 May '22

CS42L42 supports automatic type detection, to determine headset type, or headphones. However, automatic type detection does not always work correctly. To improve detection accuracy, manual mode detection can be run. This is already done in the ASoC version of the CS42L42 driver, and this chain adds the same support into the HDA version. In addition, to better align the ASoC and HDA drivers, the ASoC header file has been moved into the general sound includes, which allows both ASoC and HDA to have access to the header. This improves the maintainability of the HDA driver, and makes it easier to port fixes/features from CS42L42 ASoC to HDA. Stefan Binding (3): ASoC: cs42l42: Move CS42L42 register descriptions to general include ALSA: hda/cs8409: Use general cs42l42 include in cs8409 hda driver ALSA: hda/cs8409: Support manual mode detection for CS42L42 include/sound/cs42l42.h | 810 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409-tables.c | 327 ++++++----- sound/pci/hda/patch_cs8409.c | 275 ++++++--- sound/pci/hda/patch_cs8409.h | 9 +- sound/soc/codecs/cs42l42.h | 826 +--------------------------- 5 files changed, 1167 insertions(+), 1080 deletions(-) create mode 100644 include/sound/cs42l42.h -- 2.25.1

4 6

[PATCH] ALSA: hdsp: remove redundant assignment to pointer kctl
by Colin Ian King 09 May '22

09 May '22

Pointer kctl is being assigned a value that is not being read, buf is being re-assigned later. The assignment is redundant and can be removed. Cleans up clang scan build warning: sound/pci/rme9652/hdsp.c:3317:28: warning: Although the value stored to 'kctl' is used in the enclosing expression, the value is never actually read from 'kctl' [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/rme9652/hdsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 3db641318d3a..dcc43a81ae0e 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3314,7 +3314,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) if (hdsp->io_type == RPM) { /* RPM Bypass, Disconnect and Input switches */ for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_rpm_controls); idx++) { - err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_rpm_controls[idx], hdsp)); + err = snd_ctl_add(card, snd_ctl_new1(&snd_hdsp_rpm_controls[idx], hdsp)); if (err < 0) return err; } -- 2.35.1

2 1

Re: [PATCH] kernel/drivers: Remove redundant driver match function
by Uwe Kleine-König 09 May '22

09 May '22

Hello, On Thu, May 05, 2022 at 09:59:52PM -0700, lizhe wrote: > If there is no driver match function, the driver core assumes that each > candidate pair (driver, device) matches, see driver_match_device(). > > Signed-off-by: lizhe <sensor1010(a)163.com> Reviewed-by: Uwe Kleine-König <u.kleine-koenig.org> Side note: While looking through this patch I was surprised to see there are two different busses for ac97: sound/ac97/bus.c + sound/ac97_bus.c . It seems the duplication exists since 2017. Best regards Uwe -- Pengutronix e.K. | Uwe Kleine-König | Industrial Linux Solutions | https://www.pengutronix.de/ |

2 1

[PATCH 0/5] soundwire: use pm_runtime_resume_and_get()
by Bard Liao 09 May '22

09 May '22

Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. Pierre-Louis Bossart (5): soundwire: qcom: return error when pm_runtime_get_sync fails soundwire: bus: use pm_runtime_resume_and_get() soundwire: cadence: use pm_runtime_resume_and_get() soundwire: intel: use pm_runtime_resume_and_get() soundwire: qcom: use pm_runtime_resume_and_get() drivers/soundwire/bus.c | 15 +++++---------- drivers/soundwire/cadence_master.c | 5 ++--- drivers/soundwire/intel.c | 5 ++--- drivers/soundwire/qcom.c | 17 ++++++++--------- 4 files changed, 17 insertions(+), 25 deletions(-) -- 2.25.1

3 8

[PATCH 0/3] soundwire: pm runtime improvements
by Bard Liao 09 May '22

09 May '22

This series provides a solution to solve a corner case issue where the manager device may become pm_runtime active, but without ALSA/ASoC requesting any functionality from the peripherals. In this case, the hardware peripheral device will report as ATTACHED and its initialization routine will be executed. If this initialization routine initiates any sort of deferred processing, there is a possibility that the manager could suspend without the peripheral suspend sequence being invoked: from the pm_runtime framework perspective, the peripheral is *already* suspended. Pierre-Louis Bossart (3): soundwire: intel: prevent pm_runtime resume prior to system suspend soundwire: intel: disable WAKEEN in pm_runtime resume soundwire: bus: pm_runtime_request_resume on peripheral attachment drivers/soundwire/bus.c | 12 ++++++++++++ drivers/soundwire/intel.c | 6 ++++++ 2 files changed, 18 insertions(+) -- 2.17.1

2 4

[PATCH] soundwire: cadence: recheck device0 attachment after status change
by Bard Liao 09 May '22

09 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> This patch adds a status check after device0 attachment to solve race conditions observed during attachment with multiple devices per link The sequence is the following 1) deviceA attaches as device0 2) the hardware detects a device0 status change and throws an interrupt. 3) the interrupt handler schedules the work function 4) the workqueue starts, we read the status slave0 = cdns_readl(cdns, CDNS_MCP_SLAVE_INTSTAT0); slave1 = cdns_readl(cdns, CDNS_MCP_SLAVE_INTSTAT1); we deal with the status change and program deviceA device number to a non-zero value. 5) deviceB attaches as device0, the device0 status seen by the hardware does not change. 6) we clear the CDNS_MCP_SLAVE_INTSTAT0/1 registers -> we will never detect deviceB! This patch suggest re-checking in a loop the device0 status with a PING frame, i.e. using the real device0 status instead of information on status changes. 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 | 37 ++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 558390af44b6..47d59190a96e 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -959,6 +959,8 @@ static void cdns_update_slave_status_work(struct work_struct *work) container_of(work, struct sdw_cdns, work); u32 slave0, slave1; u64 slave_intstat; + u32 device0_status; + int retry_count = 0; slave0 = cdns_readl(cdns, CDNS_MCP_SLAVE_INTSTAT0); slave1 = cdns_readl(cdns, CDNS_MCP_SLAVE_INTSTAT1); @@ -968,10 +970,45 @@ static void cdns_update_slave_status_work(struct work_struct *work) dev_dbg_ratelimited(cdns->dev, "Slave status change: 0x%llx\n", slave_intstat); +update_status: cdns_update_slave_status(cdns, slave_intstat); cdns_writel(cdns, CDNS_MCP_SLAVE_INTSTAT0, slave0); cdns_writel(cdns, CDNS_MCP_SLAVE_INTSTAT1, slave1); + /* + * When there is more than one peripheral per link, it's + * possible that a deviceB becomes attached after we deal with + * the attachment of deviceA. Since the hardware does a + * logical AND, the attachment of the second device does not + * change the status seen by the driver. + * + * In that case, clearing the registers above would result in + * the deviceB never being detected - until a change of status + * is observed on the bus. + * + * To avoid this race condition, re-check if any device0 needs + * attention with PING commands. There is no need to check for + * ALERTS since they are not allowed until a non-zero + * device_number is assigned. + */ + + device0_status = cdns_readl(cdns, CDNS_MCP_SLAVE_STAT); + device0_status &= 3; + + if (device0_status == SDW_SLAVE_ATTACHED) { + if (retry_count++ < SDW_MAX_DEVICES) { + dev_dbg_ratelimited(cdns->dev, + "Device0 detected after clearing status, iteration %d\n", + retry_count); + slave_intstat = CDNS_MCP_SLAVE_INTSTAT_ATTACHED; + goto update_status; + } else { + dev_err_ratelimited(cdns->dev, + "Device0 detected after %d iterations\n", + retry_count); + } + } + /* clear and unmask Slave interrupt now */ cdns_writel(cdns, CDNS_MCP_INTSTAT, CDNS_MCP_INT_SLAVE_MASK); cdns_updatel(cdns, CDNS_MCP_INTMASK, -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Dell Latitude 7520
by gabriele.mzt＠gmail.com 08 May '22

08 May '22

From: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> This fixes the pop noises that can be heard at boot time if the headphone jack is plugged in or when inserting/removing the jack. Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62fbf3772b41..fa9878daa3bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6780,6 +6780,23 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc_fixup_dell4_mic_vref(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + /** + * Set the vref of pin 0x19 (Headset Mic) and pin 0x1b (Headphone Mic) + * to Hi-Z to avoid pop noises at startup and when plugging and + * unplugging headphones/speakers. + */ + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); + snd_hda_codec_set_pin_target(codec, 0x1b, PIN_VREFHIZ); + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7011,6 +7028,7 @@ enum { ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, + ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_VREF, }; static const struct hda_fixup alc269_fixups[] = { @@ -8799,6 +8817,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_MUTE_LED, }, + [ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_dell4_mic_vref, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8889,6 +8913,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), 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, 0x0a38, "Dell Latitude 7520", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_VREF), 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, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), -- 2.36.0

3 2

[PATCH] ALSA: hda - fix unused Realtek function when PM is not enabled
by Randy Dunlap 08 May '22

08 May '22

When CONFIG_PM is not enabled, alc_shutup() is not needed, so move it inside the #ifdef CONFIG_PM guard. Also drop some contiguous #endif / #ifdef CONFIG_PM for simplicity. Fixes this build warning: sound/pci/hda/patch_realtek.c:886:20: warning: unused function 'alc_shutup' Fixes: 08c189f2c552 ("ALSA: hda - Use generic parser codes for Realtek driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -937,6 +937,9 @@ static int alc_init(struct hda_codec *co return 0; } +#define alc_free snd_hda_gen_free + +#ifdef CONFIG_PM static inline void alc_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -950,9 +953,6 @@ static inline void alc_shutup(struct hda alc_shutup_pins(codec); } -#define alc_free snd_hda_gen_free - -#ifdef CONFIG_PM static void alc_power_eapd(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); @@ -966,9 +966,7 @@ static int alc_suspend(struct hda_codec spec->power_hook(codec); return 0; } -#endif -#ifdef CONFIG_PM static int alc_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec;

2 1

[PATCH] ALSA: usb-audio: Don't get sample rate for MCT Trigger 5 USB-to-HDMI
by Forest Crossman 08 May '22

08 May '22

This device doesn't support reading the sample rate, so we need to apply this quirk to avoid a 15-second delay waiting for three timeouts. Signed-off-by: Forest Crossman <cyrozap(a)gmail.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..fbbe59054c3f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1822,6 +1822,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ -- 2.36.0

2 1

[PATCH] add a new hda codec SN6140
by bo liu 08 May '22

08 May '22

The current kernel does not support the SN6140 codec chip. Add the SN6140 codec configuration item to kernel. Signed-off-by: bo liu <bo.liu(a)senarytech.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0515137a75b0..aa360a0af284 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1105,6 +1105,7 @@ static int patch_conexant_auto(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f11f87, "SN6140", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), -- 2.34.1

2 1

[GIT PULL] ASoC fixes for v5.18-rc4
by Mark Brown 08 May '22

08 May '22

The following changes since commit 5b933c7262c5b0ea11ea3c3b3ea81add04895954: firmware: cs_dsp: Fix overrun of unterminated control name string (2022-04-12 17:57:04 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.18-rc4 for you to fetch changes up to c61711c1c95791850be48dd65a1d72eb34ba719f: ASoC: SOF: Fix NULL pointer exception in sof_pci_probe callback (2022-04-27 14:15:21 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.18 A larger collection of fixes than I'd like, mainly because mixer-test is making it's way into the CI systems and turning up issues on a wider range of systems. The most substantial thing though is a revert and an alternative fix for a dmaengine issue where the fix caused disruption for some other configurations, the core fix is backed out an a driver specific thing done instead. ---------------------------------------------------------------- Ajit Kumar Pandey (1): ASoC: SOF: Fix NULL pointer exception in sof_pci_probe callback ChiYuan Huang (1): ASoC: rt9120: Correct the reg 0x09 size to one byte Codrin Ciubotariu (2): ASoC: atmel: mchp-pdmc: set prepare_slave_config ASoC: dmaengine: Restore NULL prepare_slave_config() callback Mark Brown (8): ASoC: wm8958: Fix change notifications for DSP controls ASoC: da7219: Fix change notifications for tone generator frequency ASoC: meson: Fix event generation for AUI ACODEC mux ASoC: meson: Fix event generation for AUI CODEC mux ASoC: meson: Fix event generation for G12A tohdmi mux ASoC: max98090: Reject invalid values in custom control put() ASoC: max98090: Generate notifications on changes for custom control ASoC: ops: Validate input values in snd_soc_put_volsw_range() Neil Armstrong (2): ASoC: meson: axg-tdm-interface: Fix formatters in trigger" ASoC: meson: axg-card: Fix nonatomic links Olivier Moysan (1): ASoC: simple-card-utils: fix sysclk shutdown Pierre-Louis Bossart (1): ASoC: soc-ops: fix error handling sound/soc/atmel/mchp-pdmc.c | 1 + sound/soc/codecs/da7219.c | 14 ++++++++++---- sound/soc/codecs/max98090.c | 5 ++++- sound/soc/codecs/rt9120.c | 1 - sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- sound/soc/generic/simple-card-utils.c | 2 +- sound/soc/meson/aiu-acodec-ctrl.c | 2 +- sound/soc/meson/aiu-codec-ctrl.c | 2 +- sound/soc/meson/axg-card.c | 1 - sound/soc/meson/axg-tdm-interface.c | 26 +++++--------------------- sound/soc/meson/g12a-tohdmitx.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +++--- sound/soc/soc-ops.c | 20 ++++++++++++++++++-- sound/soc/sof/sof-pci-dev.c | 5 +++++ 14 files changed, 54 insertions(+), 41 deletions(-)

2 1

[PATCH v2 04/11] dt-bindings: sound: add Arm PL041 AACI DT schema
by Andre Przywara 07 May '22

07 May '22

The Arm PrimeCell Advanced Audio CODEC Interface (AACI aka PL041) is a peripheral that provides communication with an audio CODEC. Add a simple DT schema binding for it, so that DTs can be validated automatically. Signed-off-by: Andre Przywara <andre.przywara(a)arm.com> --- .../devicetree/bindings/sound/arm,pl041.yaml | 62 +++++++++++++++++++ 1 file changed, 62 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arm,pl041.yaml diff --git a/Documentation/devicetree/bindings/sound/arm,pl041.yaml b/Documentation/devicetree/bindings/sound/arm,pl041.yaml new file mode 100644 index 0000000000000..7896b8150cf0f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arm,pl041.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/arm,pl041.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Arm Ltd. PrimeCell PL041 AACI sound interface + +maintainers: + - Andre Przywara <andre.przywara(a)arm.com> + +description: + The Arm PrimeCell Advanced Audio CODEC Interface (AACI) is an AMBA compliant + peripheral that provides communication with an audio CODEC using the AC-link + protocol. + +# We need a select here so we don't match all nodes with 'arm,primecell' +select: + properties: + compatible: + contains: + const: arm,pl041 + required: + - compatible + +properties: + compatible: + items: + - const: arm,pl041 + - const: arm,primecell + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + description: APB register access clock + + clock-names: + const: apb_pclk + +required: + - compatible + - reg + - interrupts + - clocks + +additionalProperties: false + +examples: + - | + audio-controller@40000 { + compatible = "arm,pl041", "arm,primecell"; + reg = <0x040000 0x1000>; + interrupts = <11>; + clocks = <&v2m_clk24mhz>; + clock-names = "apb_pclk"; + }; + +... -- 2.25.1

2 1

[PATCH v2 00/48] ARM: PXA multiplatform support
by Arnd Bergmann 07 May '22

07 May '22

From: Arnd Bergmann <arnd(a)arndb.de> This revisits a series I sent a few years ago: https://lore.kernel.org/lkml/20191018154052.1276506-1-arnd@arndb.de/ All the other ARMv5 conversions are under way now, with OMAP1 being the only one still not in linux-next yet, and PXA completing the set. Most of the patches are unchanged from before, furtunately the PXA code is fairly stable. I addressed Robert's comments, pulled in two patches from Dmitry, and added the last a the final four patches to finish off the multiplatform conversion. I hope someone is left to test these on PXA: if this works, I'd like to merge it for 5.19. A git tree with these is avaialable for testing at https://git.kernel.org/pub/scm/linux/kernel/git/soc/soc.git/log/?h=pxa-mult… Arnd Arnd Bergmann (46): ARM: pxa: split mach/generic.h ARM: pxa: make mainstone.h private ARM: pxa: make mach/regs-uart.h private ARM: pxa: remove mach/dma.h ARM: pxa: split up mach/hardware.h ARM: pxa: stop using mach/bitfield.h ARM: pxa: move mach/sound.h to linux/platform_data/ ARM: pxa: move regs-lcd.h into driver watchdog: sa1100: use platform device registration ARM: pxa: pxa2xx-ac97-lib: use IRQ resource ARM: pxa: move pcmcia board data into mach-pxa ARM: pxa: make addr-map.h header local ARM: pxa: use pdev resource for palmld mmio ARM: pxa: maybe fix gpio lookup tables ARM: pxa: tosa: use gpio descriptor for audio ARM: pxa: poodle: use platform data for poodle asoc driver ARM: pxa: corgi: use gpio descriptors for audio ARM: pxa: hx4700: use gpio descriptors for audio ARM: pxa: lubbock: pass udc irqs as resource ARM: pxa: spitz: use gpio descriptors for audio ARM: pxa: eseries: use gpio lookup for audio ARM: pxa: z2: use gpio lookup for audio device ARM: pxa: magician: use platform driver for audio ARM: pxa: mainstone-wm97xx: use gpio lookup table ARM: pxa: zylonite: use gpio lookup instead mfp header input: touchscreen: mainstone: fix pxa2xx+pxa3xx configuration input: touchscreen: mainstone: sync with zylonite driver Input: touchscreen: use wrapper for pxa2xx ac97 registers ASoC: pxa: use pdev resource for FIFO regs ASoC: pxa: ac97: use normal MMIO accessors ASoC: pxa: i2s: use normal MMIO accessors ARM: pxa: pcmcia: move smemc configuration back to arch ARM: pxa: remove get_clk_frequency_khz() cpufreq: pxa3: move clk register access to clk driver ARM: pxa: move smemc register access from clk to platform ARM: pxa: move clk register definitions to driver power: tosa: simplify probe function ARM: pxa: tosa: use gpio lookup for battery ARM: pxa: remove unused mach/bitfield.h ARM: mmp: remove tavorevb board support ARM: mmp: rename pxa_register_device ARM: pxa: move plat-pxa to drivers/soc/ ARM: PXA: fix multi-cpu build of xsc3 ARM: pxa: move mach/*.h to mach-pxa/ ARM: pxa: remove support for MTD_XIP ARM: pxa: convert to multiplatform Dmitry Torokhov (2): Input: wm97xx - switch to using threaded IRQ Input: wm97xx - get rid of irq_enable method in wm97xx_mach_ops arch/arm/Kconfig | 22 -- arch/arm/Makefile | 1 - arch/arm/common/locomo.c | 1 - arch/arm/common/sa1111.c | 5 +- arch/arm/include/asm/hardware/sa1111.h | 2 - arch/arm/mach-mmp/Kconfig | 10 +- arch/arm/mach-mmp/Makefile | 1 - arch/arm/mach-mmp/devices.c | 2 +- arch/arm/mach-mmp/devices.h | 10 +- arch/arm/mach-mmp/mfp.h | 2 +- arch/arm/mach-mmp/mmp2.h | 48 ++--- arch/arm/mach-mmp/pxa168.h | 60 +++--- arch/arm/mach-mmp/pxa910.h | 38 ++-- arch/arm/mach-mmp/tavorevb.c | 113 ----------- arch/arm/mach-mmp/ttc_dkb.c | 6 +- arch/arm/mach-pxa/Kconfig | 14 ++ arch/arm/mach-pxa/Makefile | 18 +- arch/arm/mach-pxa/Makefile.boot | 3 - .../mach-pxa/{include/mach => }/addr-map.h | 0 arch/arm/mach-pxa/am300epd.c | 2 +- .../arm/mach-pxa/balloon3-pcmcia.c | 4 +- arch/arm/mach-pxa/balloon3.c | 4 +- .../mach-pxa/{include/mach => }/balloon3.h | 0 arch/arm/mach-pxa/cm-x300.c | 12 +- arch/arm/mach-pxa/colibri-evalboard.c | 1 - .../arm/mach-pxa/colibri-pcmcia.c | 2 +- arch/arm/mach-pxa/colibri-pxa270-income.c | 1 - arch/arm/mach-pxa/colibri-pxa270.c | 2 +- arch/arm/mach-pxa/colibri-pxa300.c | 3 +- arch/arm/mach-pxa/colibri-pxa320.c | 2 +- arch/arm/mach-pxa/colibri-pxa3xx.c | 3 +- arch/arm/mach-pxa/colibri.h | 2 +- arch/arm/mach-pxa/corgi.c | 23 ++- arch/arm/mach-pxa/{include/mach => }/corgi.h | 0 arch/arm/mach-pxa/corgi_pm.c | 5 +- arch/arm/mach-pxa/csb726.c | 5 +- arch/arm/mach-pxa/csb726.h | 2 +- arch/arm/mach-pxa/devices.c | 17 +- .../arm/mach-pxa/e740-pcmcia.c | 4 +- .../{include/mach => }/eseries-gpio.h | 0 arch/arm/mach-pxa/eseries.c | 36 +++- arch/arm/mach-pxa/ezx.c | 1 - arch/arm/mach-pxa/generic.c | 62 ++++-- arch/arm/mach-pxa/generic.h | 9 - arch/arm/mach-pxa/gumstix.c | 1 - arch/arm/mach-pxa/gumstix.h | 2 +- arch/arm/mach-pxa/h5000.c | 2 +- .../arm/mach-pxa/hx4700-pcmcia.c | 4 +- arch/arm/mach-pxa/hx4700.c | 18 +- arch/arm/mach-pxa/{include/mach => }/hx4700.h | 0 arch/arm/mach-pxa/idp.c | 2 - arch/arm/mach-pxa/idp.h | 2 +- arch/arm/mach-pxa/include/mach/bitfield.h | 114 ----------- arch/arm/mach-pxa/include/mach/dma.h | 17 -- arch/arm/mach-pxa/include/mach/generic.h | 1 - arch/arm/mach-pxa/include/mach/mtd-xip.h | 36 ---- arch/arm/mach-pxa/include/mach/uncompress.h | 70 ------- arch/arm/mach-pxa/irq.c | 5 +- arch/arm/mach-pxa/{include/mach => }/irqs.h | 0 arch/arm/mach-pxa/littleton.c | 1 - arch/arm/mach-pxa/lpd270.c | 6 +- arch/arm/mach-pxa/lubbock.c | 17 +- .../arm/mach-pxa/{include/mach => }/lubbock.h | 4 +- arch/arm/mach-pxa/magician.c | 56 +++++- .../mach-pxa/{include/mach => }/magician.h | 2 +- arch/arm/mach-pxa/mainstone.c | 17 +- .../mach-pxa/{include/mach => }/mainstone.h | 4 +- arch/arm/mach-pxa/mfp-pxa2xx.c | 3 +- arch/arm/mach-pxa/mfp-pxa2xx.h | 2 +- arch/arm/mach-pxa/mfp-pxa3xx.c | 3 +- arch/arm/mach-pxa/mfp-pxa3xx.h | 2 +- arch/arm/mach-pxa/{include/mach => }/mfp.h | 2 +- arch/arm/mach-pxa/mioa701.c | 4 +- arch/arm/mach-pxa/mxm8x10.c | 8 +- arch/arm/mach-pxa/palm27x.c | 2 +- .../arm/mach-pxa/palmld-pcmcia.c | 5 +- arch/arm/mach-pxa/palmld.c | 23 ++- arch/arm/mach-pxa/{include/mach => }/palmld.h | 0 arch/arm/mach-pxa/palmt5.c | 11 +- arch/arm/mach-pxa/palmt5.h | 2 +- .../arm/mach-pxa/palmtc-pcmcia.c | 4 +- arch/arm/mach-pxa/palmtc.c | 4 +- arch/arm/mach-pxa/{include/mach => }/palmtc.h | 0 arch/arm/mach-pxa/palmte2.c | 2 +- arch/arm/mach-pxa/palmtreo.c | 4 +- .../arm/mach-pxa/palmtx-pcmcia.c | 4 +- arch/arm/mach-pxa/palmtx.c | 13 +- arch/arm/mach-pxa/{include/mach => }/palmtx.h | 0 arch/arm/mach-pxa/palmz72.c | 2 +- arch/arm/mach-pxa/pcm027.h | 2 +- arch/arm/mach-pxa/pcm990-baseboard.c | 2 +- arch/arm/mach-pxa/pcm990_baseboard.h | 2 +- arch/arm/mach-pxa/poodle.c | 31 ++- arch/arm/mach-pxa/{include/mach => }/poodle.h | 2 - arch/arm/mach-pxa/pxa-dt.c | 2 +- arch/arm/mach-pxa/pxa-regs.h | 52 +++++ arch/arm/mach-pxa/pxa25x.c | 12 +- arch/arm/mach-pxa/pxa25x.h | 6 +- arch/arm/mach-pxa/pxa27x-udc.h | 2 + arch/arm/mach-pxa/pxa27x.c | 12 +- arch/arm/mach-pxa/pxa27x.h | 6 +- .../mach-pxa/{include/mach => }/pxa2xx-regs.h | 47 +---- arch/arm/mach-pxa/pxa2xx.c | 30 ++- arch/arm/mach-pxa/pxa300.c | 1 + arch/arm/mach-pxa/pxa320.c | 1 + .../mach-pxa/{include/mach => }/pxa3xx-regs.h | 71 +------ arch/arm/mach-pxa/pxa3xx-ulpi.c | 2 +- arch/arm/mach-pxa/pxa3xx.c | 19 +- arch/arm/mach-pxa/pxa3xx.h | 6 +- arch/arm/mach-pxa/pxa930.c | 1 + .../mach-pxa/{include/mach => }/regs-ost.h | 4 +- arch/arm/mach-pxa/regs-rtc.h | 2 +- arch/arm/mach-pxa/regs-u2d.h | 2 - .../mach-pxa/{include/mach => }/regs-uart.h | 2 + arch/arm/mach-pxa/reset.c | 9 +- arch/arm/mach-pxa/{include/mach => }/reset.h | 2 +- arch/arm/mach-pxa/sharpsl_pm.c | 2 +- arch/arm/mach-pxa/sleep.S | 9 +- arch/arm/mach-pxa/smemc.c | 13 +- arch/arm/mach-pxa/{include/mach => }/smemc.h | 0 arch/arm/mach-pxa/spitz.c | 37 +++- arch/arm/mach-pxa/{include/mach => }/spitz.h | 0 arch/arm/mach-pxa/spitz_pm.c | 3 +- arch/arm/mach-pxa/standby.S | 3 +- arch/arm/mach-pxa/tosa.c | 47 ++++- arch/arm/mach-pxa/{include/mach => }/tosa.h | 0 .../arm/mach-pxa/trizeps4-pcmcia.c | 6 +- arch/arm/mach-pxa/trizeps4.c | 6 +- .../mach-pxa/{include/mach => }/trizeps4.h | 1 + .../arm/mach-pxa/viper-pcmcia.c | 6 +- .../arm/mach-pxa/viper-pcmcia.h | 0 arch/arm/mach-pxa/viper.c | 8 +- .../arm/mach-pxa/vpac270-pcmcia.c | 4 +- arch/arm/mach-pxa/vpac270.c | 4 +- .../arm/mach-pxa/{include/mach => }/vpac270.h | 0 arch/arm/mach-pxa/xcep.c | 4 +- arch/arm/mach-pxa/z2.c | 13 +- arch/arm/mach-pxa/{include/mach => }/z2.h | 0 arch/arm/mach-pxa/zeus.c | 8 +- arch/arm/mach-pxa/zylonite.c | 34 +++- arch/arm/mach-pxa/zylonite.h | 2 + arch/arm/mach-pxa/zylonite_pxa300.c | 1 + arch/arm/mach-pxa/zylonite_pxa320.c | 1 + arch/arm/mach-sa1100/generic.c | 6 +- arch/arm/mach-sa1100/include/mach/reset.h | 1 - arch/arm/mm/copypage-xsc3.c | 2 + arch/mips/alchemy/devboards/db1300.c | 9 - drivers/ata/pata_palmld.c | 3 +- drivers/clk/pxa/clk-pxa.c | 8 +- drivers/clk/pxa/clk-pxa.h | 9 +- drivers/clk/pxa/clk-pxa25x.c | 46 ++--- drivers/clk/pxa/clk-pxa27x.c | 68 +++---- drivers/clk/pxa/clk-pxa2xx.h | 58 ++++++ drivers/clk/pxa/clk-pxa3xx.c | 139 +++++++++++-- drivers/cpufreq/pxa2xx-cpufreq.c | 6 +- drivers/cpufreq/pxa3xx-cpufreq.c | 65 +++--- drivers/input/mouse/pxa930_trkball.c | 1 - drivers/input/touchscreen/Kconfig | 2 + drivers/input/touchscreen/mainstone-wm97xx.c | 130 ++++++------ drivers/input/touchscreen/wm97xx-core.c | 42 +--- drivers/input/touchscreen/zylonite-wm97xx.c | 43 ++-- drivers/leds/leds-locomo.c | 1 - drivers/mmc/host/pxamci.c | 2 +- drivers/mtd/maps/pxa2xx-flash.c | 2 - drivers/pcmcia/Makefile | 13 -- drivers/pcmcia/pxa2xx_base.c | 48 ++--- drivers/pcmcia/pxa2xx_sharpsl.c | 3 +- drivers/pcmcia/sa1111_generic.c | 1 - drivers/pcmcia/sa1111_lubbock.c | 1 - drivers/pcmcia/soc_common.c | 2 - drivers/pcmcia/soc_common.h | 120 +---------- drivers/power/supply/tosa_battery.c | 189 ++++++++++-------- drivers/rtc/rtc-pxa.c | 2 - drivers/soc/Kconfig | 1 + drivers/soc/Makefile | 1 + .../arm/plat-pxa => drivers/soc/pxa}/Kconfig | 5 +- .../arm/plat-pxa => drivers/soc/pxa}/Makefile | 4 - {arch/arm/plat-pxa => drivers/soc/pxa}/mfp.c | 2 +- {arch/arm/plat-pxa => drivers/soc/pxa}/ssp.c | 0 drivers/usb/gadget/udc/pxa25x_udc.c | 37 ++-- drivers/usb/gadget/udc/pxa25x_udc.h | 7 +- drivers/usb/host/ohci-pxa27x.c | 3 +- .../video/fbdev/pxa3xx-regs.h | 24 +-- drivers/video/fbdev/pxafb.c | 4 +- drivers/watchdog/sa1100_wdt.c | 88 +++++--- include/linux/clk/pxa.h | 16 ++ include/linux/platform_data/asoc-poodle.h | 16 ++ .../linux/platform_data/asoc-pxa.h | 4 +- include/linux/platform_data/video-pxafb.h | 22 +- .../hardware.h => include/linux/soc/pxa/cpu.h | 61 +----- .../plat => include/linux/soc/pxa}/mfp.h | 6 +- include/linux/soc/pxa/smemc.h | 13 ++ include/linux/wm97xx.h | 4 - include/pcmcia/soc_common.h | 125 ++++++++++++ include/sound/pxa2xx-lib.h | 4 + sound/arm/pxa2xx-ac97-lib.c | 145 +++++++++----- .../arm/pxa2xx-ac97-regs.h | 42 ++-- sound/arm/pxa2xx-ac97.c | 3 +- sound/soc/pxa/corgi.c | 43 ++-- sound/soc/pxa/e740_wm9705.c | 37 ++-- sound/soc/pxa/e750_wm9705.c | 33 ++- sound/soc/pxa/e800_wm9712.c | 33 ++- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/hx4700.c | 34 ++-- sound/soc/pxa/magician.c | 141 ++++--------- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/pxa/palm27x.c | 2 +- sound/soc/pxa/poodle.c | 51 ++--- sound/soc/pxa/pxa2xx-ac97.c | 24 ++- sound/soc/pxa/pxa2xx-i2s.c | 112 ++++++----- sound/soc/pxa/spitz.c | 58 +++--- sound/soc/pxa/tosa.c | 18 +- sound/soc/pxa/z2.c | 8 +- 213 files changed, 1902 insertions(+), 1936 deletions(-) delete mode 100644 arch/arm/mach-mmp/tavorevb.c delete mode 100644 arch/arm/mach-pxa/Makefile.boot rename arch/arm/mach-pxa/{include/mach => }/addr-map.h (100%) rename drivers/pcmcia/pxa2xx_balloon3.c => arch/arm/mach-pxa/balloon3-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/balloon3.h (100%) rename drivers/pcmcia/pxa2xx_colibri.c => arch/arm/mach-pxa/colibri-pcmcia.c (99%) rename arch/arm/mach-pxa/{include/mach => }/corgi.h (100%) rename drivers/pcmcia/pxa2xx_e740.c => arch/arm/mach-pxa/e740-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/eseries-gpio.h (100%) rename drivers/pcmcia/pxa2xx_hx4700.c => arch/arm/mach-pxa/hx4700-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/hx4700.h (100%) delete mode 100644 arch/arm/mach-pxa/include/mach/bitfield.h delete mode 100644 arch/arm/mach-pxa/include/mach/dma.h delete mode 100644 arch/arm/mach-pxa/include/mach/generic.h delete mode 100644 arch/arm/mach-pxa/include/mach/mtd-xip.h delete mode 100644 arch/arm/mach-pxa/include/mach/uncompress.h rename arch/arm/mach-pxa/{include/mach => }/irqs.h (100%) rename arch/arm/mach-pxa/{include/mach => }/lubbock.h (95%) rename arch/arm/mach-pxa/{include/mach => }/magician.h (99%) rename arch/arm/mach-pxa/{include/mach => }/mainstone.h (98%) rename arch/arm/mach-pxa/{include/mach => }/mfp.h (91%) rename drivers/pcmcia/pxa2xx_palmld.c => arch/arm/mach-pxa/palmld-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/palmld.h (100%) rename drivers/pcmcia/pxa2xx_palmtc.c => arch/arm/mach-pxa/palmtc-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/palmtc.h (100%) rename drivers/pcmcia/pxa2xx_palmtx.c => arch/arm/mach-pxa/palmtx-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/palmtx.h (100%) rename arch/arm/mach-pxa/{include/mach => }/poodle.h (98%) create mode 100644 arch/arm/mach-pxa/pxa-regs.h rename arch/arm/mach-pxa/{include/mach => }/pxa2xx-regs.h (76%) rename arch/arm/mach-pxa/{include/mach => }/pxa3xx-regs.h (61%) rename arch/arm/mach-pxa/{include/mach => }/regs-ost.h (94%) rename arch/arm/mach-pxa/{include/mach => }/regs-uart.h (99%) rename arch/arm/mach-pxa/{include/mach => }/reset.h (92%) rename arch/arm/mach-pxa/{include/mach => }/smemc.h (100%) rename arch/arm/mach-pxa/{include/mach => }/spitz.h (100%) rename arch/arm/mach-pxa/{include/mach => }/tosa.h (100%) rename drivers/pcmcia/pxa2xx_trizeps4.c => arch/arm/mach-pxa/trizeps4-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/trizeps4.h (99%) rename drivers/pcmcia/pxa2xx_viper.c => arch/arm/mach-pxa/viper-pcmcia.c (97%) rename include/linux/platform_data/pcmcia-pxa2xx_viper.h => arch/arm/mach-pxa/viper-pcmcia.h (100%) rename drivers/pcmcia/pxa2xx_vpac270.c => arch/arm/mach-pxa/vpac270-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/vpac270.h (100%) rename arch/arm/mach-pxa/{include/mach => }/z2.h (100%) create mode 100644 drivers/clk/pxa/clk-pxa2xx.h rename {arch/arm/plat-pxa => drivers/soc/pxa}/Kconfig (83%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/Makefile (51%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/mfp.c (99%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/ssp.c (100%) rename arch/arm/mach-pxa/include/mach/regs-lcd.h => drivers/video/fbdev/pxa3xx-regs.h (90%) create mode 100644 include/linux/clk/pxa.h create mode 100644 include/linux/platform_data/asoc-poodle.h rename arch/arm/mach-pxa/include/mach/audio.h => include/linux/platform_data/asoc-pxa.h (93%) rename arch/arm/mach-pxa/include/mach/hardware.h => include/linux/soc/pxa/cpu.h (75%) rename {arch/arm/plat-pxa/include/plat => include/linux/soc/pxa}/mfp.h (98%) create mode 100644 include/linux/soc/pxa/smemc.h create mode 100644 include/pcmcia/soc_common.h rename arch/arm/mach-pxa/include/mach/regs-ac97.h => sound/arm/pxa2xx-ac97-regs.h (71%) -- 2.29.2 Cc: Daniel Mack <daniel(a)zonque.org> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: Marek Vasut <marek.vasut(a)gmail.com> Cc: Philipp Zabel <philipp.zabel(a)gmail.com> Cc: Lubomir Rintel <lkundrak(a)v3.sk> Cc: Paul Parsons <lost.distance(a)yahoo.com> Cc: Tomas Cech <sleep_walker(a)suse.com> Cc: Sergey Lapin <slapin(a)ossfans.org> Cc: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> Cc: Michael Turquette <mturquette(a)baylibre.com> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: "Rafael J. Wysocki" <rafael(a)kernel.org> Cc: Viresh Kumar <viresh.kumar(a)linaro.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Ulf Hansson <ulf.hansson(a)linaro.org> Cc: Dominik Brodowski <linux(a)dominikbrodowski.net> Cc: Helge Deller <deller(a)gmx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-mips(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: linux-clk(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: patches(a)opensource.cirrus.com Cc: linux-leds(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: alsa-devel(a)alsa-project.org

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[PATCH -next] ASoC: mediatek: mt8195: Fix build warning without CONFIG_OF
by YueHaibing 07 May '22

07 May '22

sound/soc/mediatek/mt8195/mt8195-mt6359.c:1639:32: warning: ‘mt8195_mt6359_max98390_rt5682_card’ defined but not used [-Wunused-variable] 1639 | static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/mt8195/mt8195-mt6359.c:1634:32: warning: ‘mt8195_mt6359_rt1011_rt5682_card’ defined but not used [-Wunused-variable] 1634 | static struct mt8195_card_data mt8195_mt6359_rt1011_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/mt8195/mt8195-mt6359.c:1629:32: warning: ‘mt8195_mt6359_rt1019_rt5682_card’ defined but not used [-Wunused-variable] 1629 | static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ These variables is only used with CONFIG_OF, move it into the ifdef block. Fixes: 86a6b9c9dfff ("ASoC: mediatek: mt8195: add machine support for max98390 and rt5682") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 3e32fe801b3c..f90675f14d60 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -1626,6 +1626,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) return ret; } +#ifdef CONFIG_OF static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = { .name = "mt8195_r1019_5682", .quirk = RT1019_SPEAKER_AMP_PRESENT, @@ -1641,7 +1642,6 @@ static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = { .quirk = MAX98390_SPEAKER_AMP_PRESENT, }; -#ifdef CONFIG_OF static const struct of_device_id mt8195_mt6359_dt_match[] = { { .compatible = "mediatek,mt8195_mt6359_rt1019_rt5682", -- 2.17.1

2 2

Re: [PATCH] kernel/drivers: Remove redundant driver match function
by Finn Thain 07 May '22

07 May '22

On Thu, 5 May 2022, lizhe wrote: > If there is no driver match function, the driver core assumes that each > candidate pair (driver, device) matches, see driver_match_device(). > > Signed-off-by: lizhe <sensor1010(a)163.com> Acked-by: Finn Thain <fthain(a)linux-m68k.org> for the nubus changes. > --- > drivers/mfd/mcp-core.c | 6 ------ > drivers/nubus/bus.c | 6 ------ > drivers/s390/crypto/vfio_ap_drv.c | 6 ------ > drivers/scsi/scsi_debug.c | 7 ------- > drivers/target/loopback/tcm_loop.c | 7 ------- > drivers/w1/w1.c | 6 ------ > sound/ac97_bus.c | 11 ----------- > 7 files changed, 49 deletions(-) > > diff --git a/drivers/mfd/mcp-core.c b/drivers/mfd/mcp-core.c > index 2fa592c37c6f..281a9369f2b3 100644 > --- a/drivers/mfd/mcp-core.c > +++ b/drivers/mfd/mcp-core.c > @@ -20,11 +20,6 @@ > #define to_mcp(d) container_of(d, struct mcp, attached_device) > #define to_mcp_driver(d) container_of(d, struct mcp_driver, drv) > > -static int mcp_bus_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > static int mcp_bus_probe(struct device *dev) > { > struct mcp *mcp = to_mcp(dev); > @@ -43,7 +38,6 @@ static void mcp_bus_remove(struct device *dev) > > static struct bus_type mcp_bus_type = { > .name = "mcp", > - .match = mcp_bus_match, > .probe = mcp_bus_probe, > .remove = mcp_bus_remove, > }; > diff --git a/drivers/nubus/bus.c b/drivers/nubus/bus.c > index 17fad660032c..72921e4f35f6 100644 > --- a/drivers/nubus/bus.c > +++ b/drivers/nubus/bus.c > @@ -14,11 +14,6 @@ > #define to_nubus_board(d) container_of(d, struct nubus_board, dev) > #define to_nubus_driver(d) container_of(d, struct nubus_driver, driver) > > -static int nubus_bus_match(struct device *dev, struct device_driver *driver) > -{ > - return 1; > -} > - > static int nubus_device_probe(struct device *dev) > { > struct nubus_driver *ndrv = to_nubus_driver(dev->driver); > @@ -39,7 +34,6 @@ static void nubus_device_remove(struct device *dev) > > struct bus_type nubus_bus_type = { > .name = "nubus", > - .match = nubus_bus_match, > .probe = nubus_device_probe, > .remove = nubus_device_remove, > }; > diff --git a/drivers/s390/crypto/vfio_ap_drv.c b/drivers/s390/crypto/vfio_ap_drv.c > index 29ebd54f8919..0a662c451f2a 100644 > --- a/drivers/s390/crypto/vfio_ap_drv.c > +++ b/drivers/s390/crypto/vfio_ap_drv.c > @@ -172,14 +172,8 @@ static void vfio_ap_matrix_dev_release(struct device *dev) > kfree(matrix_dev); > } > > -static int matrix_bus_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > static struct bus_type matrix_bus = { > .name = "matrix", > - .match = &matrix_bus_match, > }; > > static struct device_driver matrix_driver = { > diff --git a/drivers/scsi/scsi_debug.c b/drivers/scsi/scsi_debug.c > index 592a290e6cfa..8107489b36e8 100644 > --- a/drivers/scsi/scsi_debug.c > +++ b/drivers/scsi/scsi_debug.c > @@ -7844,15 +7844,8 @@ static void sdebug_driver_remove(struct device *dev) > scsi_host_put(sdbg_host->shost); > } > > -static int pseudo_lld_bus_match(struct device *dev, > - struct device_driver *dev_driver) > -{ > - return 1; > -} > - > static struct bus_type pseudo_lld_bus = { > .name = "pseudo", > - .match = pseudo_lld_bus_match, > .probe = sdebug_driver_probe, > .remove = sdebug_driver_remove, > .drv_groups = sdebug_drv_groups, > diff --git a/drivers/target/loopback/tcm_loop.c b/drivers/target/loopback/tcm_loop.c > index 4407b56aa6d1..eeb63deff94f 100644 > --- a/drivers/target/loopback/tcm_loop.c > +++ b/drivers/target/loopback/tcm_loop.c > @@ -83,15 +83,8 @@ static int tcm_loop_show_info(struct seq_file *m, struct Scsi_Host *host) > static int tcm_loop_driver_probe(struct device *); > static void tcm_loop_driver_remove(struct device *); > > -static int pseudo_lld_bus_match(struct device *dev, > - struct device_driver *dev_driver) > -{ > - return 1; > -} > - > static struct bus_type tcm_loop_lld_bus = { > .name = "tcm_loop_bus", > - .match = pseudo_lld_bus_match, > .probe = tcm_loop_driver_probe, > .remove = tcm_loop_driver_remove, > }; > diff --git a/drivers/w1/w1.c b/drivers/w1/w1.c > index f2ae2e563dc5..a6ecfa1b3417 100644 > --- a/drivers/w1/w1.c > +++ b/drivers/w1/w1.c > @@ -58,11 +58,6 @@ MODULE_PARM_DESC(slave_ttl, > DEFINE_MUTEX(w1_mlock); > LIST_HEAD(w1_masters); > > -static int w1_master_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > static int w1_master_probe(struct device *dev) > { > return -ENODEV; > @@ -174,7 +169,6 @@ static int w1_uevent(struct device *dev, struct kobj_uevent_env *env); > > static struct bus_type w1_bus_type = { > .name = "w1", > - .match = w1_master_match, > .uevent = w1_uevent, > }; > > diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c > index b4685c53ff11..c7aee8c42c55 100644 > --- a/sound/ac97_bus.c > +++ b/sound/ac97_bus.c > @@ -75,19 +75,8 @@ int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, > } > EXPORT_SYMBOL_GPL(snd_ac97_reset); > > -/* > - * Let drivers decide whether they want to support given codec from their > - * probe method. Drivers have direct access to the struct snd_ac97 > - * structure and may decide based on the id field amongst other things. > - */ > -static int ac97_bus_match(struct device *dev, struct device_driver *drv) > -{ > - return 1; > -} > - > struct bus_type ac97_bus_type = { > .name = "ac97", > - .match = ac97_bus_match, > }; > > static int __init ac97_bus_init(void) >

1 0

Lenovo ThinkPad T14 AMD - Broken headphone detection
by GitHub issues - opened 06 May '22

06 May '22

alsa-project/alsa-ucm-conf issue #152 was opened from xsmile: ### Summary Booting the system with headphones plugged in results in a broken volume control. Sometimes the headphones are not recognized at all. ### Environment - Lenovo ThinkPad T14 Gen 1 (AMD), BIOS 1.40 - Onboard Realtek ALC257 ### Steps to reproduce - plug in headphones - boot the system - play back audio and try changing the volume ### What is the current *bug* behavior? The previously set volume level is ignored and set to the maximum, despite PulseAudio telling otherwise. Changing the volume has no effect. Occasionally, headphone presence is not recognized after a reboot instead and audio is played back through the speakers. ### What is the expected *correct* behavior? The volume level is persisted between reboots, headphones are recognized reliably and volume control is functional. ### Additional Information As a workaround, headphones can be removed and plugged in again, which restores the normal volume level and the ability to control volume. The bug does not happen when using ALSA without PulseAudio or when using PulseAudio without UCM. Here is a series of `pa-info` log files where volume control was functional but both speakers and headphones were active after the reboot. [1-speakers.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8643462… headphones are not plugged in; speakers are active [2-headphones.txt](https://github.com/alsa-project/alsa-ucm-conf/files/86434… headphones were plugged in; headphones are active and speakers are muted [3-reboot-with-headphones.txt](https://github.com/alsa-project/alsa-ucm-conf… rebooted the notebook with headphones plugged in; both speakers and headphones are active, while speakers should have been muted; volume control changes speaker instead of headphone volume [4-speakers.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8643465… removed headphones; speakers are active [5-headphones.txt](https://github.com/alsa-project/alsa-ucm-conf/files/86434… plugged in headphones; headphones are active and speakers are muted --- Related issue: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1356 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/152 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/14] ASoC: Intel: avs: Driver core and PCM operations
by Cezary Rojewski 06 May '22

06 May '22

Part three of main AVS driver series. This series was originally part of the initial series which was later divided [1] into smaller, easier-to-review chunks. Thus, many patches found here were already present on the list. This series consists of code typical to many drivers - PCI driver operations, trace ability, PM operations - as well as PCM handlers for all standard audio interfaces, that is, HDA, I2S and DMIC are found here. Series starts with updating firmware boot flow - libraries are no longer ignored. This change is dependent on already merged topology code [2] and because of that could not be part of the initial series [1]. PCM operations are split into four changes. First component operations alone i.e. operations which are usually agnostic towards path position (FE/BE). Then it continues with "generic" FE operations - there is no interface split here as from Intel ADSP point of view, FE, or HOST side as it's called in the specs, involves HD-Audio operations only. BE (also known as LINK) side on the other hand is divided into "non-HD-Audio" and HD-Audio part. The former represents transfer over DMIC and I2S interfaces both. While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: - event tracing - preparation for firmware tracing (debugability) - coredump (debugability) - recovery flow (attempt recovery after IPC timeout or exception) - D0ix (D0 device substate, complements standard power management) Series is finalized by actual addition of supported platforms: SKL and APL-based. Platform-specific files are limited to firmware-specific bits, that is, bits that are specific to given firmware generation. Everything else is shared and is part of already upstream messaging code found in ipc.c, messages.c and messages.h files. [1]: https://lore.kernel.org/all/20220311153544.136854-1-cezary.rojewski@intel.c… [2]: https://lore.kernel.org/all/20220331135246.993089-1-cezary.rojewski@intel.c… Cezary Rojewski (14): ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support include/sound/soc-acpi.h | 2 + sound/soc/intel/Kconfig | 4 +- sound/soc/intel/avs/Makefile | 7 +- sound/soc/intel/avs/apl.c | 250 ++++++ sound/soc/intel/avs/avs.h | 79 ++ sound/soc/intel/avs/board_selection.c | 463 ++++++++++ sound/soc/intel/avs/core.c | 655 ++++++++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 249 +++++- sound/soc/intel/avs/loader.c | 83 ++ sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 ++ sound/soc/intel/avs/pcm.c | 1182 +++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 +++ sound/soc/intel/avs/topology.c | 2 - sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 158 ++++ sound/soc/intel/avs/utils.c | 23 + 19 files changed, 3421 insertions(+), 15 deletions(-) create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h -- 2.25.1

5 39

[RESEND PATCH v7 0/1] firmware: mtk: add adsp ipc protocol for SOF
by Tinghan Shen 06 May '22

06 May '22

This patch provides mediatek adsp ipc support for SOF. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This patch was tested and confirmed to work with SOF fw on MT8195 cherry board. changes since v6: - rebase to matthias.bgg/linux.git, v5.18-next/soc - Prefer "GPL" over "GPL v2" for MODULE_LICENSE changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free TingHan Shen (1): firmware: mediatek: add adsp ipc protocol interface drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 161 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 6 files changed, 239 insertions(+) create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 5

RE: [PATCH 02/32] Introduce flexible array struct memcpy() helpers
by David Laight 06 May '22

06 May '22

From: Johannes Berg > Sent: 05 May 2022 21:13 > On Thu, 2022-05-05 at 13:08 -0700, Keith Packard wrote: > > > > I bet you've already considered the simpler form: > > > > struct something *instance = mem_to_flex_dup(byte_array, count, GFP_KERNEL); > > if (IS_ERR(instance)) > > return PTR_ERR(instance); > > > > Sadly, this doesn't work in any way because mem_to_flex_dup() needs to > know at least the type, hence passing 'instance', which is simpler than > passing 'struct something'. You can use: struct something *instance; mem_to_flex_dup(instance, byte_array, count, GFP_KERNEL); if (IS_ERR(instance)) return PTR_ERR(instance); and have mem_to_flex_dup() (which must be a #define) update 'instance'. (You can require &instance - and just precede all the uses with an extra '*' to make it more obvious the variable is updated. But there is little point requiring it be NULL.) If you really want to define the variable mid-block you can use: mem_to_flex_dup(struct something *, instance, byte_array, count, GFP_KERNEL); but I really hate having declarations anywhere other than the top of a function because it makes them hard for the 'mk1 eyeball' to spot. David - Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales)

1 0

[PATCH 0/3] ASoC: SOF: IPC4: Introduce message handling functionality
by Peter Ujfalusi 05 May '22

05 May '22

Hi, The series adds the basic IPC4 message handling code, implementing the ipc callbacks. Due to the difference between IPC3 and IPC4 messaging we need to introduce new message container for IPC4, but the SOF internal callbacks and structures can be kept as they were and leaving it for the IPC specific code to handle the differences. The series provides the foundation for both lowe level (sound/soc/sof/intel) and high level IPC4 implementation (topologies, firmware loading, control handling, etc). Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Add rx_data pointer to snd_sof_ipc_msg struct ASoC: SOF: Add initial header file for ipc4 ASoC: SOF: ipc4: Add support for mandatory message handling functionality include/sound/sof/ipc4/header.h | 460 ++++++++++++++++++++++++ sound/soc/sof/Makefile | 3 +- sound/soc/sof/ipc4.c | 604 ++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 4 + 4 files changed, 1070 insertions(+), 1 deletion(-) create mode 100644 include/sound/sof/ipc4/header.h create mode 100644 sound/soc/sof/ipc4.c -- 2.36.0

2 4

[PATCH v2 RESEND] ASoC: ti: davinci-mcasp: Add dma-type for bcdma
by Jai Luthra 05 May '22

05 May '22

From: Jayesh Choudhary <j-choudhary(a)ti.com> Set DMA type for ti-bcdma controller for AM62-SK. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- Resending the ack-ed patch, as I missed cc'ing all maintainers. sound/soc/ti/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 2c146b91fca3..377be2e2b6ee 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2047,6 +2047,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) return PCM_SDMA; else if (strstr(tmp, "udmap")) return PCM_UDMA; + else if (strstr(tmp, "bcdma")) + return PCM_UDMA; return PCM_EDMA; } -- 2.17.1

2 1

[PATCH] ASoC: fsl_micfil: fix the naming style for mask definition
by Shengjiu Wang 05 May '22

05 May '22

Remove the _SHIFT for the mask definition. Fixes: 17f2142bae4b ("ASoC: fsl_micfil: use GENMASK to define register bit fields") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_micfil.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index 08901827047d..053caba3caf3 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -74,9 +74,9 @@ #define MICFIL_FIFO_STAT_FIFOX_UNDER(ch) BIT((ch) + 8) /* MICFIL HWVAD0 Control 1 Register -- REG_MICFIL_VAD0_CTRL1*/ -#define MICFIL_VAD0_CTRL1_CHSEL_SHIFT GENMASK(26, 24) -#define MICFIL_VAD0_CTRL1_CICOSR_SHIFT GENMASK(19, 16) -#define MICFIL_VAD0_CTRL1_INITT_SHIFT GENMASK(12, 8) +#define MICFIL_VAD0_CTRL1_CHSEL GENMASK(26, 24) +#define MICFIL_VAD0_CTRL1_CICOSR GENMASK(19, 16) +#define MICFIL_VAD0_CTRL1_INITT GENMASK(12, 8) #define MICFIL_VAD0_CTRL1_ST10 BIT(4) #define MICFIL_VAD0_CTRL1_ERIE BIT(3) #define MICFIL_VAD0_CTRL1_IE BIT(2) @@ -106,7 +106,7 @@ /* MICFIL HWVAD0 Zero-Crossing Detector - REG_MICFIL_VAD0_ZCD */ #define MICFIL_VAD0_ZCD_ZCDTH GENMASK(25, 16) -#define MICFIL_VAD0_ZCD_ZCDADJ_SHIFT GENMASK(11, 8) +#define MICFIL_VAD0_ZCD_ZCDADJ GENMASK(11, 8) #define MICFIL_VAD0_ZCD_ZCDAND BIT(4) #define MICFIL_VAD0_ZCD_ZCDAUT BIT(2) #define MICFIL_VAD0_ZCD_ZCDEN BIT(0) -- 2.17.1

3 2

[PATCH v1 1/1] ASoC: cs43130: Re-use generic struct u16_fract
by Andy Shevchenko 05 May '22

05 May '22

Instead of custom data type re-use generic struct u16_fract. No changes intended. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/codecs/cs43130.c | 16 ++-- sound/soc/codecs/cs43130.h | 151 +++++++++++++++++++------------------ 2 files changed, 84 insertions(+), 83 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 04548b577ad5..a2bce0f9f247 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -712,30 +712,30 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, case CS43130_ASP_PCM_DAI: case CS43130_ASP_DOP_DAI: regmap_write(cs43130->regmap, CS43130_ASP_DEN_1, - (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >> + (clk_gen->v.denominator & CS43130_SP_M_LSB_DATA_MASK) >> CS43130_SP_M_LSB_DATA_SHIFT); regmap_write(cs43130->regmap, CS43130_ASP_DEN_2, - (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >> + (clk_gen->v.denominator & CS43130_SP_M_MSB_DATA_MASK) >> CS43130_SP_M_MSB_DATA_SHIFT); regmap_write(cs43130->regmap, CS43130_ASP_NUM_1, - (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >> + (clk_gen->v.numerator & CS43130_SP_N_LSB_DATA_MASK) >> CS43130_SP_N_LSB_DATA_SHIFT); regmap_write(cs43130->regmap, CS43130_ASP_NUM_2, - (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >> + (clk_gen->v.numerator & CS43130_SP_N_MSB_DATA_MASK) >> CS43130_SP_N_MSB_DATA_SHIFT); break; case CS43130_XSP_DOP_DAI: regmap_write(cs43130->regmap, CS43130_XSP_DEN_1, - (clk_gen->den & CS43130_SP_M_LSB_DATA_MASK) >> + (clk_gen->v.denominator & CS43130_SP_M_LSB_DATA_MASK) >> CS43130_SP_M_LSB_DATA_SHIFT); regmap_write(cs43130->regmap, CS43130_XSP_DEN_2, - (clk_gen->den & CS43130_SP_M_MSB_DATA_MASK) >> + (clk_gen->v.denominator & CS43130_SP_M_MSB_DATA_MASK) >> CS43130_SP_M_MSB_DATA_SHIFT); regmap_write(cs43130->regmap, CS43130_XSP_NUM_1, - (clk_gen->num & CS43130_SP_N_LSB_DATA_MASK) >> + (clk_gen->v.numerator & CS43130_SP_N_LSB_DATA_MASK) >> CS43130_SP_N_LSB_DATA_SHIFT); regmap_write(cs43130->regmap, CS43130_XSP_NUM_2, - (clk_gen->num & CS43130_SP_N_MSB_DATA_MASK) >> + (clk_gen->v.numerator & CS43130_SP_N_MSB_DATA_MASK) >> CS43130_SP_N_MSB_DATA_SHIFT); break; default: diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index e62d671e95bb..1dd893674313 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -10,6 +10,8 @@ #ifndef __CS43130_H__ #define __CS43130_H__ +#include <linux/math.h> + /* CS43130 registers addresses */ /* all reg address is shifted by a byte for control byte to be LSB */ #define CS43130_FIRSTREG 0x010000 @@ -372,97 +374,96 @@ enum cs43130_dai_id { }; struct cs43130_clk_gen { - unsigned int mclk_int; - int fs; - u16 den; - u16 num; + unsigned int mclk_int; + int fs; + struct u16_fract v; }; /* frm_size = 16 */ static const struct cs43130_clk_gen cs43130_16_clk_gen[] = { - {22579200, 32000, 441, 10,}, - {22579200, 44100, 32, 1,}, - {22579200, 48000, 147, 5,}, - {22579200, 88200, 16, 1,}, - {22579200, 96000, 147, 10,}, - {22579200, 176400, 8, 1,}, - {22579200, 192000, 147, 20,}, - {22579200, 352800, 4, 1,}, - {22579200, 384000, 147, 40,}, - {24576000, 32000, 48, 1,}, - {24576000, 44100, 5120, 147,}, - {24576000, 48000, 32, 1,}, - {24576000, 88200, 2560, 147,}, - {24576000, 96000, 16, 1,}, - {24576000, 176400, 1280, 147,}, - {24576000, 192000, 8, 1,}, - {24576000, 352800, 640, 147,}, - {24576000, 384000, 4, 1,}, + { 22579200, 32000, .v = { 441, 10, }, }, + { 22579200, 44100, .v = { 32, 1, }, }, + { 22579200, 48000, .v = { 147, 5, }, }, + { 22579200, 88200, .v = { 16, 1, }, }, + { 22579200, 96000, .v = { 147, 10, }, }, + { 22579200, 176400, .v = { 8, 1, }, }, + { 22579200, 192000, .v = { 147, 20, }, }, + { 22579200, 352800, .v = { 4, 1, }, }, + { 22579200, 384000, .v = { 147, 40, }, }, + { 24576000, 32000, .v = { 48, 1, }, }, + { 24576000, 44100, .v = { 5120, 147, }, }, + { 24576000, 48000, .v = { 32, 1, }, }, + { 24576000, 88200, .v = { 2560, 147, }, }, + { 24576000, 96000, .v = { 16, 1, }, }, + { 24576000, 176400, .v = { 1280, 147, }, }, + { 24576000, 192000, .v = { 8, 1, }, }, + { 24576000, 352800, .v = { 640, 147, }, }, + { 24576000, 384000, .v = { 4, 1, }, }, }; /* frm_size = 32 */ static const struct cs43130_clk_gen cs43130_32_clk_gen[] = { - {22579200, 32000, 441, 20,}, - {22579200, 44100, 16, 1,}, - {22579200, 48000, 147, 10,}, - {22579200, 88200, 8, 1,}, - {22579200, 96000, 147, 20,}, - {22579200, 176400, 4, 1,}, - {22579200, 192000, 147, 40,}, - {22579200, 352800, 2, 1,}, - {22579200, 384000, 147, 80,}, - {24576000, 32000, 24, 1,}, - {24576000, 44100, 2560, 147,}, - {24576000, 48000, 16, 1,}, - {24576000, 88200, 1280, 147,}, - {24576000, 96000, 8, 1,}, - {24576000, 176400, 640, 147,}, - {24576000, 192000, 4, 1,}, - {24576000, 352800, 320, 147,}, - {24576000, 384000, 2, 1,}, + { 22579200, 32000, .v = { 441, 20, }, }, + { 22579200, 44100, .v = { 16, 1, }, }, + { 22579200, 48000, .v = { 147, 10, }, }, + { 22579200, 88200, .v = { 8, 1, }, }, + { 22579200, 96000, .v = { 147, 20, }, }, + { 22579200, 176400, .v = { 4, 1, }, }, + { 22579200, 192000, .v = { 147, 40, }, }, + { 22579200, 352800, .v = { 2, 1, }, }, + { 22579200, 384000, .v = { 147, 80, }, }, + { 24576000, 32000, .v = { 24, 1, }, }, + { 24576000, 44100, .v = { 2560, 147, }, }, + { 24576000, 48000, .v = { 16, 1, }, }, + { 24576000, 88200, .v = { 1280, 147, }, }, + { 24576000, 96000, .v = { 8, 1, }, }, + { 24576000, 176400, .v = { 640, 147, }, }, + { 24576000, 192000, .v = { 4, 1, }, }, + { 24576000, 352800, .v = { 320, 147, }, }, + { 24576000, 384000, .v = { 2, 1, }, }, }; /* frm_size = 48 */ static const struct cs43130_clk_gen cs43130_48_clk_gen[] = { - {22579200, 32000, 147, 100,}, - {22579200, 44100, 32, 3,}, - {22579200, 48000, 49, 5,}, - {22579200, 88200, 16, 3,}, - {22579200, 96000, 49, 10,}, - {22579200, 176400, 8, 3,}, - {22579200, 192000, 49, 20,}, - {22579200, 352800, 4, 3,}, - {22579200, 384000, 49, 40,}, - {24576000, 32000, 16, 1,}, - {24576000, 44100, 5120, 441,}, - {24576000, 48000, 32, 3,}, - {24576000, 88200, 2560, 441,}, - {24576000, 96000, 16, 3,}, - {24576000, 176400, 1280, 441,}, - {24576000, 192000, 8, 3,}, - {24576000, 352800, 640, 441,}, - {24576000, 384000, 4, 3,}, + { 22579200, 32000, .v = { 147, 100, }, }, + { 22579200, 44100, .v = { 32, 3, }, }, + { 22579200, 48000, .v = { 49, 5, }, }, + { 22579200, 88200, .v = { 16, 3, }, }, + { 22579200, 96000, .v = { 49, 10, }, }, + { 22579200, 176400, .v = { 8, 3, }, }, + { 22579200, 192000, .v = { 49, 20, }, }, + { 22579200, 352800, .v = { 4, 3, }, }, + { 22579200, 384000, .v = { 49, 40, }, }, + { 24576000, 32000, .v = { 16, 1, }, }, + { 24576000, 44100, .v = { 5120, 441, }, }, + { 24576000, 48000, .v = { 32, 3, }, }, + { 24576000, 88200, .v = { 2560, 441, }, }, + { 24576000, 96000, .v = { 16, 3, }, }, + { 24576000, 176400, .v = { 1280, 441, }, }, + { 24576000, 192000, .v = { 8, 3, }, }, + { 24576000, 352800, .v = { 640, 441, }, }, + { 24576000, 384000, .v = { 4, 3, }, }, }; /* frm_size = 64 */ static const struct cs43130_clk_gen cs43130_64_clk_gen[] = { - {22579200, 32000, 441, 40,}, - {22579200, 44100, 8, 1,}, - {22579200, 48000, 147, 20,}, - {22579200, 88200, 4, 1,}, - {22579200, 96000, 147, 40,}, - {22579200, 176400, 2, 1,}, - {22579200, 192000, 147, 80,}, - {22579200, 352800, 1, 1,}, - {24576000, 32000, 12, 1,}, - {24576000, 44100, 1280, 147,}, - {24576000, 48000, 8, 1,}, - {24576000, 88200, 640, 147,}, - {24576000, 96000, 4, 1,}, - {24576000, 176400, 320, 147,}, - {24576000, 192000, 2, 1,}, - {24576000, 352800, 160, 147,}, - {24576000, 384000, 1, 1,}, + { 22579200, 32000, .v = { 441, 40, }, }, + { 22579200, 44100, .v = { 8, 1, }, }, + { 22579200, 48000, .v = { 147, 20, }, }, + { 22579200, 88200, .v = { 4, 1, }, }, + { 22579200, 96000, .v = { 147, 40, }, }, + { 22579200, 176400, .v = { 2, 1, }, }, + { 22579200, 192000, .v = { 147, 80, }, }, + { 22579200, 352800, .v = { 1, 1, }, }, + { 24576000, 32000, .v = { 12, 1, }, }, + { 24576000, 44100, .v = { 1280, 147, }, }, + { 24576000, 48000, .v = { 8, 1, }, }, + { 24576000, 88200, .v = { 640, 147, }, }, + { 24576000, 96000, .v = { 4, 1, }, }, + { 24576000, 176400, .v = { 320, 147, }, }, + { 24576000, 192000, .v = { 2, 1, }, }, + { 24576000, 352800, .v = { 160, 147, }, }, + { 24576000, 384000, .v = { 1, 1, }, }, }; struct cs43130_bitwidth_map { -- 2.35.1

3 2

[PATCH v3 0/6] Add driver for SAMA7G5's PDMC
by Codrin Ciubotariu 05 May '22

05 May '22

This patch series adds support for Pulse Density Microphone Controller (PDMC), present on Microchip's SAMA7G5. The PDMC interfaces up to 4 digital microphones having Pulse Density Modulated (PDM) outputs. It generates a single clock line and samples 1 or 2 data lines. The signal path includes an audio grade programmable decimation filter and outputs 24-bit audio words. The source of each channel can be independently defined as PDMC_DS0 or PDMC_DS1, sampled at the rising or falling edge of PDMC_CLK. The patch series starts with a fix on the ASoC DMA engine support. Then continues with the bindings and the driver of PDMC. It is followed by the DT nodes for SAMA7G5 and SAMA7G5-EK. In the end, the drivers for PDMC and PDM microphones are enabled in sama7_defconfig. Changes in v3: - addressed new DT bindings comments from Krzysztof Kozlowski Changes in v2: - addressed DT bindings comments from Krzysztof Kozlowski Codrin Ciubotariu (6): ASoC: dmaengine: do not use a NULL prepare_slave_config() callback ASoC: dt-bindings: Document Microchip's PDMC ASoC: atmel: mchp-pdmc: add PDMC driver ARM: dts: at91: sama7g5: add nodes for PDMC ARM: dts: at91: sama7g5ek: add node for PDMC0 ARM: configs: at91: sama7_defconfig: add MCHP PDMC and DMIC drivers .../bindings/sound/microchip,pdmc.yaml | 100 ++ arch/arm/boot/dts/at91-sama7g5ek.dts | 21 +- arch/arm/boot/dts/sama7g5.dtsi | 24 + arch/arm/configs/sama7_defconfig | 2 + include/dt-bindings/sound/microchip,pdmc.h | 13 + sound/soc/atmel/Kconfig | 16 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-pdmc.c | 1084 +++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +- 9 files changed, 1264 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/microchip,pdmc.yaml create mode 100644 include/dt-bindings/sound/microchip,pdmc.h create mode 100644 sound/soc/atmel/mchp-pdmc.c -- 2.32.0

6 16

[PATCH v3 20/21] xen/sndfront: use xenbus_setup_ring() and xenbus_teardown_ring()
by Juergen Gross 05 May '22

05 May '22

Simplify sndfront's ring creation and removal via xenbus_setup_ring() and xenbus_teardown_ring(). Signed-off-by: Juergen Gross <jgross(a)suse.com> --- sound/xen/xen_snd_front_evtchnl.c | 44 +++++++------------------------ 1 file changed, 10 insertions(+), 34 deletions(-) diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c index 3e21369c8216..26d1b3987887 100644 --- a/sound/xen/xen_snd_front_evtchnl.c +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -143,12 +143,12 @@ void xen_snd_front_evtchnl_flush(struct xen_snd_front_evtchnl *channel) static void evtchnl_free(struct xen_snd_front_info *front_info, struct xen_snd_front_evtchnl *channel) { - unsigned long page = 0; + void *page = NULL; if (channel->type == EVTCHNL_TYPE_REQ) - page = (unsigned long)channel->u.req.ring.sring; + page = channel->u.req.ring.sring; else if (channel->type == EVTCHNL_TYPE_EVT) - page = (unsigned long)channel->u.evt.page; + page = channel->u.evt.page; if (!page) return; @@ -167,10 +167,7 @@ static void evtchnl_free(struct xen_snd_front_info *front_info, xenbus_free_evtchn(front_info->xb_dev, channel->port); /* End access and free the page. */ - if (channel->gref != INVALID_GRANT_REF) - gnttab_end_foreign_access(channel->gref, page); - else - free_page(page); + xenbus_teardown_ring(&page, 1, &channel->gref); memset(channel, 0, sizeof(*channel)); } @@ -196,8 +193,7 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, enum xen_snd_front_evtchnl_type type) { struct xenbus_device *xb_dev = front_info->xb_dev; - unsigned long page; - grant_ref_t gref; + void *page; irq_handler_t handler; char *handler_name = NULL; int ret; @@ -207,12 +203,9 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, channel->index = index; channel->front_info = front_info; channel->state = EVTCHNL_STATE_DISCONNECTED; - channel->gref = INVALID_GRANT_REF; - page = get_zeroed_page(GFP_KERNEL); - if (!page) { - ret = -ENOMEM; + ret = xenbus_setup_ring(xb_dev, GFP_KERNEL, &page, 1, &channel->gref); + if (ret) goto fail; - } handler_name = kasprintf(GFP_KERNEL, "%s-%s", XENSND_DRIVER_NAME, type == EVTCHNL_TYPE_REQ ? @@ -226,33 +219,18 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, mutex_init(&channel->ring_io_lock); if (type == EVTCHNL_TYPE_REQ) { - struct xen_sndif_sring *sring = (struct xen_sndif_sring *)page; + struct xen_sndif_sring *sring = page; init_completion(&channel->u.req.completion); mutex_init(&channel->u.req.req_io_lock); - SHARED_RING_INIT(sring); - FRONT_RING_INIT(&channel->u.req.ring, sring, XEN_PAGE_SIZE); - - ret = xenbus_grant_ring(xb_dev, sring, 1, &gref); - if (ret < 0) { - channel->u.req.ring.sring = NULL; - goto fail; - } + XEN_FRONT_RING_INIT(&channel->u.req.ring, sring, XEN_PAGE_SIZE); handler = evtchnl_interrupt_req; } else { - ret = gnttab_grant_foreign_access(xb_dev->otherend_id, - virt_to_gfn((void *)page), 0); - if (ret < 0) - goto fail; - - channel->u.evt.page = (struct xensnd_event_page *)page; - gref = ret; + channel->u.evt.page = page; handler = evtchnl_interrupt_evt; } - channel->gref = gref; - ret = xenbus_alloc_evtchn(xb_dev, &channel->port); if (ret < 0) goto fail; @@ -279,8 +257,6 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, return 0; fail: - if (page) - free_page(page); kfree(handler_name); dev_err(&xb_dev->dev, "Failed to allocate ring: %d\n", ret); return ret; -- 2.35.3

1 0

[PATCH v3 08/21] xen/sound: switch xen_snd_front to use INVALID_GRANT_REF
by Juergen Gross 05 May '22

05 May '22

Instead of using a private macro for an invalid grant reference use the common one. Signed-off-by: Juergen Gross <jgross(a)suse.com> --- sound/xen/xen_snd_front_evtchnl.c | 4 ++-- sound/xen/xen_snd_front_evtchnl.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c index ecbc294fc59a..3e21369c8216 100644 --- a/sound/xen/xen_snd_front_evtchnl.c +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -167,7 +167,7 @@ static void evtchnl_free(struct xen_snd_front_info *front_info, xenbus_free_evtchn(front_info->xb_dev, channel->port); /* End access and free the page. */ - if (channel->gref != GRANT_INVALID_REF) + if (channel->gref != INVALID_GRANT_REF) gnttab_end_foreign_access(channel->gref, page); else free_page(page); @@ -207,7 +207,7 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, channel->index = index; channel->front_info = front_info; channel->state = EVTCHNL_STATE_DISCONNECTED; - channel->gref = GRANT_INVALID_REF; + channel->gref = INVALID_GRANT_REF; page = get_zeroed_page(GFP_KERNEL); if (!page) { ret = -ENOMEM; diff --git a/sound/xen/xen_snd_front_evtchnl.h b/sound/xen/xen_snd_front_evtchnl.h index cbe51fd1ec15..3675fba70564 100644 --- a/sound/xen/xen_snd_front_evtchnl.h +++ b/sound/xen/xen_snd_front_evtchnl.h @@ -15,15 +15,6 @@ struct xen_snd_front_info; -#ifndef GRANT_INVALID_REF -/* - * FIXME: usage of grant reference 0 as invalid grant reference: - * grant reference 0 is valid, but never exposed to a PV driver, - * because of the fact it is already in use/reserved by the PV console. - */ -#define GRANT_INVALID_REF 0 -#endif - /* Timeout in ms to wait for backend to respond. */ #define VSND_WAIT_BACK_MS 3000 -- 2.35.3

1 0

[PATCH] ASoC: cs35l45: Enable BOOST
by Richard Fitzgerald 04 May '22

04 May '22

Do not disable the boost converter during probe. The silicon contains functional default tunings so the boost converter can be left at the chip default enabled state. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 86daa2574388..2367c1a4c10e 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -587,10 +587,6 @@ static int cs35l45_initialize(struct cs35l45_private *cs35l45) return ret; } - /* Default to boost bypass */ - regmap_update_bits(cs35l45->regmap, CS35L45_BLOCK_ENABLES, CS35L45_BST_EN_MASK, - CS35L45_BST_DISABLE_FET_ON << CS35L45_BST_EN_SHIFT); - ret = cs35l45_apply_property_config(cs35l45); if (ret < 0) return ret; -- 2.30.2

2 1

[PATCH] ASoC: SOF: ipc3: Remove the ipc3-ops.h header file
by Peter Ujfalusi 04 May '22

04 May '22

Only the main IPC ops struct should be visible outside of IPC3 code to make sure that the code is correctly abstracted. Instead of keeping the ipc3-ops.h with only the high level ops struct declaration, put the ipc3_ops to sof-priv.h and move all other ops struct declaration into ipc3-priv.h New IPC implementation should follow this route: the main IPC ops should be declared in sof-priv.h and no other IPC version related header be used for generic code. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc.c | 1 - sound/soc/sof/ipc3-control.c | 2 +- sound/soc/sof/ipc3-loader.c | 1 - sound/soc/sof/ipc3-ops.h | 21 --------------------- sound/soc/sof/ipc3-pcm.c | 2 +- sound/soc/sof/ipc3-priv.h | 3 +++ sound/soc/sof/ipc3-topology.c | 2 +- sound/soc/sof/ipc3.c | 1 - sound/soc/sof/sof-priv.h | 3 +++ 9 files changed, 9 insertions(+), 27 deletions(-) delete mode 100644 sound/soc/sof/ipc3-ops.h diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index a6ec4b5a4923..41f3a217be5d 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -17,7 +17,6 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ops.h" -#include "ipc3-ops.h" /** * sof_ipc_send_msg - generic function to prepare and send one IPC message diff --git a/sound/soc/sof/ipc3-control.c b/sound/soc/sof/ipc3-control.c index 4347adcc6543..3fdc0d854e65 100644 --- a/sound/soc/sof/ipc3-control.c +++ b/sound/soc/sof/ipc3-control.c @@ -9,7 +9,7 @@ #include "sof-priv.h" #include "sof-audio.h" -#include "ipc3-ops.h" +#include "ipc3-priv.h" /* IPC set()/get() for kcontrols. */ static int sof_ipc3_set_get_kcontrol_data(struct snd_sof_control *scontrol, bool set) diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c index 14158c52a2b7..f3c741b49519 100644 --- a/sound/soc/sof/ipc3-loader.c +++ b/sound/soc/sof/ipc3-loader.c @@ -9,7 +9,6 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ipc3-priv.h" -#include "ipc3-ops.h" #include "ops.h" static int ipc3_fw_ext_man_get_version(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/ipc3-ops.h b/sound/soc/sof/ipc3-ops.h deleted file mode 100644 index a4784626a3d7..000000000000 --- a/sound/soc/sof/ipc3-ops.h +++ /dev/null @@ -1,21 +0,0 @@ -/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ -/* - * This file is provided under a dual BSD/GPLv2 license. When using or - * redistributing this file, you may do so under either license. - * - * Copyright(c) 2021 Intel Corporation. All rights reserved. - * - * Author: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> - */ - -#ifndef __SOUND_SOC_SOF_IPC3_OPS_H -#define __SOUND_SOC_SOF_IPC3_OPS_H - -#include "sof-priv.h" - -extern const struct sof_ipc_tplg_ops ipc3_tplg_ops; -extern const struct sof_ipc_ops ipc3_ops; -extern const struct sof_ipc_tplg_control_ops tplg_ipc3_control_ops; -extern const struct sof_ipc_pcm_ops ipc3_pcm_ops; - -#endif diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index d7b6c67b2180..c8774a891d6f 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -8,7 +8,7 @@ // #include <sound/pcm_params.h> -#include "ipc3-ops.h" +#include "ipc3-priv.h" #include "ops.h" #include "sof-priv.h" #include "sof-audio.h" diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h index a9b9201508a5..82f9d0cbfb93 100644 --- a/sound/soc/sof/ipc3-priv.h +++ b/sound/soc/sof/ipc3-priv.h @@ -12,6 +12,9 @@ #include "sof-priv.h" /* IPC3 specific ops */ +extern const struct sof_ipc_pcm_ops ipc3_pcm_ops; +extern const struct sof_ipc_tplg_ops ipc3_tplg_ops; +extern const struct sof_ipc_tplg_control_ops tplg_ipc3_control_ops; extern const struct sof_ipc_fw_loader_ops ipc3_loader_ops; /* helpers for fw_ready and ext_manifest parsing */ diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 220ab6c6803f..043554d7cb4a 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -11,7 +11,7 @@ #include <sound/pcm_params.h> #include "sof-priv.h" #include "sof-audio.h" -#include "ipc3-ops.h" +#include "ipc3-priv.h" #include "ops.h" /* Full volume for default values */ diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index efcd201597fa..a8ffc4f99565 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -12,7 +12,6 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ipc3-priv.h" -#include "ipc3-ops.h" #include "ops.h" typedef void (*ipc3_rx_callback)(struct snd_sof_dev *sdev, void *msg_buf); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index a1141ea8d907..4801849cb2ab 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -763,4 +763,7 @@ static inline int sof_resume_clients(struct snd_sof_dev *sdev) } #endif /* CONFIG_SND_SOC_SOF_CLIENT */ +/* Main ops for IPC implementations */ +extern const struct sof_ipc_ops ipc3_ops; + #endif -- 2.36.0

2 1

[PATCH] ASoC: soc-generic-dmaengine-pcm: Use default config when none is given
by Sascha Hauer 04 May '22

04 May '22

snd_dmaengine_pcm_register() can be passed a NULL pointer for the config which means that the we have to test for pcm->config being non NULL before accessing it. Make the code more straight forward by providing a default config when none is passed. With this pcm->config is never NULL and we can skip all the if (pcm->config) tests. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 57 ++++++++++++--------------- 1 file changed, 25 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 285441d6aeed4..87858462bba99 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -79,29 +79,19 @@ static int dmaengine_pcm_hw_params(struct snd_soc_component *component, { struct dmaengine_pcm *pcm = soc_component_to_pcm(component); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - int (*prepare_slave_config)(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct dma_slave_config *slave_config); struct dma_slave_config slave_config; + int ret; - memset(&slave_config, 0, sizeof(slave_config)); - - if (!pcm->config) - prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; - else - prepare_slave_config = pcm->config->prepare_slave_config; + if (!pcm->config->prepare_slave_config) + return 0; - if (prepare_slave_config) { - int ret = prepare_slave_config(substream, params, &slave_config); - if (ret) - return ret; + memset(&slave_config, 0, sizeof(slave_config)); - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - } + ret = pcm->config->prepare_slave_config(substream, params, &slave_config); + if (ret) + return ret; - return 0; + return dmaengine_slave_config(chan, &slave_config); } static int @@ -121,7 +111,7 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, return -EINVAL; } - if (pcm->config && pcm->config->pcm_hardware) + if (pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); @@ -188,7 +178,6 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_component_to_pcm(component); struct snd_dmaengine_dai_dma_data *dma_data; - dma_filter_fn fn = NULL; if (rtd->num_cpus > 1) { dev_err(rtd->dev, @@ -201,13 +190,11 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config && pcm->config->compat_request_channel) + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - if (pcm->config) - fn = pcm->config->compat_filter_fn; - - return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); + return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, + dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct device *dev, @@ -239,12 +226,12 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, size_t max_buffer_size; unsigned int i; - if (config && config->prealloc_buffer_size) + if (config->prealloc_buffer_size) prealloc_buffer_size = config->prealloc_buffer_size; else prealloc_buffer_size = prealloc_buffer_size_kbytes * 1024; - if (config && config->pcm_hardware && config->pcm_hardware->buffer_bytes_max) + if (config->pcm_hardware && config->pcm_hardware->buffer_bytes_max) max_buffer_size = config->pcm_hardware->buffer_bytes_max; else max_buffer_size = SIZE_MAX; @@ -254,7 +241,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, if (!substream) continue; - if (!pcm->chan[i] && config && config->chan_names[i]) + if (!pcm->chan[i] && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, config->chan_names[i]); @@ -367,10 +354,10 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct dma_chan *chan; if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && - !(config && config->dma_dev && config->dma_dev->of_node))) + !(config->dma_dev && config->dma_dev->of_node))) return 0; - if (config && config->dma_dev) { + if (config->dma_dev) { /* * If this warning is seen, it probably means that your Linux * device structure does not match your HW device structure. @@ -387,7 +374,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = "rx-tx"; else name = dmaengine_pcm_dma_channel_names[i]; - if (config && config->chan_names[i]) + if (config->chan_names[i]) name = config->chan_names[i]; chan = dma_request_chan(dev, name); if (IS_ERR(chan)) { @@ -425,6 +412,10 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) } } +static const struct snd_dmaengine_pcm_config snd_dmaengine_pcm_default_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -445,6 +436,8 @@ int snd_dmaengine_pcm_register(struct device *dev, #ifdef CONFIG_DEBUG_FS pcm->component.debugfs_prefix = "dma"; #endif + if (!config) + config = &snd_dmaengine_pcm_default_config; pcm->config = config; pcm->flags = flags; @@ -452,7 +445,7 @@ int snd_dmaengine_pcm_register(struct device *dev, if (ret) goto err_free_dma; - if (config && config->process) + if (config->process) driver = &dmaengine_pcm_component_process; else driver = &dmaengine_pcm_component; -- 2.30.2

2 1

ASoC component/card relationship
by Pierre-Louis Bossart 04 May '22

04 May '22

Hi, In the existing ASoC code, there is a fixed mapping between ASoC card and component. A component relies on a ->card pointer that is set during the probe. A component cannot be used by or "bound to" more than one card [1] This has interesting impacts on how a codec or DSP driver need to be implemented. In the AVS series posted this week, multiple components are registered by the DSP driver, following an interface-based split. There's in addition a second-level split, where the logic is pushed further: the DSP driver partitions the SSP DAIs in different set of 'dai_driver's used by different components, which are in turn used by different cards. What is done in these patches is not wrong, and is probably the only solution to support a real-world platform with the existing ASoC code, but are the framework assumptions correct? In this example, the board-level information on which interface is used for what functionality trickles down to the lowest level of the DSP driver implementation. I believe this breaks to some extent the 'clean' split between platform and machine driver(s), and it's not quite aligned with the usual notion of register/probe used across frameworks, be it for drivers/clocks/you name it. A similar case could happen in a codec driver, if independent functionality such as headset and amplifier support was exposed by separate cards, that would in turn mandate that the codec driver exposed N components, each handling different functionality but the same type of DAI. An alternative approach would be that the DSP driver exposes all the possible DAIs that can be used, and the binding is refined to allow for more flexibility. I think it's really the individual DAI that cannot be used by more than one card. I figured I would ask on this mailing list if a) I am not mistaken on the component/card relationship and b) if this is by design, or if we want to clarify what a component is and what its restrictions might be. Thanks for your feedback/comments -Pierre [1] https://elixir.bootlin.com/linux/latest/source/sound/soc/soc-core.c#L1364

6 12

[PATCH v5 0/2] Add generic serial MIDI driver using serial bus API
by Daniel Kaehn 04 May '22

04 May '22

Generic serial MIDI driver adding support for using serial devices compatible with the serial bus as raw MIDI devices, allowing using additional serial devices not compatible with the existing serial-u16550 driver. Supports only setting standard serial baudrates on the underlying serial device; however, the underlying serial device can be configured so that a requested 38.4 kBaud is actually the standard MIDI 31.25 kBaud. Supports DeviceTree configuration. Changes in v5: - Reword and add to description in dt-binding - Change 'speed' dt property to 'current-speed' - Move MIDI output loop onto workqueue (since this could loop quite a while, if ALSA provides a continuous stream of bytes) - Add tx_state bit flags to snd_serial_generic struct - Safegard critical section in tx_work with atomic bit ops on tx_state - Switch operations on `filemode` to use atomic bit ops Changes in v4: - Fix regressed typo - Correct 3.84 kBaud -> 38.4 kBaud in DT & Kconfig (sorry about spam - noticed after sending v3 and didn't want to let the error sit around for too long) Changes in v3: - Replace use of snd_printk() with dev_* alternatives - Removed unnecessary initialization of err variables - Replaced instances of `== SERIAL_MODE_NOT_OPENED` with zero check - Loop on output_write to completely fill output buffer if data available - Depend on CONFIG_OF in Kconfig - Replace use of devm_kzalloc() with extra_size allocation in snd_devm_card_new() - Use module_serdev_device_driver() instead of module_init() and module_exit(0) Changes in v2: - Fix 'snd_serial_generic_write_wakeup' missing static keyword - Correct 3.125 kBaud > 31.25 kBaud in documentation for MIDI The need for this driver arose from a project using a Raspberry Pi4 which needed to receive and send raw MIDI with low latency. The pl011 UART used is not compatible with the existing serial MIDI driver made for u16550-style devices. Using a userspace program such as ttymidi to feed input from the TTY device to a virtual ALSA MIDI device was functional, but not ideal. I am not sure if a MIDI driver needing the mentioned 'hack' to clock 38.4 kBaud down to the standard MIDI baud is permissible in the mainline kernel, but am submitting nevertheless in case it is useful. To my knowledge, it doesn't seem that there would be any way for this driver to manually configure a serial port to 31.25 kBaud using the serial bus API (please correct me f I'm wrong). In my use case, I am actually configuring one port to run at 115.2 kBaud for faster communication with a custom onboard MIDI controller. Daniel Kaehn (2): dt-bindings: sound: Add generic serial MIDI device Add generic serial MIDI driver using serial bus API .../devicetree/bindings/sound/serialmidi.yaml | 46 +++ sound/drivers/Kconfig | 18 + sound/drivers/Makefile | 2 + sound/drivers/serial-generic.c | 377 ++++++++++++++++++ 4 files changed, 443 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/serialmidi.yaml create mode 100644 sound/drivers/serial-generic.c base-commit: 3e71713c9e75c34fc03f55ea86b381856ca952ee -- 2.33.0

2 5

Re: [PATCH] ASoC: Intel: sof_cs42l42: add support for adl_mx98360a_cs4242 and BT offload audio
by Cezary Rojewski 04 May '22

04 May '22

On 2022-05-04 2:24 PM, Terry Chen wrote: > This patch adds driver data for adl_mx98360a_cs4242 which supports > two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec > on SSP0 running on ADL platform. Also add the capability to machine > driver of creating DAI Link for BT offload. Although BT offload > always uses SSP2 port but we reserve the flexibility to assign > the port number in macro. Title's length seems off. Please shorten it a bit. Commit message formatting (line width) is not cohevise across the lines either. Another suggestion: skip redundant bits at the front e.g.: s/This patch adds/Add/. When adding new code (new functionality), it's good to plainly state: "To be able to do X we add capability Y". The last sentence in your message is a side note. It's good that you're being transparent here, my only advice is that you add a new line before such notes as these are not the main dish here, really. Title suggest you're doing two things here - adding support for some ADL configuration (1) and BT audio offload (2). Recommendation is to splitting non-atomic changes so each patch in the series answers single problem. > Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> > --- > sound/soc/intel/boards/sof_cs42l42.c | 88 ++++++++++++++++++- > .../intel/common/soc-acpi-intel-adl-match.c | 8 ++ > 2 files changed, 92 insertions(+), 4 deletions(-) > ... > @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { > } > }; > > +static struct snd_soc_dai_link_component dummy_component[] = { > + { > + .name = "snd-soc-dummy", > + .dai_name = "snd-soc-dummy-dai", > + } > +}; Isn't there some SND_SOC_DAILINK_DEF() macro for that already? > static int create_spk_amp_dai_links(struct device *dev, > struct snd_soc_dai_link *links, > struct snd_soc_dai_link_component *cpus, > @@ -467,9 +480,52 @@ static int create_hdmi_dai_links(struct device *dev, > return -ENOMEM; > } > > +static int create_bt_offload_dai_links(struct device *dev, > + struct snd_soc_dai_link *links, > + struct snd_soc_dai_link_component *cpus, > + int *id, int ssp_bt) > +{ > + int ret = 0; Looks like there is no need for 'ret' variable here. > + > + /* bt offload */ > + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) > + return 0; > + > + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", > + ssp_bt); Current limit is 100. > + if (!links[*id].name) > + goto devm_err; This 'goto' statement seems redundant. I might be missing something but returning -ENOMEM should suffice. Label 'devm_err' does not do anything special anyway. > + > + links[*id].id = *id; > + links[*id].codecs = dummy_component; > + links[*id].num_codecs = ARRAY_SIZE(dummy_component); > + links[*id].platforms = platform_component; > + links[*id].num_platforms = ARRAY_SIZE(platform_component); > + > + links[*id].dpcm_playback = 1; > + links[*id].dpcm_capture = 1; > + links[*id].no_pcm = 1; > + links[*id].cpus = &cpus[*id]; > + links[*id].num_cpus = 1; > + > + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, > + "SSP%d Pin", > + ssp_bt); Ditto - limit 100. Applies to every single occurence below too. > + if (!links[*id].cpus->dai_name) > + goto devm_err; > + > + (*id)++; Are the parenthesis needed? Should be save to drop them. > + > + return 0; > + > +devm_err: > + return ret; Once 'ret/devm_err' situation is clarified, this block can be dropped. > +}

1 0

pcm: multi: return correct hwptr and avail from snd_pcm_multi_status()
by GitHub pull_request - opened 04 May '22

04 May '22

alsa-project/alsa-lib pull request #224 was opened from infmagic2047: Ensure the logic of snd_pcm_multi_status() is consistent with snd_pcm_multi_avail_update(). Signed-off-by: Yutao Yuan <infmagic2047reg(a)outlook.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/224 Patch URL : https://github.com/alsa-project/alsa-lib/pull/224.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa starts muted every reboot. /var/lib/alsa/asound.conf file does not exist
by GitHub issues - opened 04 May '22

04 May '22

alsa-project/alsa-utils issue #145 was opened from dontbejay: This is a multi distro issue. This has occured in EndeavourOS(XFCE), and Ubuntu(GNOME) as well. Currently I am on openSUSE(KDE) and alsa has the same issue. Every boot alsa has "Front" speakers Muted on my Sound Blaster Z. I have to go into alsamixer and unmute them. "sudo alsactl store" seems to do nothing to automatically restore settings on boot. Typing "sudo alsactl restore" Does fix the issue when it is used. http://alsa-project.org/db/?f=8354e69bed4316f798b2d8300f8109cb6b672d18 (the Front speaker has been manually unmuted) Issue URL : https://github.com/alsa-project/alsa-utils/issues/145 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Arch Linux Alsa Driver mic misconfigured 0000:00:1f.3
by GitHub issues - opened 04 May '22

04 May '22

alsa-project/alsa-firmware issue #6 was opened from WalterSmuts: Hi, I have a `Xiaomi Mi Laptop Pro X 15` running arch linux and the `pulseaudio-alsa` package seems to make sound work out of the box. Unfortunately it doesn't configure the mic properly. Attempting on a ubuntu live-usb I found the mic working. I ran the info script for both OS's (see below). My interpretation of the info dumped by the script is that ubuntu loads the `sof-audio-pci` drivers and arch linux seems to load some other intel drivers. Looking through the [arch wiki](https://wiki.archlinux.org/title/Advanced_Linux_Sound_Architecture#AL… I see reference to some newer laptops requiring the sof drivers: > [sof-firmware](https://archlinux.org/packages/?name=sof-firmware) and/or [alsa-ucm-conf](https://archlinux.org/packages/?name=alsa-ucm-conf) are required for some newer laptop models (mainly since 2019) because they implement their drivers with firmware provided by the [Sound Open Firmware](https://www.sofproject.org/) project. So installing the `sof-firmware` package and restarting the laptop unfortunately still does not fix my issue. Looks to me like the old drivers are still taking preference. Can someone guide me in the right direction here? * [Working Ubuntu Info](http://alsa-project.org/db/?f=e66f34b60915348fa835bdde9e849119e103165… * [Non-working Arch Info](http://alsa-project.org/db/?f=1a4c03f9abd116c5a0b149e51e436f004fa2022… Issue URL : https://github.com/alsa-project/alsa-firmware/issues/6 Repository URL: https://github.com/alsa-project/alsa-firmware

2 1

[v4 00/18] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 03 May '22

03 May '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codecs - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (18): ASoC: mediatek: mt6366: support for mt6366 codec dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: support audio clock control in platform driver ASoC: mediatek: mt8186: support gpio control in platform driver ASoC: mediatek: mt8186: add platform driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 71 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 71 + sound/soc/codecs/mt6358.c | 1 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 651 ++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 106 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 244 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3005 +++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + .../soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 873 +++++ .../soc/mediatek/mt8186/mt8186-dai-hostless.c | 298 ++ .../soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 236 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1355 ++++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 423 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 695 ++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 695 ++++ .../mediatek/mt8186/mt8186-interconnection.h | 69 + .../soc/mediatek/mt8186/mt8186-misc-control.c | 294 ++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1003 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 979 ++++++ sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 ++++++++++++++++ 32 files changed, 15028 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h -- 2.25.1

4 31

[PATCH v4] ASoC: SOF: ipc3-topology: Correct get_control_data for non bytes payload
by Peter Ujfalusi 03 May '22

03 May '22

It is possible to craft a topology where sof_get_control_data() would do out of bounds access because it expects that it is only called when the payload is bytes type. Confusingly it also handles other types of controls, but the payload parsing implementation is only valid for bytes. Fix the code to count the non bytes controls and instead of storing a pointer to sof_abi_hdr in sof_widget_data (which is only valid for bytes), store the pointer to the data itself and add a new member to save the size of the data. In case of non bytes controls we store the pointer to the chanv itself, which is just an array of values at the end. In case of bytes control, drop the wrong cdata->data (wdata[i].pdata) check against NULL since it is incorrect and invalid in this context. The data is pointing to the end of cdata struct, so it should never be null. Reported-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v3: - Move the comment about validating data along with the code which validates the data (checks the SOF_ABI_MAGIC) Changes since v2: - Drop the cdata->data check against NULL as it is not a valid test and since we are in sof_get_control_data() the memory has been already allocated changes since v1: - adjust the payload size for non bytes controls by subtracting the size of the sof_ipc_ctrl_data struct, plus add comment to note this Regards, Peter sound/soc/sof/ipc3-topology.c | 39 +++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 572bcbfdb356..bc848f556b88 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -20,7 +20,8 @@ struct sof_widget_data { int ctrl_type; int ipc_cmd; - struct sof_abi_hdr *pdata; + void *pdata; + size_t pdata_size; struct snd_sof_control *control; }; @@ -784,16 +785,26 @@ static int sof_get_control_data(struct snd_soc_component *scomp, } cdata = wdata[i].control->ipc_control_data; - wdata[i].pdata = cdata->data; - if (!wdata[i].pdata) - return -EINVAL; - /* make sure data is valid - data can be updated at runtime */ - if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES && - wdata[i].pdata->magic != SOF_ABI_MAGIC) - return -EINVAL; + if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES) { + /* make sure data is valid - data can be updated at runtime */ + if (cdata->data->magic != SOF_ABI_MAGIC) + return -EINVAL; + + wdata[i].pdata = cdata->data->data; + wdata[i].pdata_size = cdata->data->size; + } else { + /* points to the control data union */ + wdata[i].pdata = cdata->chanv; + /* + * wdata[i].control->size is calculated with struct_size + * and includes the size of struct sof_ipc_ctrl_data + */ + wdata[i].pdata_size = wdata[i].control->size - + sizeof(struct sof_ipc_ctrl_data); + } - *size += wdata[i].pdata->size; + *size += wdata[i].pdata_size; /* get data type */ switch (cdata->cmd) { @@ -876,10 +887,12 @@ static int sof_process_load(struct snd_soc_component *scomp, */ if (ipc_data_size) { for (i = 0; i < widget->num_kcontrols; i++) { - memcpy(&process->data[offset], - wdata[i].pdata->data, - wdata[i].pdata->size); - offset += wdata[i].pdata->size; + if (!wdata[i].pdata_size) + continue; + + memcpy(&process->data[offset], wdata[i].pdata, + wdata[i].pdata_size); + offset += wdata[i].pdata_size; } } -- 2.36.0

3 2

[PATCH] ASoC: adau1761: Add ADAU1761-as-ADAU1361 compatibility mode
by Ricard Wanderlof 03 May '22

03 May '22

During probe, determine if the chip is in fact an ADAU1761 even though an ADAU1361 is specified, and perform additional operations to enable the ADAU1761 to behave as an ADAU1361, i.e. disregarding the DSP and setting up routing and PM transparently. This enables either chip to be mounted when an ADAU1361 is specified. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/adau1761.c | 86 +++++++++++++++++++++++++++++++++++-- sound/soc/codecs/adau17x1.c | 20 +++++++-- sound/soc/codecs/adau17x1.h | 1 + 3 files changed, 99 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index fb006fc81653..8f887227981f 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -556,8 +556,6 @@ static const struct snd_soc_dapm_route adau1761_dapm_routes[] = { { "Left DAC", NULL, "Interpolator Resync Clock" }, { "Right DAC", NULL, "Interpolator Resync Clock" }, - { "DSP", NULL, "Digital Clock 0" }, - { "Slew Clock", NULL, "Digital Clock 0" }, { "Right Playback Mixer", NULL, "Slew Clock" }, { "Left Playback Mixer", NULL, "Slew Clock" }, @@ -569,6 +567,56 @@ static const struct snd_soc_dapm_route adau1761_dapm_routes[] = { { "Digital Clock 1", NULL, "SYSCLK" }, }; +static const struct snd_soc_dapm_route adau1761_dapm_dsp_routes[] = { + { "DSP", NULL, "Digital Clock 0" }, +}; + +static int adau1761_compatibility_probe(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + struct regmap *regmap = adau->regmap; + int val, ret = 0; + + /* Only consider compatibility mode when ADAU1361 was specified. */ + if (adau->type != ADAU1361) + return 0; + + regcache_cache_bypass(regmap, true); + + /* + * This will enable the core clock and bypass the PLL, + * so that we can access the registers for probing purposes + * (without having to set up the PLL). + */ + regmap_write(regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); + + /* + * ADAU17X1_SERIAL_SAMPLING_RATE doesn't exist in non-DSP chips; + * reading it results in zero at all times, and write is a no-op. + * Use this register to probe for ADAU1761. + */ + regmap_write(regmap, ADAU17X1_SERIAL_SAMPLING_RATE, 1); + ret = regmap_read(regmap, ADAU17X1_SERIAL_SAMPLING_RATE, &val); + if (ret) + goto exit; + if (val != 1) + goto exit; + regmap_write(regmap, ADAU17X1_SERIAL_SAMPLING_RATE, 0); + ret = regmap_read(regmap, ADAU17X1_SERIAL_SAMPLING_RATE, &val); + if (ret) + goto exit; + if (val != 0) + goto exit; + + adau->type = ADAU1761_AS_1361; +exit: + /* Disable core clock after probing. */ + regmap_write(regmap, ADAU17X1_CLOCK_CONTROL, 0); + regcache_cache_bypass(regmap, false); + return ret; +} + static int adau1761_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { @@ -823,7 +871,11 @@ static int adau1761_component_probe(struct snd_soc_component *component) if (ret) return ret; - if (adau->type == ADAU1761) { + /* + * If we've got an ADAU1761, or an ADAU1761 operating as an + * ADAU1361, we need these non-DSP related DAPM widgets and routes. + */ + if (adau->type == ADAU1761 || adau->type == ADAU1761_AS_1361) { ret = snd_soc_dapm_new_controls(dapm, adau1761_dapm_widgets, ARRAY_SIZE(adau1761_dapm_widgets)); if (ret) @@ -834,7 +886,29 @@ static int adau1761_component_probe(struct snd_soc_component *component) if (ret) return ret; } - + /* + * These routes are DSP related and only used when we have a + * bona fide ADAU1761. + */ + if (adau->type == ADAU1761) { + ret = snd_soc_dapm_add_routes(dapm, adau1761_dapm_dsp_routes, + ARRAY_SIZE(adau1761_dapm_dsp_routes)); + if (ret) + return ret; + } + /* + * In the ADAU1761, by default, the AIF is routed to the DSP, whereas + * for the ADAU1361, the AIF is permanently routed to the ADC and DAC. + * Thus, if we have an ADAU1761 masquerading as an ADAU1361, + * we need to explicitly route the AIF to the ADC and DAC. + * For the ADAU1761, this is normally done by set_tdm_slot, but this + * function is not necessarily called during stream setup, so set up + * the compatible AIF routings here from the start. + */ + if (adau->type == ADAU1761_AS_1361) { + regmap_write(adau->regmap, ADAU17X1_SERIAL_INPUT_ROUTE, 0x01); + regmap_write(adau->regmap, ADAU17X1_SERIAL_OUTPUT_ROUTE, 0x01); + } ret = adau17x1_add_routes(component); if (ret < 0) return ret; @@ -919,6 +993,10 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, if (ret) return ret; + ret = adau1761_compatibility_probe(dev); + if (ret) + return ret; + /* Enable cache only mode as we could miss writes before bias level * reaches standby and the core clock is enabled */ regcache_cache_only(regmap, true); diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index af05463af4ac..c0f44ecef606 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -334,6 +334,17 @@ static bool adau17x1_has_dsp(struct adau *adau) } } +/* Chip has a DSP but we're pretending it doesn't. */ +static bool adau17x1_has_disused_dsp(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761_AS_1361: + return true; + default: + return false; + } +} + static bool adau17x1_has_safeload(struct adau *adau) { switch (adau->type) { @@ -516,10 +527,11 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, ADAU17X1_CONVERTER0_CONVSR_MASK, div); - if (adau17x1_has_dsp(adau)) { + + if (adau17x1_has_dsp(adau) || adau17x1_has_disused_dsp(adau)) regmap_write(adau->regmap, ADAU17X1_SERIAL_SAMPLING_RATE, div); + if (adau17x1_has_dsp(adau)) regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div); - } if (adau->sigmadsp) { ret = adau17x1_setup_firmware(component, params_rate(params)); @@ -663,7 +675,7 @@ static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, switch (slot_width * slots) { case 32: - if (adau->type == ADAU1761) + if (adau->type == ADAU1761 || adau->type == ADAU1761_AS_1361) return -EINVAL; ser_ctrl1 = ADAU17X1_SERIAL_PORT1_BCLK32; @@ -738,7 +750,7 @@ static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, ADAU17X1_SERIAL_PORT1_BCLK_MASK, ser_ctrl1); - if (!adau17x1_has_dsp(adau)) + if (!adau17x1_has_dsp(adau) && !adau17x1_has_disused_dsp(adau)) return 0; if (adau->dsp_bypass[SNDRV_PCM_STREAM_PLAYBACK]) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 98a3b6f5bc96..5e58abfffc3d 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -10,6 +10,7 @@ enum adau17x1_type { ADAU1361, ADAU1761, + ADAU1761_AS_1361, ADAU1381, ADAU1781, }; -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[PATCH] ASoC: dapm: Don't fold register value changes into notifications
by Mark Brown 03 May '22

03 May '22

DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ca917a849c42..869c76506b66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3437,7 +3437,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3539,7 +3538,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.30.2

1 1

[PATCH] ASoC: rt5514: Fix event generation for "DSP Voice Wake Up" control
by Mark Brown 03 May '22

03 May '22

The driver has a custom put function for "DSP Voice Wake Up" which does not generate event notifications on change, instead returning 0. Since we already exit early in the case that there is no change this can be fixed by unconditionally returning 1 at the end of the function. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1b9263460142..be8ece4630df 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -419,7 +419,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, } } - return 0; + return 1; } static const struct snd_kcontrol_new rt5514_snd_controls[] = { -- 2.30.2

1 1

[PATCH 00/11] ASoC: SOF: Miscellaneous preparatory patches for IPC4
by Ranjani Sridharan 03 May '22

03 May '22

This series includes last few remaining miscellaneous patches to prepare for the introduction of new IPC version, IPC4, in the SOF driver. The changes include new IPC ops for topology parsing to set up the volume table, prepare the widgets for set up and free the routes. The remaining patches introduce new fields in the existing data structures for use in IPC4 and align the flows for widget/route set up so that they are common for both IPC3 and IPC4. Ranjani Sridharan (11): ASoC: SOF: Add a new op to set up volume table ASoC: SOF: sof-audio: reset route status before freeing widget ASoC: SOF: sof-audio: Set up widgets from source to sink ASoC: SOF: pcm: Move the call to snd_sof_pcm_platform_hw_params() ASoC: SOF: expose a couple of functions ASoC: SOF: Add a route_free op in struct sof_ipc_tplg_ops ASoC: SOF: Add two new fields to struct snd_sof_widget ASoC: SOF: pcm: remove unnecessary function declaration ASoC: SOF: topology: Skip parsing DAI link tokens if not needed ASoC: SOF: clarify use of widget complete flag ASoC: SOF: Add a prepare op to IPC topology widget ops sound/soc/sof/ipc3-control.c | 38 +++-- sound/soc/sof/pcm.c | 23 +-- sound/soc/sof/sof-audio.c | 302 +++++++++++++++++++++++++++++------ sound/soc/sof/sof-audio.h | 68 +++++++- sound/soc/sof/topology.c | 44 +++-- 5 files changed, 368 insertions(+), 107 deletions(-) -- 2.25.1

2 12

[PATCH] ASoC: max98090: Fix event generation for custom controls
by Mark Brown 03 May '22

03 May '22

The previous fix for event generation for custom controls compared the value already in the register with the value being written, missing the logic that only applies the value to the register when the control is already enabled. Fix this, compare the value cached in the driver data rather than the register. This should really be an autodisable control rather than open coded. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98090.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 5146837c0fce..48dcf071bb5a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -396,6 +396,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, unsigned int sel = ucontrol->value.integer.value[0]; unsigned int val = snd_soc_component_read(component, mc->reg); unsigned int *select; + int change; switch (mc->reg) { case M98090_REG_MIC1_INPUT_LEVEL: @@ -416,6 +417,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, if (sel < 0 || sel > mc->max) return -EINVAL; + change = *select != sel; *select = sel; /* Setting a volume is only valid if it is already On */ @@ -430,7 +432,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return *select != val; + return change; } static const char *max98090_perf_pwr_text[] = -- 2.30.2

1 1

[PATCH AUTOSEL 5.17 01/22] ASoC: soc-pcm: use GFP_KERNEL when the code is sleepable
by Sasha Levin 02 May '22

02 May '22

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit fb6d679fee95d272c0a94912c4e534146823ee89 ] At the kzalloc() call in dpcm_be_connect(), there is no spin lock involved. It's merely protected by card->pcm_mutex, instead. The spinlock is applied at the later call with snd_soc_pcm_stream_lock_irq() only for the list manipulations. (See it's *_irq(), not *_irqsave(); that means the context being sleepable at that point.) So, we can use GFP_KERNEL safely there. This patch revert commit d8a9c6e1f676 ("ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure") which is no longer needed since commit b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking"). Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Link: https://lore.kernel.org/r/e740f1930843060e025e3c0f17ec1393cfdafb26.16487579… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9a954680d492..11c9853e9e80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1214,7 +1214,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, be_substream->pcm->nonatomic = 1; } - dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_ATOMIC); + dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL); if (!dpcm) return -ENOMEM; -- 2.35.1

4 10

[PATCH v2 0/2] Let userspace know when snd-hda-intel needs i915
by Mauro Carvalho Chehab 01 May '22

01 May '22

Currently, kernel/module annotates module dependencies when request_symbol is used, but it doesn't cover more complex inter-driver dependencies that are subsystem and/or driver-specific. In the case of hdmi sound, depending on the CPU/GPU, sometimes the snd_hda_driver can talk directly with the hardware, but sometimes, it uses the i915 driver. When the snd_hda_driver uses i915, it should first be unbind/rmmod, as otherwise trying to unbind/rmmod the i915 driver cause driver issues, as as reported by CI tools with different GPU models: https://intel-gfx-ci.01.org/tree/drm-tip/IGT_6415/fi-tgl-1115g4/igt@core_ho… https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_11495/bat-adlm-1/igt@i915_m… In the past, just a few CPUs were doing such bindings, but this issue now applies to all "modern" Intel CPUs that have onboard graphics, as well as to the newer discrete GPUs. With the discrete GPU case, the HDA controller is physically separate and requires i915 to power on the hardware for all hardware access. In this case, the issue is hit basicly 100% of the time. With on-board graphics, i915 driver is needed only when the display codec is accessed. If i915 is unbind during runtime suspend, while snd-hda-intel is still bound, nothing bad happens, but unbinding i915 on other situations may also cause issues. So, add support at kernel/modules to allow snd-hda drivers to properly annotate when a dependency on a DRM driver dependencies exists, and add a call to such new function at the snd-hda driver when it successfully binds into the DRM driver. This would allow userspace tools to check and properly remove the audio driver before trying to remove or unbind the GPU driver. It should be noticed that this series conveys the hidden module dependencies. Other changes are needed in order to allow removing or unbinding the i915 driver while keeping the snd-hda-intel driver loaded/bound. With that regards, there are some discussions on how to improve this at alsa-devel a while back: https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/190099… So, future improvements on both in i915 and the audio drivers could be made. E.g. with discrete GPUs, it's the only codec of the card, so it seems feasible to detach the ALSA card if i915 is bound (using infra made for VGA switcheroo), but, until these improvements are done and land in upstream, audio drivers needs to be unbound if i915 driver goes unbind. Yet, even if such fixes got merged, this series is still needed, as it makes such dependencies more explicit and easier to debug. PS.: This series was generated against next-20220428. --- v2: the dependencies are now handled directly at try_module_get(). Mauro Carvalho Chehab (2): module: update dependencies at try_module_get() ALSA: hda - identify when audio is provided by a video driver include/linux/module.h | 4 +++- kernel/module/main.c | 35 +++++++++++++++++++++++++++++++++-- sound/hda/hdac_component.c | 2 +- 3 files changed, 37 insertions(+), 4 deletions(-) -- 2.35.1

3 5

[PATCH v3 0/2] Let userspace know when snd-hda-intel needs i915
by Mauro Carvalho Chehab 30 Apr '22

30 Apr '22

Currently, kernel/module annotates module dependencies when request_symbol is used, but it doesn't cover more complex inter-driver dependencies that are subsystem and/or driver-specific. In the case of hdmi sound, depending on the CPU/GPU, sometimes the snd_hda_driver can talk directly with the hardware, but sometimes, it uses the i915 driver. When the snd_hda_driver uses i915, it should first be unbind/rmmod, as otherwise trying to unbind/rmmod the i915 driver cause driver issues, as as reported by CI tools with different GPU models: https://intel-gfx-ci.01.org/tree/drm-tip/IGT_6415/fi-tgl-1115g4/igt@core_ho… https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_11495/bat-adlm-1/igt@i915_m… In the past, just a few CPUs were doing such bindings, but this issue now applies to all "modern" Intel CPUs that have onboard graphics, as well as to the newer discrete GPUs. With the discrete GPU case, the HDA controller is physically separate and requires i915 to power on the hardware for all hardware access. In this case, the issue is hit basicly 100% of the time. With on-board graphics, i915 driver is needed only when the display codec is accessed. If i915 is unbind during runtime suspend, while snd-hda-intel is still bound, nothing bad happens, but unbinding i915 on other situations may also cause issues. So, add support at kernel/modules to allow snd-hda drivers to properly annotate when a dependency on a DRM driver dependencies exists, and add a call to such new function at the snd-hda driver when it successfully binds into the DRM driver. This would allow userspace tools to check and properly remove the audio driver before trying to remove or unbind the GPU driver. It should be noticed that this series conveys the hidden module dependencies. Other changes are needed in order to allow removing or unbinding the i915 driver while keeping the snd-hda-intel driver loaded/bound. With that regards, there are some discussions on how to improve this at alsa-devel a while back: https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/190099… So, future improvements on both in i915 and the audio drivers could be made. E.g. with discrete GPUs, it's the only codec of the card, so it seems feasible to detach the ALSA card if i915 is bound (using infra made for VGA switcheroo), but, until these improvements are done and land in upstream, audio drivers needs to be unbound if i915 driver goes unbind. Yet, even if such fixes got merged, this series is still needed, as it makes such dependencies more explicit and easier to debug. PS.: This series was generated against next-20220428. --- v3: minor fixes: - fixed a checkpatch warning; - use a single line for the new function prototype. v2: - the dependencies are now handled directly at try_module_get(). Mauro Carvalho Chehab (2): module: update dependencies at try_module_get() ALSA: hda - identify when audio is provided by a video driver include/linux/module.h | 4 +++- kernel/module/main.c | 33 +++++++++++++++++++++++++++++++-- sound/hda/hdac_component.c | 2 +- 3 files changed, 35 insertions(+), 4 deletions(-) -- 2.35.1

3 5

[PATCH v4 0/2] Let userspace know when snd-hda-intel needs i915
by Mauro Carvalho Chehab 30 Apr '22

30 Apr '22

Currently, kernel/module annotates module dependencies when request_symbol is used, but it doesn't cover more complex inter-driver dependencies that are subsystem and/or driver-specific. In the case of hdmi sound, depending on the CPU/GPU, sometimes the snd_hda_driver can talk directly with the hardware, but sometimes, it uses the i915 driver. When the snd_hda_driver uses i915, it should first be unbind/rmmod, as otherwise trying to unbind/rmmod the i915 driver cause driver issues, as as reported by CI tools with different GPU models: https://intel-gfx-ci.01.org/tree/drm-tip/IGT_6415/fi-tgl-1115g4/igt@core_ho… https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_11495/bat-adlm-1/igt@i915_m… In the past, just a few CPUs were doing such bindings, but this issue now applies to all "modern" Intel CPUs that have onboard graphics, as well as to the newer discrete GPUs. With the discrete GPU case, the HDA controller is physically separate and requires i915 to power on the hardware for all hardware access. In this case, the issue is hit basicly 100% of the time. With on-board graphics, i915 driver is needed only when the display codec is accessed. If i915 is unbind during runtime suspend, while snd-hda-intel is still bound, nothing bad happens, but unbinding i915 on other situations may also cause issues. So, add support at kernel/modules to allow snd-hda drivers to properly annotate when a dependency on a DRM driver dependencies exists, and add a call to such new function at the snd-hda driver when it successfully binds into the DRM driver. This would allow userspace tools to check and properly remove the audio driver before trying to remove or unbind the GPU driver. It should be noticed that this series conveys the hidden module dependencies. Other changes are needed in order to allow removing or unbinding the i915 driver while keeping the snd-hda-intel driver loaded/bound. With that regards, there are some discussions on how to improve this at alsa-devel a while back: https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/190099… So, future improvements on both in i915 and the audio drivers could be made. E.g. with discrete GPUs, it's the only codec of the card, so it seems feasible to detach the ALSA card if i915 is bound (using infra made for VGA switcheroo), but, until these improvements are done and land in upstream, audio drivers needs to be unbound if i915 driver goes unbind. Yet, even if such fixes got merged, this series is still needed, as it makes such dependencies more explicit and easier to debug. PS.: This series was generated against next-20220428. --- v4: - fix a compilation warning reported by Intel's Kernel robot when !CONFIG_MODULE_UNLOAD or !CONFIG_MODULE. v3: minor fixes: - fixed a checkpatch warning; - use a single line for the new function prototype. v2: - the dependencies are now handled directly at try_module_get(). Mauro Carvalho Chehab (2): module: update dependencies at try_module_get() ALSA: hda - identify when audio is provided by a video driver include/linux/module.h | 8 +++++--- kernel/module/main.c | 33 +++++++++++++++++++++++++++++++-- sound/hda/hdac_component.c | 2 +- 3 files changed, 37 insertions(+), 6 deletions(-) -- 2.35.1

1 2

how can i swich between sound card without reboot in archlinux
by GitHub issues - edited 30 Apr '22

30 Apr '22

alsa-project/alsa-utils issue #144 was edited from EN-KS: I use riverwm and kitty. Currently there is no sound output when i try to plugin a usb 2.4G wireless headset in and then use alsamixer F6 to select the usb sound card unless I leave the usb adapter on the machine and reboot. the usb sound card was detected tho, Is this what it suppose to be? or any procedure that I missed? Issue URL : https://github.com/alsa-project/alsa-utils/issues/144 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

how can i swich between sound card without reboot in archlinux
by GitHub issues - opened 30 Apr '22

30 Apr '22

alsa-project/alsa-utils issue #144 was opened from EN-KS: currently there is no sound output when i try to plugin a usb 2.4G wireless headset in and then F6 to select the usb sound card unless I leave the usb adapter on the machine and reboot. the usb sound card was detected tho, Is this what it suppose to be? or any procedure that I missed? Issue URL : https://github.com/alsa-project/alsa-utils/issues/144 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] conf: vc4-hdmi: use a proper hdmi pcm, fix broken default pcm
by Matthias Reichl 30 Apr '22

30 Apr '22

Userspace expects to see a HDMI pcm, so remove the somewhat broken front and iec958 pcms and add a proper hdmi pcm instead. The hdmi pcm supports HD/HBR audio passthrough and relays the IEC958 status bits on to the kernel driver so it can switch to HBR audio packets if needed. The control hook is marked optional because kernels before 5.14 didn't expose the IEC958 controls. The default pcm never worked as dmix doesn't support the iec958 format. So drop dmix, only use plug and softvol, and use the hdmi pcm for iec958 formatting. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- src/conf/cards/vc4-hdmi.conf | 78 ++++++++++++++++++++++-------------- 1 file changed, 48 insertions(+), 30 deletions(-) diff --git a/src/conf/cards/vc4-hdmi.conf b/src/conf/cards/vc4-hdmi.conf index 027804a1..af87b3a8 100644 --- a/src/conf/cards/vc4-hdmi.conf +++ b/src/conf/cards/vc4-hdmi.conf @@ -3,36 +3,9 @@ # subframe conversion # -<confdir:pcm/front.conf> +<confdir:pcm/hdmi.conf> -vc4-hdmi.pcm.front.0 { - @args [ CARD ] - @args.CARD { - type string - } - type hw - card $CARD -} - -# default with dmix -vc4-hdmi.pcm.default { - @args [ CARD ] - @args.CARD { - type string - } - type asym - playback.pcm { - type plug - slave.pcm { - @func concat - strings [ "dmix:" $CARD ] - } - } -} - -<confdir:pcm/iec958.conf> - -vc4-hdmi.pcm.iec958.0 { +vc4-hdmi.pcm.hdmi.0 { @args [ CARD AES0 AES1 AES2 AES3 ] @args.CARD { type string @@ -53,12 +26,57 @@ vc4-hdmi.pcm.iec958.0 { slave { format IEC958_SUBFRAME_LE pcm { - type plug + type hooks slave.pcm { type hw card $CARD + device 0 + } + hooks.0 { + type ctl_elems + hook_args [ + { + name "IEC958 Playback Default" + interface PCM + optional true + lock true + preserve true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + ] } } } status [ $AES0 $AES1 $AES2 $AES3 ] + hdmi_mode true +} + +# default with plug and softvol +vc4-hdmi.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + type softvol + slave.pcm { + @func concat + strings [ + "cards.vc4-hdmi.pcm.hdmi.0:" + "CARD=" $CARD "," + "AES0=0x04," # IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE + "AES1=0x82," # IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER + "AES2=0x00," # IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC + "AES3=0x01" # IEC958_AES3_CON_FS_NOTID (iec958 plugin will fill in actual rate) + ] + } + control { + name "PCM Playback Volume" + card $CARD + } + } + } } -- 2.30.2

1 0

[PATCH v2] ASoC: ti: davinci-mcasp: Add dma-type for bcdma
by Jai Luthra 29 Apr '22

29 Apr '22

From: Jayesh Choudhary <j-choudhary(a)ti.com> Set DMA type for ti-bcdma controller for AM62-SK. Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 2c146b91fca3..377be2e2b6ee 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2047,6 +2047,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) return PCM_SDMA; else if (strstr(tmp, "udmap")) return PCM_UDMA; + else if (strstr(tmp, "bcdma")) + return PCM_UDMA; return PCM_EDMA; } -- 2.17.1

2 1

[PATCH] soundwire: qcom: adjust autoenumeration timeout
by Srinivas Kandagatla 29 Apr '22

29 Apr '22

Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index da1ad7ebb1aa..432e5cb9a4d2 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -105,7 +105,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS (HZ/10) #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.21.0

2 2

[PATCH 00/10] ASoC: SOF: IPC Abstraction for FW loading
by Ranjani Sridharan 29 Apr '22

29 Apr '22

This series introduces IPC abstraction for FW loading in the SOF driver in preparation for supporting the new IPC version in the SOF firmware. Peter Ujfalusi (10): ASoC: SOF: Introduce IPC dependent ops for firmware handling, loading ASoC: SOF: ipc3-loader: Implement firmware parsing and loading ASoC: SOF: ipc: Add check for fw_loader ops ASoC: SOF: loader: Switch to use the fw_loader ops ASoC: SOF: amd: renoir: Do not set the load_module ops ASoC: SOF: imx: Do not set the load_module ops ASoC: SOF: Intel: bdw/byt/pci-tng: Do not set the load_module ops ASoC: SOF: mediatek: mt8195: Do not set the load_module ops ASoC: SOF: loader: Remove snd_sof_parse_module_memcpy() as it is not used ASoC: SOF: loader: Call optional query_fw_configuration on first boot sound/soc/sof/Makefile | 2 +- sound/soc/sof/amd/renoir.c | 3 - sound/soc/sof/imx/imx8.c | 6 +- sound/soc/sof/imx/imx8m.c | 3 +- sound/soc/sof/intel/bdw.c | 3 - sound/soc/sof/intel/byt.c | 6 - sound/soc/sof/intel/pci-tng.c | 3 - sound/soc/sof/ipc.c | 57 +-- sound/soc/sof/ipc3-loader.c | 416 ++++++++++++++++++++++ sound/soc/sof/ipc3-priv.h | 24 ++ sound/soc/sof/ipc3.c | 62 +++- sound/soc/sof/loader.c | 471 +------------------------ sound/soc/sof/mediatek/mt8195/mt8195.c | 2 - sound/soc/sof/sof-priv.h | 24 +- 14 files changed, 539 insertions(+), 543 deletions(-) create mode 100644 sound/soc/sof/ipc3-loader.c create mode 100644 sound/soc/sof/ipc3-priv.h -- 2.25.1

2 11

[PATCH] ASoC: hdmi-codec: Add option for ELD bypass
by Sugar Zhang 29 Apr '22

29 Apr '22

This patch allow users to enable "ELD Bypass" who don't care anything from EDID Link Data. Currently, this driver gets ELD(from EDID) to constraint channels and rates. Unfortunately, EDID is not always valid, maybe caused by the fragile HDMI port or cable, in this situation, the max features are limited to 48kHz stereo. So, add this option to allow user to select the manual way to output audio as expected. such as multi-channels LPCM(7.1), or HBR bitstream for these sink devices. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/codecs/hdmi-codec.c | 44 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 42 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b07607a..3b78aa8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -275,6 +275,7 @@ struct hdmi_codec_priv { unsigned int chmap_idx; struct mutex lock; bool busy; + bool eld_bypass; struct snd_soc_jack *jack; unsigned int jack_status; u8 iec_status[AES_IEC958_STATUS_SIZE]; @@ -427,6 +428,44 @@ static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } +static int hdmi_codec_eld_bypass_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int hdmi_codec_eld_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = hcp->eld_bypass; + + return 0; +} + +static int hdmi_codec_eld_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + hcp->eld_bypass = ucontrol->value.integer.value[0]; + + return 0; +} + +#define HDMI_CODEC_ELD_BYPASS_DECL(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = hdmi_codec_eld_bypass_info, .get = hdmi_codec_eld_bypass_get, \ + .put = hdmi_codec_eld_bypass_put, } + static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -447,7 +486,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, goto err; } - if (tx && hcp->hcd.ops->get_eld) { + if (tx && !hcp->eld_bypass && hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); if (ret) @@ -770,6 +809,7 @@ static struct snd_kcontrol_new hdmi_codec_controls[] = { .info = hdmi_eld_ctl_info, .get = hdmi_eld_ctl_get, }, + HDMI_CODEC_ELD_BYPASS_DECL("ELD Bypass"), }; static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, @@ -854,7 +894,7 @@ static void plugged_cb(struct device *dev, bool plugged) struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); if (plugged) { - if (hcp->hcd.ops->get_eld) { + if (!hcp->eld_bypass && hcp->hcd.ops->get_eld) { hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); } -- 2.7.4

2 2

[PATCH v4] dt-bindings: dsp: mediatek: add mt8195 dsp document
by YC Hung 29 Apr '22

29 Apr '22

From: "YC Hung" <yc.hung(a)mediatek.com> This patch adds mt8195 dsp document. The dsp is used for Sound Open Firmware driver node. It includes registers, clocks, memory regions, and mailbox for dsp. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v3: Fix patch v3 error : v3 only provide difference between v3 and v2. Changes since v2: Remove useless watchdog interrupt. Add commit message more detail description. Changes since v1: Rename yaml file name as mediatek,mt8195-dsp.yaml Refine descriptions for mailbox, memory-region and drop unused labels in examples. --- .../bindings/dsp/mediatek,mt8195-dsp.yaml | 105 ++++++++++++++++++ 1 file changed, 105 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml diff --git a/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml new file mode 100644 index 000000000000..b7e68b0dfa13 --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/mediatek,mt8195-dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek mt8195 DSP core + +maintainers: + - YC Hung <yc.hung(a)mediatek.com> + +description: | + Some boards from mt8195 contain a DSP core used for + advanced pre- and post- audio processing. + +properties: + compatible: + const: mediatek,mt8195-dsp + + reg: + items: + - description: Address and size of the DSP Cfg registers + - description: Address and size of the DSP SRAM + + reg-names: + items: + - const: cfg + - const: sram + + clocks: + items: + - description: mux for audio dsp clock + - description: 26M clock + - description: mux for audio dsp local bus + - description: default audio dsp local bus clock source + - description: clock gate for audio dsp clock + - description: mux for audio dsp access external bus + + clock-names: + items: + - const: adsp_sel + - const: clk26m_ck + - const: audio_local_bus + - const: mainpll_d7_d2 + - const: scp_adsp_audiodsp + - const: audio_h + + power-domains: + maxItems: 1 + + mboxes: + items: + - description: ipc reply between host and audio DSP. + - description: ipc request between host and audio DSP. + + mbox-names: + items: + - const: mbox0 + - const: mbox1 + + memory-region: + items: + - description: dma buffer between host and DSP. + - description: DSP system memory. + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - memory-region + - power-domains + - mbox-names + - mboxes + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + dsp@10803000 { + compatible = "mediatek,mt8195-dsp"; + reg = <0x10803000 0x1000>, + <0x10840000 0x40000>; + reg-names = "cfg", "sram"; + clocks = <&topckgen 10>, //CLK_TOP_ADSP + <&clk26m>, + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 136>, //CLK_TOP_MAINPLL_D7_D2 + <&scp_adsp 0>, //CLK_SCP_ADSP_AUDIODSP + <&topckgen 34>; //CLK_TOP_AUDIO_H + clock-names = "adsp_sel", + "clk26m_ck", + "audio_local_bus", + "mainpll_d7_d2", + "scp_adsp_audiodsp", + "audio_h"; + memory-region = <&adsp_dma_mem_reserved>, + <&adsp_mem_reserved>; + power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP + mbox-names = "mbox0", "mbox1"; + mboxes = <&adsp_mailbox0>, <&adsp_mailbox1>; + }; -- 2.18.0

4 6

[PATCH 11/37] sound: add HAS_IOPORT dependencies
by Niklas Schnelle 29 Apr '22

29 Apr '22

In a future patch HAS_IOPORT=n will result in inb()/outb() and friends not being declared. We thus need to add HAS_IOPORT as dependency for those drivers using them. Co-developed-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Niklas Schnelle <schnelle(a)linux.ibm.com> --- sound/drivers/Kconfig | 5 +++++ sound/pci/Kconfig | 45 ++++++++++++++++++++++++++++++++----------- 2 files changed, 39 insertions(+), 11 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index ca4cdf666f82..4d250e619786 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,10 +1,12 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART tristate + depends on HAS_IOPORT select SND_RAWMIDI config SND_OPL3_LIB tristate + depends on HAS_IOPPORT select SND_TIMER select SND_HWDEP select SND_SEQ_DEVICE if SND_SEQUENCER != n @@ -128,6 +130,7 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" + depends on HAS_IOPORT select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -152,6 +155,7 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" + depends on HAS_IOPORT select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -167,6 +171,7 @@ config SND_SERIAL_U16550 config SND_MPU401 tristate "Generic MPU-401 UART driver" + depends on HAS_IOPORT select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index a55836225401..51c1f38530f6 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -26,7 +26,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -51,7 +51,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -96,6 +96,7 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -110,6 +111,7 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -123,6 +125,7 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -157,7 +160,7 @@ config SND_AZT3328 select SND_RAWMIDI select SND_AC97_CODEC select SND_TIMER - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -193,6 +196,7 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" + depends on HAS_IOPORT select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -205,6 +209,7 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -221,6 +226,7 @@ config SND_OXYGEN_LIB config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -246,6 +252,7 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC @@ -257,6 +264,7 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select FW_LOADER @@ -290,6 +298,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on X86_32 || MIPS || COMPILE_TEST + depends on HAS_IOPORT select SND_PCM select SND_AC97_CODEC help @@ -307,6 +316,7 @@ config SND_CS5535AUDIO config SND_CTXFI tristate "Creative Sound Blaster X-Fi" + depends on HAS_IOPORT select SND_PCM help If you want to use soundcards based on Creative Sound Blastr X-Fi @@ -462,13 +472,14 @@ config SND_INDIGODJX config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" + depends on HAS_IOPORT select FW_LOADER select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC select SND_TIMER select SND_SEQ_DEVICE if SND_SEQUENCER != n - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -491,7 +502,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -501,6 +512,7 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" + depends on HAS_IOPORT select SND_RAWMIDI select SND_PCM help @@ -511,6 +523,7 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC help @@ -525,7 +538,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -537,7 +550,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -569,6 +582,7 @@ config SND_ES1968_RADIO config SND_FM801 tristate "ForteMedia FM801" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -624,7 +638,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -640,6 +654,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER @@ -712,7 +727,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -753,6 +768,7 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" + depends on HAS_IOPORT select FW_LOADER select SND_PCM select SND_HWDEP @@ -764,6 +780,7 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" + depends on HAS_IOPORT select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -808,6 +825,7 @@ config SND_RME9652 config SND_SE6X tristate "Studio Evolution SE6X" depends on SND_OXYGEN=n && SND_VIRTUOSO=n # PCI ID conflict + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -830,7 +848,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -842,7 +860,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. @@ -852,6 +870,7 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -863,6 +882,7 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" + depends on HAS_IOPORT select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -873,6 +893,7 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 66/100/200 (Xonar)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -889,6 +910,7 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" + depends on HAS_IOPORT select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -898,6 +920,7 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC -- 2.32.0

1 1

[PATCH] ASoC: rk3328: fix disabling mclk on pclk probe failure
by Nicolas Frattaroli 29 Apr '22

29 Apr '22

If preparing/enabling the pclk fails, the probe function should unprepare and disable the previously prepared and enabled mclk, which it doesn't do. This commit rectifies this. Fixes: c32759035ad2 ("ASoC: rockchip: support ACODEC for rk3328") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/codecs/rk3328_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 758d439e8c7a..86b679cf7aef 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -481,7 +481,7 @@ static int rk3328_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(rk3328->pclk); if (ret < 0) { dev_err(&pdev->dev, "failed to enable acodec pclk\n"); - return ret; + goto err_unprepare_mclk; } base = devm_platform_ioremap_resource(pdev, 0); -- 2.36.0

3 3

[PATCH -next] ASoC: SOF: sof-pci-dev: fix missing pci_release_regions() on error in sof_pci_probe()
by Yang Yingliang 29 Apr '22

29 Apr '22

Fix the missing pci_release_regions() before return from sof_pci_probe() in the error handling case. Fixes: 4bfbbb76e82e ("ASOC: SOF: pci: add ipc_type override for Intel IPC4 tests") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/sof-pci-dev.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index cd90da7c92c2..3b0978b02f9f 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -217,12 +217,14 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) desc->ipc_default, sof_pci_ipc_type); if (sof_pci_ipc_type >= SOF_IPC_TYPE_COUNT) { dev_err(dev, "invalid request value %d\n", sof_pci_ipc_type); - return -EINVAL; + ret = -EINVAL; + goto out; } if (!(BIT(sof_pci_ipc_type) & desc->ipc_supported_mask)) { dev_err(dev, "invalid request value %d, supported mask is %#x\n", sof_pci_ipc_type, desc->ipc_supported_mask); - return -EINVAL; + ret = -EINVAL; + goto out; } sof_pdata->ipc_type = sof_pci_ipc_type; } @@ -291,6 +293,8 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) /* call sof helper for DSP hardware probe */ ret = snd_sof_device_probe(dev, sof_pdata); + +out: if (ret) pci_release_regions(pci); -- 2.25.1

2 1

Re: [v4 13/18] ASoC: mediatek: mt8186: add platform driver
by Mark Brown 29 Apr '22

29 Apr '22

On Thu, Apr 28, 2022 at 05:33:50PM +0800, Jiaxin Yu wrote: > Add mt8186 platform and affiliated driver. > > Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> > --- > sound/soc/mediatek/Kconfig | 44 + > sound/soc/mediatek/Makefile | 1 + > sound/soc/mediatek/mt8186/Makefile | 22 + > sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ > .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ > sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3005 +++++++++++++++++ > .../mediatek/mt8186/mt8186-interconnection.h | 69 + > .../soc/mediatek/mt8186/mt8186-misc-control.c | 294 ++ > .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + > .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + > sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 ++++++++++++++++ > 11 files changed, 6920 insertions(+) This looks mostly good though it is enormous so I might've missed some things. The patch series is already very large but it might still be worth splitting this up a bit more, perhaps split the code and data tables/register definitions into separate patches? A few relatively minor issues with the controls. > +/* this order must match reg bit amp_div_ch1/2 */ > +static const char * const mt8186_sgen_amp_str[] = { > + "1/128", "1/64", "1/32", "1/16", "1/8", "1/4", "1/2", "1" }; > +static const char * const mt8186_sgen_mute_str[] = { > + "Off", "On" > +}; On/off controls should be normal Switch controls not enums so userspace can display things sensibly. > +static int mt8186_sgen_set(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); > + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); > + struct mt8186_afe_private *afe_priv = afe->platform_priv; > + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; > + int mode; > + int mode_idx; > + > + if (ucontrol->value.enumerated.item[0] >= e->items) > + return -EINVAL; ... > + 0x3f << INNER_LOOP_BACK_MODE_SFT); > + } > + > + afe_priv->sgen_mode = mode; > + > + return 0; > +} This should return 1 if the value is different from the previous value so event generation works, please run mixer-test against a system using the driver to help spot issues like this. The same issue applies to at least some of the other custom controls. > +static int mt8186_sgen_mute_set(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); > + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); > + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; > + int mute; > + > + if (ucontrol->value.enumerated.item[0] >= e->items) > + return -EINVAL; > + > + mute = ucontrol->value.integer.value[0]; > + > + dev_dbg(afe->dev, "%s(), kcontrol name %s, mute %d\n", > + __func__, kcontrol->id.name, mute); > + > + if (strcmp(kcontrol->id.name, SGEN_MUTE_CH1_KCONTROL_NAME) == 0) { > + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, > + MUTE_SW_CH1_MASK_SFT, > + mute << MUTE_SW_CH1_SFT); > + } else { > + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, > + MUTE_SW_CH2_MASK_SFT, > + mute << MUTE_SW_CH2_SFT); > + } > + > + return 0; > +} I can't tell why some of these are done with custom code rather than using a normal SOC_SINGLE()?

2 1

[PATCH 0/2] Let userspace know when snd-hda-intel needs i915
by Mauro Carvalho Chehab 29 Apr '22

29 Apr '22

Currently, kernel/module annotates module dependencies when request_symbol is used, but it doesn't cover more complex inter-driver dependencies that are subsystem and/or driver-specific. In the case of hdmi sound, depending on the CPU/GPU, sometimes the snd_hda_driver can talk directly with the hardware, but sometimes, it uses the i915 driver. When the snd_hda_driver uses i915, it should first be unbind/rmmod, as otherwise trying to unbind/rmmod the i915 driver cause driver issues, as as reported by CI tools with different GPU models: https://intel-gfx-ci.01.org/tree/drm-tip/IGT_6415/fi-tgl-1115g4/igt@core_ho… https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_11495/bat-adlm-1/igt@i915_m… In the past, just a few CPUs were doing such bindings, but this issue now applies to all "modern" Intel CPUs that have onboard graphics, as well as to the newer discrete GPUs. With the discrete GPU case, the HDA controller is physically separate and requires i915 to power on the hardware for all hardware access. In this case, the issue is hit basicly 100% of the time. With on-board graphics, i915 driver is needed only when the display codec is accessed. If i915 is unbind during runtime suspend, while snd-hda-intel is still bound, nothing bad happens, but unbinding i915 on other situations may also cause issues. So, add support at kernel/modules to allow snd-hda drivers to properly annotate when a dependency on a DRM driver dependencies exists, and add a call to such new function at the snd-hda driver when it successfully binds into the DRM driver. This would allow userspace tools to check and properly remove the audio driver before trying to remove or unbind the GPU driver. It should be noticed that this series conveys the hidden module dependencies. Other changes are needed in order to allow removing or unbinding the i915 driver while keeping the snd-hda-intel driver loaded/bound. With that regards, there are some discussions on how to improve this at alsa-devel a while back: https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/190099… So, future improvements on both in i915 and the audio drivers could be made. E.g. with discrete GPUs, it's the only codec of the card, so it seems feasible to detach the ALSA card if i915 is bound (using infra made for VGA switcheroo), but, until these improvements are done and land in upstream, audio drivers needs to be unbound if i915 driver goes unbind. Yet, even if such fixes got merged, this series is still needed, as it makes such dependencies more explicit and easier to debug. PS.: This series was generated against next-20220428. Luis/Takashi/Daniel/David, If OK for you, I would prefer to have such patches applied via the drm-tip tree once reviewed, in order to make easier to use them by some patches I'm preparing to improve the CI tests that use i915 unbind logic. Mauro Carvalho Chehab (2): module: add a function to add module references ALSA: hda - identify when audio is provided by a video driver include/linux/module.h | 7 +++++++ kernel/module/main.c | 31 +++++++++++++++++++++++++++++++ sound/hda/hdac_component.c | 8 ++++++++ 3 files changed, 46 insertions(+) -- 2.35.1

1 1

conf: Use LFS calls when reading config files
by GitHub pull_request - opened 29 Apr '22

29 Apr '22

alsa-project/alsa-lib pull request #223 was opened from dos1: Although at first glance it doesn't seem useful to support config files larger than 2GB, LFS also influences inode size. Without this, 32-bit libasound may be unable to read config files on filesystems with 64-bit inodes, such as Btrfs or NFS. Signed-off-by: Sebastian Krzyszkowiak <dos(a)dosowisko.net> Request URL : https://github.com/alsa-project/alsa-lib/pull/223 Patch URL : https://github.com/alsa-project/alsa-lib/pull/223.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0