August 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 05 Feb '26

05 Feb '26

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

4 5

ALSA 1.2.8 release
by Jaroslav Kysela 13 Jan '26

13 Jan '26

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 12

[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 02 Jan '26

02 Jan '26

I own a MOTU Midi Express XT 8x8 interface which is connected through the parallel port. Recently I upgraded my system and the new motherboard does not have an ISA parallel port anymore... So I bought a PCI parallel port extension card - hoping that it will work with the ALSA driver (mtpav). So I changed the interrupt an port address in modules.conf but the driver is no longer recognized as an ALSA driver even though it loads without errors. As the device still perfectly fits my needs I would like to modify the driver to get it working again. Can anyone give me advice what should be changed in the drivers code to make it work on a pci port? Thanks Holger

4 14

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 29 Dec '25

29 Dec '25

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

8 9

[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 05 Dec '25

05 Dec '25

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

4 8

[PATCH] ASoC: SOF: Intel: Retry codec probing if it fails
by Kai Vehmanen 04 Dec '25

04 Dec '25

From: Hui Wang <hui.wang(a)canonical.com> On the latest Lenovo Thinkstation laptops, we often experience the speaker failure after rebooting, check the dmesg, we could see: sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5 The analogue codec on the machine is ALC287, then we designed a testcase to reboot and check the codec probing result repeatedly, we found the analogue codec probing always failed at least once within several minutes to several hours (roughly 1 reboot per min). This issue happens on all laptops of this Thinkstation model, but with legacy HDA driver, we couldn't reproduce this issue on those laptops. And so far, this issue is not reproduced on machines which don't belong to this model. We tried to make the hda_dsp_ctrl_init_chip() same as hda_intel_init_chip() which is the controller init routine in the legacy HDA driver, but it didn't help. We found when issue happens, the resp is -1, and if we let driver re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the desktop and all audio function work well. Here adding codec probing retries to 3 times, it could fix the issue on this Thinkstation model, and it doesn't bring impact to other machines. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) Notes: - patch for-5.16 on top of for-5.16 8a724d5f60 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..13cd96e6724a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -22,6 +22,7 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 +#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) @@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; - int ret; + int ret, retry = 0; + + do { + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); - mutex_lock(&hbus->core.cmd_mutex); - snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); - snd_hdac_bus_get_response(&hbus->core, address, &resp); - mutex_unlock(&hbus->core.cmd_mutex); if (resp == -1) return -EIO; dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", base-commit: 8a724d5f60904084e09de2685ff9e77bc8a40815 -- 2.34.1

3 2

[PATCH] ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
by Sylwester Nawrocki 20 Nov '25

20 Nov '25

The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 60 ++++++++++++++++++++++++++++----------- 1 file changed, 44 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..038be667c1a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, -- 2.17.1

4 3

[PATCH v1] slimbus: qcom-ngd-ctrl: Add stream disable support
by Viken Dadhaniya 24 Sep '25

24 Sep '25

Currently slimbus driver doesn't support stream disable callback, it only supports stream enable callback. In slimbus usecase, client is switching to new frequency with same channel and calling enable stream callback for new frequency but DSP subsystem is crashing as we are switching to new frequency with same channel without disabling stream for older frequency. Ideally, before switching to another frequency, client should call disable stream callback and then enable stream for newer frequency. Hence add support to disable stream via qcom_slim_ngd_disable_stream(). Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 70 +++++++++++++++++++++++++++++++++ drivers/slimbus/slimbus.h | 13 ++++++ 2 files changed, 83 insertions(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index e0b21f0f79c1..d952827d2e12 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited +// Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. #include <linux/irq.h> #include <linux/kernel.h> @@ -1084,6 +1085,74 @@ static int qcom_slim_ngd_enable_stream(struct slim_stream_runtime *rt) return ret; } +static int qcom_slim_ngd_disable_stream(struct slim_stream_runtime *rt) +{ + struct slim_device *sdev = rt->dev; + struct slim_controller *ctrl = sdev->ctrl; + struct slim_val_inf msg = {0}; + u8 wbuf[SLIM_MSGQ_BUF_LEN]; + u8 rbuf[SLIM_MSGQ_BUF_LEN]; + struct slim_msg_txn txn = {0,}; + int i, ret; + + txn.mt = SLIM_MSG_MT_DEST_REFERRED_USER; + txn.dt = SLIM_MSG_DEST_LOGICALADDR; + txn.la = SLIM_LA_MGR; + txn.ec = 0; + txn.msg = &msg; + txn.msg->num_bytes = 0; + txn.msg->wbuf = wbuf; + txn.msg->rbuf = rbuf; + + for (i = 0; i < rt->num_ports; i++) { + struct slim_port *port = &rt->ports[i]; + + if (txn.msg->num_bytes == 0) { + wbuf[txn.msg->num_bytes++] = (u8)(SLIM_CH_REMOVE << 6) + | (sdev->laddr & 0x1f); + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(&sdev->dev, "Fail to allocate TID\n"); + return -ENXIO; + } + wbuf[txn.msg->num_bytes++] = txn.tid; + } + wbuf[txn.msg->num_bytes++] = port->ch.id; + } + + txn.mc = SLIM_USR_MC_CHAN_CTRL; + txn.rl = txn.msg->num_bytes + 4; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + return ret; + } + + txn.mc = SLIM_USR_MC_RECONFIG_NOW; + txn.msg->num_bytes = 2; + wbuf[1] = sdev->laddr; + txn.rl = txn.msg->num_bytes + 4; + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(ctrl->dev, "Fail to allocate TID ret:%d\n", ret); + return ret; + } + + wbuf[0] = txn.tid; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + } + + return ret; +} + static int qcom_slim_ngd_get_laddr(struct slim_controller *ctrl, struct slim_eaddr *ea, u8 *laddr) { @@ -1642,6 +1711,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) ctrl->ctrl.clkgear = SLIM_MAX_CLK_GEAR; ctrl->ctrl.get_laddr = qcom_slim_ngd_get_laddr; ctrl->ctrl.enable_stream = qcom_slim_ngd_enable_stream; + ctrl->ctrl.disable_stream = qcom_slim_ngd_disable_stream; ctrl->ctrl.xfer_msg = qcom_slim_ngd_xfer_msg; ctrl->ctrl.wakeup = NULL; ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..21543d125614 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,6 +1,7 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. */ #ifndef _DRIVERS_SLIMBUS_H @@ -316,6 +317,18 @@ enum slim_transport_protocol { SLIM_PROTO_EXT_HALF_DUP, }; +/* + * enum slim_ch_control: Channel control. + * Activate will schedule channel and/or group of channels in the TDM frame. + * Suspend will keep the schedule but data-transfer won't happen. + * Remove will remove the channel/group from the TDM frame. + */ +enum slim_ch_control { + SLIM_CH_ACTIVATE, + SLIM_CH_SUSPEND, + SLIM_CH_REMOVE, +}; + /** * struct slim_stream_runtime - SLIMbus stream runtime instance * -- 2.17.1

2 2

retire_capture_urb: Corrected urb data len
by Sean Anderson 01 Aug '25

01 Aug '25

Hi all, I have a "FiiO DigiHug USB Audio" sound card (1852:7022) [3]. I have had no problems with the audio, but I did notice a large number of message like retire_capture_urb: 4992 callbacks suppressed in my dmesg [1]. This is caused by the "Corrected urb data len." warning. The patch adding this warning [2] makes it seem like this warning should be an uncommon occurance. However, based on the number of suppressed callbacks, this seems to be happening at a rate of around 500 Hz. Is this buggy hardware? Or is this a bug in the driver? Does there need to be a quirk? Or perhaps the warning above should be a debug instead? I'm using 5.15.0-46-generic. I know, I know; this is a vendor kernel. However, I went through the commits for sound/usb/pcm.c and didn't see any major differences. I also previously saw this issue on 5.4, so this probably isn't a regression. --Sean [1] I would have liked to post the actual message, but I was unable to figure out how to disable ratelimiting. I tried setting /proc/sys/kernel/printk_ratelimit to 0, /proc/sys/kernel/printk_ratelimit_burst to 5000, and /proc/sys/kernel/printk to 8. However, nothing seemed to have any effect (except that the callbacks suppressed messages were disabled). I find this very strange, given that the callbacks suppressed message uses KERN_WARN, and so does dev_warn_ratelimited. [2] https://lore.kernel.org/all/4B28A659.6070303@ladisch.de/T/ [3] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1852 GYROCOM C&C Co., LTD idProduct 0x7022 bcdDevice 0.01 iManufacturer 1 FiiO iProduct 2 DigiHug USB Audio iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0182 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 58 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0012 1x 18 bytes bInterval 32 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 FiiO USB DAC-E10 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x003e bInCollection 2 baInterfaceNr(0) 2 baInterfaceNr(1) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 9 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 16 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 9 bControlSize 1 bmaControls(0) 0x01 Mute Control bmaControls(1) 0x02 Volume Control bmaControls(2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x2001 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

2 8

[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

2 2

[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 17 Jun '25

17 Jun '25

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

2 3

[PATCH] ASoC: amd: acp: Fix support for a Huawei Matebook laptop
by Marian Postevca 17 Jun '25

17 Jun '25

Previous commit that added support for Huawei MateBook D16 2021 with Ryzen 4600H (HVY-WXX9 M1010) was incomplete. To activate support for this laptop, the DMI table in acp3x-es83xx machine driver must also be updated. Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop") Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index f85b85ea4be9..2b0aa270a3e9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -354,6 +354,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = { }, .driver_data = (void *)(ES83XX_ENABLE_DMIC|ES83XX_48_MHZ_MCLK), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + .driver_data = (void *)(ES83XX_ENABLE_DMIC), + }, { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), -- 2.43.0

2 1

[PATCH 0/1] Swapped l/r channels on ThinkBook 13x Gen4
by MadEnvel 28 May '25

28 May '25

Earlier commits introduced audio support for Lenovo ThinBook 13x Gen 4 that needed a patch due to missing _DSD config. Unfortunately, it looks like the left and right channels in the config are swapped. This patch fixes channel assignment. Tested on Lenovo ThinkBook 13x Gen4. MadEnvel (1): Fix swapped left and right channels for Lenovo ThinBook 13x Gen4 sound sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- 2.45.2

4 4

[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

2 23

[PATCH 00/34] address all -Wunused-const warnings
by Arnd Bergmann 27 Jan '25

27 Jan '25

From: Arnd Bergmann <arnd(a)arndb.de> Compilers traditionally warn for unused 'static' variables, but not if they are constant. The reason here is a custom for C++ programmers to define named constants as 'static const' variables in header files instead of using macros or enums. In W=1 builds, we get warnings only static const variables in C files, but not in headers, which is a good compromise, but this still produces warning output in at least 30 files. These warnings are almost all harmless, but also trivial to fix, and there is no good reason to warn only about the non-const variables being unused. I've gone through all the files that I found using randconfig and allmodconfig builds and created patches to avoid these warnings, with the goal of retaining a clean build once the option is enabled by default. Unfortunately, there is one fairly large patch ("drivers: remove incorrect of_match_ptr/ACPI_PTR annotations") that touches 34 individual drivers that all need the same one-line change. If necessary, I can split it up by driver or by subsystem, but at least for reviewing I would keep it as one piece for the moment. Please merge the individual patches through subsystem trees. I expect that some of these will have to go through multiple revisions before they are picked up, so anything that gets applied early saves me from resending. Arnd Arnd Bergmann (31): powerpc/fsl-soc: hide unused const variable ubsan: fix unused variable warning in test module platform: goldfish: remove ACPI_PTR() annotations i2c: pxa: hide unused icr_bits[] variable 3c515: remove unused 'mtu' variable tracing: hide unused ftrace_event_id_fops Input: synaptics: hide unused smbus_pnp_ids[] array power: rt9455: hide unused rt9455_boost_voltage_values efi: sysfb: don't build when EFI is disabled clk: ti: dpll: fix incorrect #ifdef checks apm-emulation: hide an unused variable sisfb: hide unused variables dma/congiguous: avoid warning about unused size_bytes leds: apu: remove duplicate DMI lookup data iio: ad5755: hook up of_device_id lookup to platform driver greybus: arche-ctrl: move device table to its right location lib: checksum: hide unused expected_csum_ipv6_magic[] sunrpc: suppress warnings for unused procfs functions comedi: ni_atmio: avoid warning for unused device_ids[] table iwlegacy: don't warn for unused variables with DEBUG_FS=n drm/komeda: don't warn for unused debugfs files firmware: qcom_scm: mark qcom_scm_qseecom_allowlist as __maybe_unused crypto: ccp - drop platform ifdef checks usb: gadget: omap_udc: remove unused variable isdn: kcapi: don't build unused procfs code cpufreq: intel_pstate: hide unused intel_pstate_cpu_oob_ids[] net: xgbe: remove extraneous #ifdef checks Input: imagis - remove incorrect ifdef checks sata: mv: drop unnecessary #ifdef checks ASoC: remove incorrect of_match_ptr/ACPI_PTR annotations spi: remove incorrect of_match_ptr annotations drivers: remove incorrect of_match_ptr/ACPI_PTR annotations kbuild: always enable -Wunused-const-variable Krzysztof Kozlowski (1): Input: stmpe-ts - mark OF related data as maybe unused arch/powerpc/sysdev/fsl_msi.c | 2 + drivers/ata/sata_mv.c | 64 +++++++++---------- drivers/char/apm-emulation.c | 5 +- drivers/char/ipmi/ipmb_dev_int.c | 2 +- drivers/char/tpm/tpm_ftpm_tee.c | 2 +- drivers/clk/ti/dpll.c | 10 ++- drivers/comedi/drivers/ni_atmio.c | 2 +- drivers/cpufreq/intel_pstate.c | 2 + drivers/crypto/ccp/sp-platform.c | 14 +--- drivers/dma/img-mdc-dma.c | 2 +- drivers/firmware/efi/Makefile | 3 +- drivers/firmware/efi/sysfb_efi.c | 2 - drivers/firmware/qcom/qcom_scm.c | 2 +- drivers/fpga/versal-fpga.c | 2 +- .../gpu/drm/arm/display/komeda/komeda_dev.c | 8 --- drivers/hid/hid-google-hammer.c | 6 +- drivers/i2c/busses/i2c-pxa.c | 2 +- drivers/i2c/muxes/i2c-mux-ltc4306.c | 2 +- drivers/i2c/muxes/i2c-mux-reg.c | 2 +- drivers/iio/dac/ad5755.c | 1 + drivers/input/mouse/synaptics.c | 2 + drivers/input/touchscreen/imagis.c | 4 +- drivers/input/touchscreen/stmpe-ts.c | 2 +- drivers/input/touchscreen/wdt87xx_i2c.c | 2 +- drivers/isdn/capi/Makefile | 3 +- drivers/isdn/capi/kcapi.c | 7 +- drivers/leds/leds-apu.c | 3 +- drivers/mux/adg792a.c | 2 +- drivers/net/ethernet/3com/3c515.c | 3 - drivers/net/ethernet/amd/xgbe/xgbe-platform.c | 8 --- drivers/net/ethernet/apm/xgene-v2/main.c | 2 +- drivers/net/ethernet/hisilicon/hns_mdio.c | 2 +- drivers/net/wireless/intel/iwlegacy/4965-rs.c | 15 +---- drivers/net/wireless/intel/iwlegacy/common.h | 2 - drivers/platform/goldfish/goldfish_pipe.c | 2 +- drivers/power/supply/rt9455_charger.c | 2 + drivers/regulator/pbias-regulator.c | 2 +- drivers/regulator/twl-regulator.c | 2 +- drivers/regulator/twl6030-regulator.c | 2 +- drivers/rtc/rtc-fsl-ftm-alarm.c | 2 +- drivers/scsi/hisi_sas/hisi_sas_v1_hw.c | 2 +- drivers/scsi/hisi_sas/hisi_sas_v2_hw.c | 2 +- drivers/spi/spi-armada-3700.c | 2 +- drivers/spi/spi-img-spfi.c | 2 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-meson-spifc.c | 2 +- drivers/spi/spi-orion.c | 2 +- drivers/spi/spi-pic32-sqi.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-rockchip.c | 2 +- drivers/spi/spi-s3c64xx.c | 2 +- drivers/spi/spi-st-ssc4.c | 2 +- drivers/staging/greybus/arche-apb-ctrl.c | 1 + drivers/staging/greybus/arche-platform.c | 9 +-- drivers/staging/pi433/pi433_if.c | 2 +- drivers/tty/serial/amba-pl011.c | 6 +- drivers/tty/serial/ma35d1_serial.c | 2 +- drivers/usb/gadget/udc/omap_udc.c | 10 +-- drivers/video/fbdev/sis/init301.c | 3 +- kernel/dma/contiguous.c | 2 +- kernel/trace/trace_events.c | 4 ++ lib/checksum_kunit.c | 2 + lib/test_ubsan.c | 2 +- net/sunrpc/cache.c | 10 +-- scripts/Makefile.extrawarn | 1 - sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/samsung/aries_wm8994.c | 2 +- 69 files changed, 121 insertions(+), 169 deletions(-) -- 2.39.2 Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: Christophe Leroy <christophe.leroy(a)csgroup.eu> Cc: Damien Le Moal <dlemoal(a)kernel.org> Cc: Jiri Kosina <jikos(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Corey Minyard <minyard(a)acm.org> Cc: Peter Huewe <peterhuewe(a)gmx.de> Cc: Jarkko Sakkinen <jarkko(a)kernel.org> Cc: Tero Kristo <kristo(a)kernel.org> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Ian Abbott <abbotti(a)mev.co.uk> Cc: H Hartley Sweeten <hsweeten(a)visionengravers.com> Cc: Srinivas Pandruvada <srinivas.pandruvada(a)linux.intel.com> Cc: Len Brown <lenb(a)kernel.org> Cc: "Rafael J. Wysocki" <rafael(a)kernel.org> Cc: John Allen <john.allen(a)amd.com> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Ard Biesheuvel <ardb(a)kernel.org> Cc: Bjorn Andersson <andersson(a)kernel.org> Cc: Moritz Fischer <mdf(a)kernel.org> Cc: Liviu Dudau <liviu.dudau(a)arm.com> Cc: Benjamin Tissoires <benjamin.tissoires(a)redhat.com> Cc: Andi Shyti <andi.shyti(a)kernel.org> Cc: Michael Hennerich <michael.hennerich(a)analog.com> Cc: Peter Rosin <peda(a)axentia.se> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Markuss Broks <markuss.broks(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)foss.st.com> Cc: Lee Jones <lee(a)kernel.org> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Shyam Sundar S K <Shyam-sundar.S-k(a)amd.com> Cc: Iyappan Subramanian <iyappan(a)os.amperecomputing.com> Cc: Yisen Zhuang <yisen.zhuang(a)huawei.com> Cc: Stanislaw Gruszka <stf_xl(a)wp.pl> Cc: Kalle Valo <kvalo(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Cc: Xiang Chen <chenxiang66(a)hisilicon.com> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Neil Armstrong <neil.armstrong(a)linaro.org> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Vaibhav Hiremath <hvaibhav.linux(a)gmail.com> Cc: Alex Elder <elder(a)kernel.org> Cc: Jiri Slaby <jirislaby(a)kernel.org> Cc: Jacky Huang <ychuang3(a)nuvoton.com> Cc: Helge Deller <deller(a)gmx.de> Cc: Christoph Hellwig <hch(a)lst.de> Cc: Robin Murphy <robin.murphy(a)arm.com> Cc: Steven Rostedt <rostedt(a)goodmis.org> Cc: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Kees Cook <keescook(a)chromium.org> Cc: Trond Myklebust <trond.myklebust(a)hammerspace.com> Cc: Anna Schumaker <anna(a)kernel.org> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: openipmi-developer(a)lists.sourceforge.net Cc: linux-integrity(a)vger.kernel.org Cc: linux-omap(a)vger.kernel.org Cc: linux-clk(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: dmaengine(a)vger.kernel.org Cc: linux-efi(a)vger.kernel.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-fpga(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-input(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-amlogic(a)lists.infradead.org Cc: linux-rockchip(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: greybus-dev(a)lists.linaro.org Cc: linux-staging(a)lists.linux.dev Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: iommu(a)lists.linux.dev Cc: linux-trace-kernel(a)vger.kernel.org Cc: kasan-dev(a)googlegroups.com Cc: linux-hardening(a)vger.kernel.org Cc: linux-nfs(a)vger.kernel.org Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-sound(a)vger.kernel.org

7 7

[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

3 6

[PATCH v4 0/9] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 29 Dec '24

29 Dec '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Changes in v4: - Replaced manual regulator toggles with SND_SOC_DAPM_REGULATOR_SUPPLY, both for existing mic bias regulators and for the new headset mic bias regulator - Link to v3: https://lore.kernel.org/r/20240519-midas-wm1811-gpio-jack-v3-0-0c1736144c0e… Changes in v3: - Re-added pipe (|) character to samsung,headset-button-threshold-microvolt description to prevent it from being parsed as a mapping (fixes syntax error) - Split out "ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection" into two patches - Link to v2: https://lore.kernel.org/r/20240508-midas-wm1811-gpio-jack-v2-0-b4d36cd02c6e… Changes in v2: - Added vendor prefix to threshold properties - Added separate headset mic bias regulator - Changed some cases of dev_err + return with return dev_err_probe - Added an extra patch to replace some previous dev_err + return cases with dev_err_probe - Moved tab3 DTS wm1811 codec config changes to separate commit --- Artur Weber (9): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use SND_SOC_DAPM_REGULATOR_SUPPLY for bias regulators ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection ARM: dts: samsung: exynos4212-tab3: Add MCLK2 clock to WM1811 codec config ARM: dts: samsung: exynos4212-tab3: Drop interrupt from WM1811 codec .../bindings/sound/samsung,midas-audio.yaml | 33 ++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 31 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 348 +++++++++++++++++---- 4 files changed, 343 insertions(+), 71 deletions(-) --- base-commit: a38297e3fb012ddfa7ce0321a7e5a8daeb1872b6 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

3 12

hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

3 5

[PATCH] dt-bindings: sound:atmel-at91sam9g20ek: convert bindings to json-schema
by Balakrishnan Sambath 10 Dec '24

10 Dec '24

Convert atmel-at91sam9g20ek-wm8731-audio DT binding to yaml based json-schema.Change file name to match json-scheme naming. Signed-off-by: Balakrishnan Sambath <balakrishnan.s(a)microchip.com> --- .../bindings/sound/atmel,at91sam9g20ek-wm8731.yaml | 60 ++++++++++++++++++++++ .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 ---------- 2 files changed, 60 insertions(+), 26 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml new file mode 100644 index 000000000000..f6330707fe1b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,at91sam9g20ek-wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9g20ek wm8731 audio complex + +maintainers: + - Balakrishnan Sambath <balakrishnan.s(a)microchip.com> + +properties: + compatible: + const: atmel,at91sam9g20ek-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + atmel,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: A list of the connections between audio components. + minItems: 2 + items: + enum: + # Board Connectors + - "Ext Spk" + - "Int MIC" + + # CODEC Pins + - LHPOUT + - MICIN + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of WM8731 audio codec + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,at91sam9g20ek-wm8731-audio"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + atmel,model = "wm8731 @ AT91SAMG20EK"; + atmel,audio-routing = + "Ext Spk", "LHPOUT", + "Int MIC", "MICIN"; + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt deleted file mode 100644 index 9c5a9947b64d..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt +++ /dev/null @@ -1,26 +0,0 @@ -* Atmel at91sam9g20ek wm8731 audio complex - -Required properties: - - compatible: "atmel,at91sam9g20ek-wm8731-audio" - - atmel,model: The user-visible name of this sound complex. - - atmel,audio-routing: A list of the connections between audio components. - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8731 audio codec -Optional properties: - - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt - -Example: -sound { - compatible = "atmel,at91sam9g20ek-wm8731-audio"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pck0_as_mck>; - - atmel,model = "wm8731 @ AT91SAMG20EK"; - - atmel,audio-routing = - "Ext Spk", "LHPOUT", - "Int MIC", "MICIN"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8731>; -}; --- base-commit: 0dd3ee31125508cd67f7e7172247f05b7fd1753a change-id: 20240214-at91sam9g20ek-wm8731-yaml-404ddddbb91a Best regards, -- Balakrishnan Sambath <balakrishnan.s(a)microchip.com>

5 5

[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

3 16

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

5 18

[PATCH] ALSA: hda/realtek: tas2781: Fix ROG ALLY X audio
by Jonathan LoBue 11 Oct '24

11 Oct '24

This patch enables the TI TAS2781 amplifier SoC for the ASUS ROG ALLY X. This is a design change from the original ASUS ROG ALLY, creating the need for this patch. All other Realtek Codec settings seem to be re-used from the original ROG ALLY design (on the ROG ALLY X). This patch maintains the previous settings for the Realtek codec portion, but enables the I2C binding for the TI TAS2781 amplifier (instead of the Cirrus CS35L41 amp used on the original ASUS ROG ALLY). One other requirement must be met for audio to work on the ASUS ROG ALLY X. A proper firmware file in the correct location with a proper symlink. We had reached out to TI engineers and confirmed that the firmware found in the Windows' driver package has a GPL license. Bazzite Github is hosting this firmware file for now until proper linux-firmware upstreaming can occur. https://github.com/ublue-os/bazzite This firmware file should be placed in /usr/lib/firmware/ti/tas2781/TAS2XXX1EB3.bin with a symlink to it from /usr/lib/firmware/TAS2XXX1EB3.bin Co-developed by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Signed-off-by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Co-developed by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Antheas Kapenekakis <lkml(a)antheas.dev> Tested-by: Richard Alvarez <alvarez.richard(a)gmail.com> Tested-by: Miles Montierth <cyber_dopamine(a)intheblackmedia.com> Signed-off-by: Jonathan LoBue <jlobue10(a)gmail.com> --- 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 480e82df7a4c..9bc39c2ee6b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7410,6 +7410,7 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_X, ALC294_FIXUP_ASUS_ALLY_PINS, ALC294_FIXUP_ASUS_ALLY_VERBS, ALC294_FIXUP_ASUS_ALLY_SPEAKER, @@ -8877,6 +8878,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS }, + [ALC294_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, [ALC294_FIXUP_ASUS_ALLY_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10313,6 +10320,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.46.0

4 5

Add stacked and parallel memories support in spi-nor
by Mahapatra, Amit Kumar 10 Oct '24

10 Oct '24

Hello Everyone, As https://lore.kernel.org/all/20231125092137.2948-8-amit.kumar-mahapatra@amd.… thread discussing the addition of stacked and parallel mode support in SPI-NOR has become too long and difficult to follow. Therefore, I'm starting a new email thread to focus on the DT binding changes and the overall implementation architecture for adding stacked and parallel (multi-CS) support in the Kernel. Please share your input on this approach. Background: The AMD QSPI controller supports two advanced connection modes(Stacked and Dual Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Dual Parallel: Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… Mirochip QSPI controller also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… Initially in [1] [2] [3] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. [1] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [2] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [3]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [4] [5] [6] describe the two flash devices as being one. [4] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [5] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [6] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… SPI core changes to support the above bindings were added through the following commit: https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e91… While adding the support in SPI-NOR we received some inputs/suggestions from Tudor, based on that I am planning to add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of the spi-nor and located in mtd/spi-nor/. This layer would perform the following tasks: - During probing, store information from all the connected flashes, whether in stacked or parallel mode, and present it as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of in spi-nor/core.c - There will very minimal change in spinor/core.c as the stacked & parallel handling will be managed by the new layer on top of SPI-NOR. - In stacked mode, the new layer will select the appropriate spi-nor flash based on the address provided by the MTD layer during flash operations. - Manage flash crossover operations in stacked mode. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. For implementing this the current DT binding[1] [2] [3] need to be updated as follows. stacked-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each stacked flash will have its own flash node. This approach allows flashes of different makes and sizes to be stacked together, as each flash will be probed individually. - Each of the flash node will have its own "reg" property that will contain its physical CS. - The stacked-memories DT bindings will contain the phandles of the flash nodes connected in stacked mode. - The first flash node will contain the mtd partition that would have the cross over memory staring at a memory location in the first flash and ending at some memory location of the 2nd flash - The new layer will update the mtd->size and other mtd_info parameters after both the flashes are probed and will call mtd_device_register with the combined information. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00>; stacked-memories = <&flash@0 &flash@1>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; }; } flash@1 { compatible = "jedec,spi-nor" reg = <0x01>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-1"; reg = <0x0 0x800000>; }; } } parallel-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each flash connected in parallel mode should be identical and will have one flash node for both the flash devices. - The "reg" prop will contain the physical CS number for both the connected flashes. - The new layer will double the mtd-> size and register it with the mtd layer. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00 0x01>; parallel-memories; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; } }; } Regards, Amit

2 6

[PATCH -next 0/5] sound/soc: fix some macro definitions and usages
by Hongbo Li 01 Oct '24

01 Oct '24

Many variable in macro are not used as we used macro_check script to detect and mamually check, let us address these issues. Hongbo Li (5): sound/soc: fix macro definition on TWL4030_OUTPUT_PGA sound/soc remove unused substream in macro soc_component_mark_pop sound/soc remove unused substream in macro soc_dai_mark_pop sound/soc remove unused substream in macro soc_link_mark_pop sound/soc: fix macro definition on STM_SAI_HAS_EXT_SYNC sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/soc-component.c | 14 +++++++------- sound/soc/soc-dai.c | 10 +++++----- sound/soc/soc-link.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 2 +- 5 files changed, 24 insertions(+), 24 deletions(-) -- 2.34.1

2 7

[PATCH v2 0/2] {ASoC,media}: constify snd_soc_component_driver when used as read-only
by Javier Carrasco 01 Oct '24

01 Oct '24

Declare `snd_soc_component_driver` as const to move their declarations to read-only sections for the drivers that do not modify the struct after its declaration. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. To: Tim Harvey <tharvey(a)gateworks.com> To: Mauro Carvalho Chehab <mchehab(a)kernel.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: Ray Jui <rjui(a)broadcom.com> To: Scott Branden <sbranden(a)broadcom.com> To: Broadcom internal kernel review list <bcm-kernel-feedback-list(a)broadcom.com> To: Shenghao Ding <shenghao-ding(a)ti.com> To: Kevin Lu <kevin-lu(a)ti.com> To: Baojun Xu <baojun.xu(a)ti.com> To: Olivier Moysan <olivier.moysan(a)foss.st.com> To: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> To: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> To: Alexandre Torgue <alexandre.torgue(a)foss.st.com> To: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> To: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: linux-media(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> Changes in v2: - drop cs43130 and sti-sas, as they modifiy the struct in the probe function. - Link to v1: https://lore.kernel.org/r/20240725-const_snd_soc_component_driver-v1-0-3d7e… --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 12 files changed, 13 insertions(+), 13 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

2 3

Re: [PATCH] [RFC] ASoC: Conditional PCM support
by Pierre-Louis Bossart 17 Sep '24

17 Sep '24

On 8/21/24 12:18, Cezary Rojewski wrote: > Conditional PCM (condpcm) helps facilitate modern audio usecases such as > Echo Cancellations and Noise Reduction. These are not invoked by the > means of userspace application opening an endpoint (FrontEnd) but are a > "side effect" of selected PCMs running simultaneously e.g.: if both > Speaker (source) and Microphone Array (sink) are running, reference > data from the Speaker and take it into account when processing capture > for better voice command detection ratio. The point about dependencies between capture/playback usages is certainly valid, and we've faced it multiple times for SOF - and even before in the mobile phone days. I am not convinced however that the graph management suggested here solves the well-known DPCM routing problems? See notes in no specific order below. I am not following what the 'source' and 'sink' concepts refer to in this context. It looks like you are referring to regular PCM devices, i.e. Front Ends in soc-pcm parlance but examples and code make references to Back Ends. There are also complicated cases where the amplifiers can provide an echo reference for AEC and I/V sensing for speaker protection. You would want to capture both even if there's no capture happening at the userspace level. This is a well-know DPCM routing issue where we have to rely on a Front-End being opened and some tags in UCM to deal with loose coupling. It would help if you added precisions on your assumptions of where the processing takes place. In some cases Echo cancellation is handled in userspace, others in SOC firmware and others externally in a codec. The notion of source/sink is also problematic when the same BE provides two sources of information that will be split, again same problem with amplifier feedback being used for two separate functions. What happens if you have multiple sinks for one source? Same for the cases where the mic input is split multiple ways with different processing added on different PCM capture devices, e.g. for WebRTC there's an ask for a raw input, an AEC-processed input and AEC+NS-processed input. That's typically implemented with two splitters, the echo reference would be used by an intermediate node inside a firmware graph, not at the DAI/BE or host/FE levels, and such intermediate nodes are usually not handled by soc-pcm. We really need more than the notion of FE and BE, a two-layer solution is no longer sufficient. The other thing that looks weird is the dependency on both sink and source sharing a common state. For a noise reduction there are cases where you'd want the mic input to be stored in a history buffer so that the noise parameters can be estimated as soon as the actual capture starts. > Which PCMs are needed for given conditional PCM to be spawned is > determinated by the driver when registering the condpcm. Presumably such links should be described by a topology file? It would be odd for a driver to have to guess when to connect processing elements. > The functionality was initially proposed for the avs-driver [1] and, > depending on feedback and review may either go back into avs -or- become > a ASoC-core feature. Implementation present here is an example of how > such functionality could look and work on the ASoC side. Compared to > what was provided initially, the patch carries simplified version of the > feature: no priority/overriding for already running conditional PCMs. > Whatever is spawned is treated as a non-conflicting entity. > > Assumptions and design decisions: > > - existence and outcome of condpcm operations is entirely optional and > shall not impact the runtime flow of PCMs that spawned given condpcm, > e.g.: fail in cpcm->hw_params() shall not impact fe->hw_params() or > be->hw_params() negatively. Think of it as of debugfs. Useful? Yes. > Required for system to operate? No. that's debatable, if the AEC setup isn't successful then is the functionality implemented correctly? My take is no, don't fail silently if the AEC doesn't work. If this functionality is listed as a product requirement then it cannot be treated as a debugfs optional thing. Exhibit A for this is the countless cases where validation reported a problem with a path remaining active or conversely not being setup, or a voice quality issue. Those are not optional... > - a condpcm is a runtime entity that's audio format independent - since > certain FE/BEs are its dependencies already, that's no need to do > format ruling twice. Driver may still do custom checks thanks to > ->match() operation. > > - a condpcm allows for additional processing of data that flows from > data-source - a substream instance acting as data provider - > to sink - a substream acting as data consumer. At the same time, > regardless of substream->stream, given substream may act as data > source for one condpcm and data sink for another, simultaneously. > > - while condpcm's behaviour mimics standard PCM one, there is no > ->open() and ->close() - FE/BEs are treated as operational starting > with successful ->hw_params(), when hw_ruling is done and hardware is > configured. > > - cpcm->prepare() gets called only when both data source and sink are > prepared > - cpcm->trigger(START) gets called only when both data source and sink > are running > - cpcm->trigger(STOP) gets called when either data source or sink is > stopped > > Simplified state machine: > > | > register_condpcm() | > v > +--+-------------+ > | DISCONNECTED |<-+ > +--+-------------+ | > | | > condpcm_hw_params() | | > v | > +--+-------------+ | > | SETUP | | condpcm_hw_free() > +--+-------------+ | > | | > condpcm_prepare() | | > v | > +--+----+--------+ | > | PREPARED |--+ > +--+----------+--+ > | ^ > condpcm_start() | | condpcm_stop() > v | > +--+----------+--+ > | RUNNING | > +----------------+ > > What's missing? > I've not covered the locking part yet. While some operations are covered > by default thanks to snd_soc_dpcm_mutex(), it is insufficient. > If feature goes back to the avs-driver, then I'm set due to path->mutex. > > The locking is one of the reasons I'm leaning towards leaving the > condpcm within the avs-driver. For soc_condpcm_find_match() to be > precise and do no harm, a lock must prepend the list_for_each_entry(). > Entries (substreams) of that list may be part of number of different > components and the search may negatively impact runtime flow of > substreams that do not care about condpcms at all. > > Has this been tested? > > Unit-like only. Typical case below with avs_condpcm_ops representing > bunch of stubs with printfs. > > static struct snd_soc_condpcm_pred pred1 = { > .card_name = "ssp0-loopback", > .link_name = "SSP0-Codec", /* BE */ > .direction = SNDRV_PCM_STREAM_PLAYBACK, > }; > > static struct snd_soc_condpcm_pred pred2 = { > .card_name = "hdaudioB0D2", > .link_name = "HDMI1", /* FE */ > .direction = SNDRV_PCM_STREAM_PLAYBACK, > }; It's not intuitive to follow what HDMI and SSP might have to do with each other, nor why one is a BE and one is an FE? If I follow the code below, the SSP loopback is a source feeds into an HDMI sink, and SSP is a BE and HDMI an FE? Confusing example... > static void avs_condpcms_register(struct avs_dev *adev) > { > (...) > snd_soc_register_condpcm(&pred1, &pred2, &avs_condpcm_ops, adev); > } > > [1]: https://lore.kernel.org/alsa-devel/20211208111301.1817725-21-cezary.rojewsk… > > Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> > --- > include/sound/pcm.h | 1 + > include/sound/soc.h | 65 ++++++++ > sound/core/pcm.c | 1 + > sound/soc/Makefile | 2 +- > sound/soc/soc-condpcm.c | 348 ++++++++++++++++++++++++++++++++++++++++ > sound/soc/soc-condpcm.h | 17 ++ > sound/soc/soc-core.c | 2 + > sound/soc/soc-pcm.c | 11 ++ > 8 files changed, 446 insertions(+), 1 deletion(-) > create mode 100644 sound/soc/soc-condpcm.c > create mode 100644 sound/soc/soc-condpcm.h > > diff --git a/include/sound/pcm.h b/include/sound/pcm.h > index ac8f3aef9205..7e635b3103a2 100644 > --- a/include/sound/pcm.h > +++ b/include/sound/pcm.h > @@ -482,6 +482,7 @@ struct snd_pcm_substream { > struct list_head link_list; /* linked list member */ > struct snd_pcm_group self_group; /* fake group for non linked substream (with substream lock inside) */ > struct snd_pcm_group *group; /* pointer to current group */ > + struct list_head cpcm_candidate_node; It wouldn't hurt to describe what 'candidate' might mean here? > /* -- assigned files -- */ > int ref_count; > atomic_t mmap_count; > diff --git a/include/sound/soc.h b/include/sound/soc.h > index e844f6afc5b5..32a6b5092192 100644 > --- a/include/sound/soc.h > +++ b/include/sound/soc.h > @@ -426,6 +426,7 @@ struct snd_jack; > struct snd_soc_card; > struct snd_soc_pcm_stream; > struct snd_soc_ops; > +struct snd_soc_condpcm; > struct snd_soc_pcm_runtime; > struct snd_soc_dai; > struct snd_soc_dai_driver; > @@ -1154,6 +1155,64 @@ struct snd_soc_card { > #define for_each_card_widgets_safe(card, w, _w) \ > list_for_each_entry_safe(w, _w, &card->widgets, list) > > +/* Conditional PCM operations called by soc-pcm.c. */ > +struct snd_soc_condpcm_ops { > + int (*match)(struct snd_soc_condpcm *, struct snd_pcm_substream *, > + struct snd_pcm_substream *); > + int (*hw_params)(struct snd_soc_condpcm *, struct snd_pcm_hw_params *); > + int (*hw_free)(struct snd_soc_condpcm *); > + int (*prepare)(struct snd_soc_condpcm *, struct snd_pcm_substream *); > + int (*trigger)(struct snd_soc_condpcm *, struct snd_pcm_substream *, int); > +}; > + > +/** > + * struct snd_soc_condpcm_pred - Predicate, describes source or sink (substream) > + * dependency for given conditional PCM. > + * > + * @card_name: Name of card owning substream to find. > + * @link_name: Name of DAI LINK owning substream to find. > + * @direction: Whether its SNDRV_PCM_STREAM_PLAYBACK or CAPTURE. > + */ > +struct snd_soc_condpcm_pred { > + const char *card_name; Please tell me the runtimes and links are in the same card... If not, there's all kinds of power management and probe/remove issues... > + const char *link_name; dai link name? > + int direction; > +}; > + > +/** > + * struct snd_soc_condpcm - Conditional PCM descriptor. > + * > + * @ops: custom PCM operations. > + * @preds: predicates for identifying source and sink for given conditional PCM. predicate is a verb and a noun, not clear what you are trying to document. > + * > + * @source: substreaming acting as a data source, assigned at runtime. > + * @sink: substreaming acting as a data sink, assigned at runtime. > + * @state: current runtime state. presumably this state is already defined that the state of sink/source? > + * > + * @source_node: list navigation for rtd->source_list. > + * @sink_node: list navigation for rtd->sink_list. > + * @node: list navigation for condpcm_list (soc-condpcm.c). > + */ > +struct snd_soc_condpcm { > + const struct snd_soc_condpcm_ops *ops; > + struct snd_soc_condpcm_pred preds[2]; > + > + struct snd_pcm_substream *source; > + struct snd_pcm_substream *sink; > + snd_pcm_state_t state; > + void *private_data; > + > + /* Condpcms navigation for the pcm runtime. */ > + struct list_head source_node; > + struct list_head sink_node; > + struct list_head node; > +}; > + > +int snd_soc_unregister_condpcm(struct snd_soc_condpcm *cpcm); > +struct snd_soc_condpcm *snd_soc_register_condpcm(struct snd_soc_condpcm_pred *source, > + struct snd_soc_condpcm_pred *sink, > + const struct snd_soc_condpcm_ops *ops, > + void *private_data); > > static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) > { > @@ -1161,6 +1220,10 @@ static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) > } > > /* SoC machine DAI configuration, glues a codec and cpu DAI together */ > +/* > + * @source_cpcm_list: List of condpcms this pcm is source for. > + * @sink_cpcm_list: List of condpcms this pcm is sink for. > + */ > struct snd_soc_pcm_runtime { > struct device *dev; > struct snd_soc_card *card; > @@ -1172,6 +1235,8 @@ struct snd_soc_pcm_runtime { > /* Dynamic PCM BE runtime data */ > struct snd_soc_dpcm_runtime dpcm[SNDRV_PCM_STREAM_LAST + 1]; > struct snd_soc_dapm_widget *c2c_widget[SNDRV_PCM_STREAM_LAST + 1]; > + struct list_head cpcm_source_list; > + struct list_head cpcm_sink_list; > > long pmdown_time; > > diff --git a/sound/core/pcm.c b/sound/core/pcm.c > index dc37f3508dc7..12243cecaa11 100644 > --- a/sound/core/pcm.c > +++ b/sound/core/pcm.c > @@ -663,6 +663,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) > substream->pstr = pstr; > substream->number = idx; > substream->stream = stream; > + INIT_LIST_HEAD(&substream->cpcm_candidate_node); > sprintf(substream->name, "subdevice #%i", idx); > substream->buffer_bytes_max = UINT_MAX; > if (prev == NULL) > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 775bb38c2ed4..8004de7c500c 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -1,6 +1,6 @@ > # SPDX-License-Identifier: GPL-2.0 > snd-soc-core-y := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o > -snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o > +snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o soc-condpcm.o > snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o > > ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) > diff --git a/sound/soc/soc-condpcm.c b/sound/soc/soc-condpcm.c > new file mode 100644 > index 000000000000..786b3fafd714 > --- /dev/null > +++ b/sound/soc/soc-condpcm.c > @@ -0,0 +1,348 @@ > +// SPDX-License-Identifier: GPL-2.0-only > +// > +// Conditional-PCM management > +// > +// Copyright(c) 2024 Intel Corporation > +// > +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> > +// > + > +#include <linux/export.h> > +#include <linux/list.h> > +#include <linux/slab.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include "soc-condpcm.h" > + > +/* > + * condpcm_list - Stores all registered conditional pcms. > + * condpcm_candidate_list - Stores all substreams that passed hw_params() step. > + */ > +static LIST_HEAD(condpcm_list); > +static LIST_HEAD(condpcm_candidate_list); > + > +static bool dpcm_prepare_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state == SND_SOC_DPCM_STATE_PREPARE || > + state == SND_SOC_DPCM_STATE_START || > + state == SND_SOC_DPCM_STATE_PAUSED; > +} > + > +static bool dpcm_hw_params_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state >= SND_SOC_DPCM_STATE_HW_PARAMS && > + state < SND_SOC_DPCM_STATE_HW_FREE; > +} > + > +static bool dpcm_start_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state == SND_SOC_DPCM_STATE_START || > + state == SND_SOC_DPCM_STATE_PAUSED; > +} > + > +static int soc_condpcm_hw_params(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(cpcm->source); > + struct snd_soc_pcm_runtime *rtd2 = snd_soc_substream_to_rtd(cpcm->sink); how are the 'params' defined? I read above " a condpcm is a runtime entity that's audio format independent - since certain FE/BEs are its dependencies already, that's no need to do format ruling twice. " That doesn't tell us how this 'params' is determined. This is important for cases where the speaker output is e.g. 2ch 48kHz and the mic input is 4ch 96kHz. If this condpcm is not managed by any usersapce action, then what is the logic for selecting the settings in 'params'? > + int ret; > + > + ret = cpcm->ops->hw_params(cpcm, params); > + if (ret) > + return ret; > + > + list_add_tail(&cpcm->source_node, &rtd->cpcm_source_list); > + list_add_tail(&cpcm->sink_node, &rtd2->cpcm_sink_list); > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + return 0; > +} There's also the well-known problem that hw_params can be called multiple times. Here this wouldn't work with the same source/sink added multiple times in a list. > + > +static void soc_condpcm_hw_free(struct snd_soc_condpcm *cpcm) > +{ > + cpcm->ops->hw_free(cpcm); > + list_del(&cpcm->source_node); > + list_del(&cpcm->sink_node); > + cpcm->state = SNDRV_PCM_STATE_DISCONNECTED; > +} > + > +static void soc_condpcm_prepare(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream) > +{ > + int ret; > + > + ret = cpcm->ops->prepare(cpcm, substream); > + if (!ret) > + cpcm->state = SNDRV_PCM_STATE_PREPARED; > +} you probably need to look at the xruns and resume cases, where prepare() is used for vastly different purposes. > + > +static void soc_condpcm_start(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + ret = cpcm->ops->trigger(cpcm, substream, cmd); > + if (ret) > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + else > + cpcm->state = SNDRV_PCM_STATE_RUNNING; > +} > + > +static void soc_condpcm_stop(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + ret = cpcm->ops->trigger(cpcm, substream, cmd); > + if (ret || cmd != SNDRV_PCM_TRIGGER_PAUSE_PUSH) > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + else > + cpcm->state = SNDRV_PCM_STATE_PREPARED; > +} > + > +void snd_soc_condpcms_prepare(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Prepare conditional pcms only if source and sink are both preprared. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) { > + if (cpcm->state == SNDRV_PCM_STATE_SETUP && > + dpcm_prepare_done(cpcm->sink)) > + soc_condpcm_prepare(cpcm, substream); > + } > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) { > + if (cpcm->state == SNDRV_PCM_STATE_SETUP && > + dpcm_prepare_done(cpcm->source)) > + soc_condpcm_prepare(cpcm, substream); > + } > +} > + > +static void soc_condpcms_start(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Start conditional pcms only if source and sink are both running. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) { > + if (cpcm->state == SNDRV_PCM_STATE_PREPARED && > + dpcm_start_done(cpcm->sink)) > + soc_condpcm_start(cpcm, substream, cmd); > + } > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) { > + if (cpcm->state == SNDRV_PCM_STATE_PREPARED && > + dpcm_start_done(cpcm->source)) > + soc_condpcm_start(cpcm, substream, cmd); > + } > +} > + > +static void soc_condpcms_stop(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Stop conditional pcms if either source or sink is not running. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) > + if (cpcm->state == SNDRV_PCM_STATE_RUNNING) > + soc_condpcm_stop(cpcm, substream, cmd); > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) > + if (cpcm->state == SNDRV_PCM_STATE_RUNNING) > + soc_condpcm_stop(cpcm, substream, cmd); > +} > + > +void snd_soc_condpcms_trigger(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > + soc_condpcms_start(rtd, substream, cmd); > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > + soc_condpcms_stop(rtd, substream, cmd); > + break; > + } > +} > + > +void snd_soc_condpcms_teardown(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream) > +{ > + struct snd_soc_condpcm *cpcm, *save; > + > + list_del(&substream->cpcm_candidate_node); > + > + /* Free all condpcms this substream spawned. */ > + list_for_each_entry_safe(cpcm, save, &rtd->cpcm_source_list, source_node) > + soc_condpcm_hw_free(cpcm); > + list_for_each_entry_safe(cpcm, save, &rtd->cpcm_sink_list, sink_node) > + soc_condpcm_hw_free(cpcm); > +} > + > +static bool soc_condpcm_test(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int dir) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + struct snd_soc_condpcm_pred *pred = &cpcm->preds[dir]; > + > + return pred->direction == substream->stream && > + !strcmp(pred->card_name, rtd->card->name) && > + !strcmp(pred->link_name, rtd->dai_link->name); > +} > + > +static struct snd_pcm_substream * > +soc_condpcm_find_match(struct snd_soc_condpcm *cpcm, struct snd_pcm_substream *ignore, int dir) > +{ > + struct snd_pcm_substream *substream; > + > + list_for_each_entry(substream, &condpcm_candidate_list, cpcm_candidate_node) { > + if (substream == ignore) > + continue; > + > + /* Only substreams that passed hw_params() are valid candidates. */ > + if (!dpcm_hw_params_done(substream)) > + continue; > + > + if (soc_condpcm_test(cpcm, substream, dir)) > + return substream; > + } > + > + return NULL; > +} > + > +static int soc_condpcm_walk(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, int dir) > +{ > + /* Temporary source/sink cache. */ > + struct snd_pcm_substream *substreams[2]; > + struct snd_soc_condpcm *cpcm; > + int ret; > + > + substreams[dir] = substream; > + > + list_for_each_entry(cpcm, &condpcm_list, node) { > + if (cpcm->state != SNDRV_PCM_STATE_DISCONNECTED) > + continue; > + > + /* Does this cpcm match @substream? */ > + if (!soc_condpcm_test(cpcm, substream, dir)) > + continue; > + > + /* Find pair for the @substream. */ > + substreams[!dir] = soc_condpcm_find_match(cpcm, substream, !dir); > + if (!substreams[!dir]) > + continue; > + > + /* Allow driver to have the final word. */ > + ret = cpcm->ops->match(cpcm, substreams[0], substreams[1]); > + if (ret) > + continue; > + cpcm->source = substreams[0]; > + cpcm->sink = substreams[1]; > + > + ret = soc_condpcm_hw_params(cpcm, params); > + if (ret) { > + cpcm->source = NULL; > + cpcm->sink = NULL; > + return ret; > + } > + } > + > + return 0; > +} > + > +/* Called by soc-pcm.c after each successful hw_params(). */ > +int snd_soc_condpcms_walk_all(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + int ret; > + > + list_add_tail(&substream->cpcm_candidate_node, &condpcm_candidate_list); > + > + /* Spawn all condpcms this substream is the missing source of. */ > + ret = soc_condpcm_walk(rtd, substream, params, SNDRV_PCM_STREAM_CAPTURE); > + if (ret) > + return ret; > + > + /* Spawn all condpcms this substream is the missing sink of. */ > + return soc_condpcm_walk(rtd, substream, params, SNDRV_PCM_STREAM_PLAYBACK); > +} Are loops supported? Is the order between capture and playback intentional? Is the notion of playback/capture even relevant when trying to add loopbacks? Lots of questions...

4 7

[PATCH 000/113] ALSA: add snd_pcm_is_playback/capture() macro
by Kuninori Morimoto 17 Sep '24

17 Sep '24

Hi ALSA-ML Cc Staging-ML, USB-ML Many drivers are using below code to know the Sound direction. if (direction == SNDRV_PCM_STREAM_PLAYBACK) This patch-set add snd_pcm_is_playback/capture() macro to handle it. Thank you for your help !! Best regards --- Kuninori Morimoto

4 119

[PATCH 0/1] Add support for primary mi2s on SM8250
by Jens Reidel 11 Sep '24

11 Sep '24

This patch adds support for the primary mi2s interface on devices using SM8250 audio drivers. Tested on SM7150 (xiaomi-davinci). SM7150 sound is close to SM8250 and we intend to use it as a fallback in the future. To: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux(a)mainlining.org Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.46.0

4 5

[Patch v5 00/12] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 11 Sep '24

11 Sep '24

This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners. Piotr Wojtaszczyk (12): dt-bindings: dma: pl08x: Add dma-cells description dt-bindings: dma: Add lpc32xx DMA mux binding ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ARM: dts: lpc32xx: Use simple-mfd for clock control block ARM: dts: lpc32xx: Add missing dma properties ARM: dts: lpc32xx: Add missing i2s properties clk: lpc32xx: initialize regmap using parent syscon dmaengine: Add dma router for pl08x in LPC32XX SoC ARM: lpc32xx: Remove pl08x platform data in favor for device tree mtd: rawnand: lpx32xx: Request DMA channels using DT entries ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs i2x: pnx: Fix potential deadlock warning from del_timer_sync() call in isr .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 49 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 20 + arch/arm/boot/dts/nxp/lpc/lpc32xx.dtsi | 53 ++- arch/arm/mach-lpc32xx/phy3250.c | 54 --- drivers/clk/Kconfig | 1 + drivers/clk/nxp/clk-lpc32xx.c | 26 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 48 +-- drivers/mtd/nand/raw/lpc32xx_mlc.c | 26 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 26 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 375 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 72 ++++ 19 files changed, 993 insertions(+), 130 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/nxp,lpc3220-dmamux.yaml create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml create mode 100644 drivers/dma/lpc32xx-dmamux.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c -- 2.25.1

7 19

[PATCH] soundwire: stream: fix programming slave ports for non-continous port maps
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

Two bitmasks in 'struct sdw_slave_prop' - 'source_ports' and 'sink_ports' - define which ports to program in sdw_program_slave_port_params(). The masks are used to get the appropriate data port properties ('struct sdw_get_slave_dpn_prop') from an array. Bitmasks can be non-continuous or can start from index different than 0, thus when looking for matching port property for given port, we must iterate over mask bits, not from 0 up to number of ports. This fixes allocation and programming slave ports, when a source or sink masks start from further index. Fixes: f8101c74aa54 ("soundwire: Add Master and Slave port programming") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 7aa4900dcf31..f275143d7b18 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - u8 num_ports; + unsigned long mask; int i; if (direction == SDW_DATA_DIR_TX) { - num_ports = hweight32(slave->prop.source_ports); + mask = slave->prop.source_ports; dpn_prop = slave->prop.src_dpn_prop; } else { - num_ports = hweight32(slave->prop.sink_ports); + mask = slave->prop.sink_ports; dpn_prop = slave->prop.sink_dpn_prop; } - for (i = 0; i < num_ports; i++) { + for_each_set_bit(i, &mask, 32) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

6 15

How to handle complex Codec2Codec
by Kuninori Morimoto 09 Sep '24

09 Sep '24

Hi ALSA ML ALSA SoC is supporting Codec2Codec connection, and Audio Graph Card2 is supporting it. I think its image is... +------+ +--- |Codec1| <-- | +------+ | +------+ +--> |Codec2| --> +------+ This is OK, but how to handle below ? +----------------------+ |CPU | +------+ | +--[rate convert]<--| <-- |Codec1| <-- | | | +------+ | | | +------+ | +------------------>| --> |Codec2| --> +----------------------+ +------+ I think there are 2 issues. (1) How to handle [CPU] part on Codec2Codec ? (2) How to handle [rate convert] on Codec2Codec ? I guess (1) can be handled if [CPU] was worked as platform, but do you have any other idea/solution for it ? I think (2) [rate convert] needs DPCM connection, but how to use it with Codec2Codec ? Or, can we handle Codec2Codec via DPCM somehow ?? FE BE +------+ +------+ --> |Codec1| -> |Codec2| -> +------+ +------+ If we can do this, both (1) and (2) can be solved, like below ? In such case, [Codec1] need to work as [capture] when playback. But I'm not sure how to implement it... FE BE +------+ +-----------------+ +-----+ +------+ --> |Codec1|-|CPU[rate convert]| -> |dummy|-|Codec2| -> +------+ +-----------------+ +-----+ +------+ Thank you for your help !! Best regards --- Kuninori Morimoto

2 8

Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

1 3

Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

3 12

[PATCH v26 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Sep '24

04 Sep '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 45 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 71 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 338 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 110 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5820 insertions(+), 56 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 64

[PATCH] ASoC: tlv320aic32x4: Add multi endpoint support
by Marek Vasut 04 Sep '24

04 Sep '24

Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Baojun Xu <baojun.xu(a)ti.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Christoph Niedermaier <cniedermaier(a)dh-electronics.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kevin Lu <kevin-lu(a)ti.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Shenghao Ding <shenghao-ding(a)ti.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: kernel(a)dh-electronics.com Cc: linux-sound(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06dba..54ea4bc58c276 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.45.2

2 1

[PATCH v2 0/3] ASoC: dt-bindings: amlogic-sound-cards: document clocks property
by Neil Armstrong 02 Sep '24

02 Sep '24

Following an off-list discution with Jerome about fixing the following DTBs check errors: sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml# sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# sound: 'anyOf' conditional failed, one must be fixed: 'clocks' is a required property '#clock-cells' is a required property from schema $id: http://devicetree.org/schemas/clock/clock.yaml# It has been agreed documenting the clock in the sound card is a better solution than moving them to a random clock controller or consumer node which is not related to the actual meaning of those root frequencies. The patchset adds the clocks proprty to the bindings and finally adds the properties to the DT files. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes in v2: - drop clock-names in bindings & commit messages - fix new boards - drop RFS - Link to v1: https://lore.kernel.org/r/20240614-topic-amlogic-upstream-bindings-fixes-au… --- Neil Armstrong (3): ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property arm64: dts: amlogic: add clock and clock-names to sound cards .../devicetree/bindings/sound/amlogic,axg-sound-card.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 7 +++++++ arch/arm64/boot/dts/amlogic/meson-axg-s400.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-fbx8am.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-radxa-zero.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-sei510.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-u200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-x96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-cm4io.dts | 4 ++++ .../arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-mnt-reform2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-dreambox.dtsi | 6 ++++++ arch/arm64/boot/dts/amlogic/meson-g12b-gsking-x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-go-ultra.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2l.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-radxa-zero2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-ugoos-am6.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-libretech-pc.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-p23x-q20x.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-kii-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nanopi-k2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nexbox-a95x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-odroidc2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p201.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-vega-s95.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-hub.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-play2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-libretech-ac.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-p241.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-khadas-vim.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc-v2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-p212.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-vero4k.dts | 5 +++++ arch/arm64/boot/dts/amlogic/meson-gxm-khadas-vim2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-nexbox-a1.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-rbox-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-khadas-vim3.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-libretech-cottonwood.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m2-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m5.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-h96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-odroid.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-sei610.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air.dts | 4 ++++ 52 files changed, 217 insertions(+) --- base-commit: cb2361c78068ec47c3a6222d71f664b67665550e change-id: 20240614-topic-amlogic-upstream-bindings-fixes-audio-snd-card-2c642ccd2c6e Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

3 6

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *'
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: hexagon-randconfig-002-20240901 (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project 46fe36a4295f05d5d3731762e31fc4e6e99863e9) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010509.nOL6xTD7-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:548:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 548 | val = __raw_readb(PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:561:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 561 | val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' 37 | #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:574:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 574 | val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' 35 | #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:585:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 585 | __raw_writeb(value, PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:595:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 595 | __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:605:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 605 | __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ In file included from sound/core/pcm_memory.c:14: In file included from include/sound/pcm.h:15: In file included from include/linux/mm.h:2228: include/linux/vmstat.h:517:36: warning: arithmetic between different enumeration types ('enum node_stat_item' and 'enum lru_list') [-Wenum-enum-conversion] 517 | return node_stat_name(NR_LRU_BASE + lru) + 3; // skip "nr_" | ~~~~~~~~~~~ ^ ~~~ >> sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *' [-Werror,-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ include/linux/kstrtox.h:30:90: note: passing argument to parameter 'res' here 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ^ 7 warnings and 1 error generated. vim +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: sh-allmodconfig (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) compiler: sh4-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010425.YPS7cWeJ-lkp@intel.com/ All errors (new ones prefixed by >>): sound/core/pcm_memory.c: In function 'snd_pcm_lib_preallocate_proc_write': >> sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type [-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ | | | size_t * {aka unsigned int *} In file included from include/linux/kernel.h:25, from arch/sh/include/asm/fixmap.h:13, from arch/sh/include/asm/pgtable.h:22, from include/linux/pgtable.h:6, from arch/sh/include/asm/io.h:21, from include/linux/io.h:14, from sound/core/pcm_memory.c:7: include/linux/kstrtox.h:30:90: note: expected 'long unsigned int *' but argument is of type 'size_t *' {aka 'unsigned int *'} 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ~~~~~~~~~~~~~~~^~~ vim +/kstrtoul +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - edited 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was edited from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. EDIT: Other possibility to fix is to return `-ENOENT` in case where dB information was not found and handle that accordingly when iterating over items of a container. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - opened 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was opened from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoC: codecs: Switch to use dev_err_probe()
by Yang Ruibin 30 Aug '24

30 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 8 +++----- sound/soc/codecs/adau1701.c | 19 +++++++------------ sound/soc/codecs/ssm2602.c | 6 ++---- 3 files changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..d2f679d9d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -237,11 +237,9 @@ static int ad1980_soc_probe(struct snd_soc_component *component) u16 ext_status; ac97 = snd_soc_new_ac97_component(component, 0, 0); - if (IS_ERR(ac97)) { - ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; - } + if (IS_ERR(ac97)) + return dev_err_probe(component->dev, PTR_ERR(ac97), + "Failed to register AC97 component\n"); regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); if (IS_ERR(regmap)) { diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..f4c574447 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -677,10 +677,9 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, + "Failed to enable regulators\n"); /* * Let the pll_clkdiv variable default to something that won't happen @@ -798,17 +797,13 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to get regulators\n"); ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to enable regulators\n"); adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..153eb55a3 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -605,10 +605,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) int ret; ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); - if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, "Failed to issue reset\n"); regmap_register_patch(ssm2602->regmap, ssm2602_patch, ARRAY_SIZE(ssm2602_patch)); -- 2.34.1

2 1

[PATCH v2] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 30 Aug '24

30 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. On codec versions below 2.0, even though the core_id registers are available to read, the values of these registers are not unique to be able to determine the version of the codec dynamically. Add the version info into of_data, so that driver does not need to use core_id registers to get version number for such situations. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - updated commit text to add more details sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

5 5

[PATCH v1] ASoC: fsl_ssi: Simplify with dev_err_probe()
by Shen Lichuan 30 Aug '24

30 Aug '24

Use dev_err_probe() to simplify the error path and unify a message template. Using this helper is totally fine even if err is known to never be -EPROBE_DEFER. The benefit compared to a normal dev_err() is the standardized format of the error code, it being emitted symbolically and the fact that the error code is returned which allows more compact error paths. Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- sound/soc/fsl/fsl_ssi.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c4c1d9c44056..51927f89241f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1354,19 +1354,16 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi->clk = devm_clk_get(dev, "ipg"); else ssi->clk = devm_clk_get(dev, NULL); - if (IS_ERR(ssi->clk)) { - ret = PTR_ERR(ssi->clk); - dev_err(dev, "failed to get clock: %d\n", ret); - return ret; - } + if (IS_ERR(ssi->clk)) + return dev_err_probe(dev, PTR_ERR(ssi->clk), + "failed to get clock\n"); /* Enable the clock since regmap will not handle it in this case */ if (!ssi->has_ipg_clk_name) { ret = clk_prepare_enable(ssi->clk); - if (ret) { - dev_err(dev, "clk_prepare_enable failed: %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, + "clk_prepare_enable failed\n"); } /* Do not error out for consumer cases that live without a baud clock */ @@ -1552,10 +1549,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) &regconfig); else ssi->regs = devm_regmap_init_mmio(dev, iomem, &regconfig); - if (IS_ERR(ssi->regs)) { - dev_err(dev, "failed to init register map\n"); - return PTR_ERR(ssi->regs); - } + if (IS_ERR(ssi->regs)) + return dev_err_probe(dev, PTR_ERR(ssi->regs), + "failed to init register map\n"); ssi->irq = platform_get_irq(pdev, 0); if (ssi->irq < 0) @@ -1607,7 +1603,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) mutex_init(&ssi->ac97_reg_lock); ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); if (ret) { - dev_err(dev, "failed to set AC'97 ops\n"); + dev_err_probe(dev, ret, "failed to set AC'97 ops\n"); goto error_ac97_ops; } } @@ -1615,7 +1611,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_ssi_component, &ssi->cpu_dai_drv, 1); if (ret) { - dev_err(dev, "failed to register DAI: %d\n", ret); + dev_err_probe(dev, ret, "failed to register DAI\n"); goto error_asoc_register; } @@ -1623,7 +1619,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_request_irq(dev, ssi->irq, fsl_ssi_isr, 0, dev_name(dev), ssi); if (ret < 0) { - dev_err(dev, "failed to claim irq %u\n", ssi->irq); + dev_err_probe(dev, ret, "failed to claim irq %u\n", + ssi->irq); goto error_asoc_register; } } @@ -1649,8 +1646,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi->card_name, ssi->card_idx, NULL, 0); if (IS_ERR(ssi->card_pdev)) { ret = PTR_ERR(ssi->card_pdev); - dev_err(dev, "failed to register %s: %d\n", - ssi->card_name, ret); + dev_err_probe(dev, ret, "failed to register %s\n", + ssi->card_name); goto error_sound_card; } } -- 2.17.1

1 0

ALSA question
by Hai Nguyen 30 Aug '24

30 Aug '24

Hi, I have a C++ audio application that plays back audio data on the sound card, Digigram VX882e. https://www.digigram.com/support/vx882e-support/ The application runs on HPE ProLiant DL380 Gen9 Server. HPE ProLiant DL380 Gen9 Server It has worked well under RHEL 7 (kernel 3.10) and RHEL 8 (kernel 4.x). Asthe OS is upgraded to RHEL 9 (kernel 5.14), there is an buffer underrun. When investigating the issue, I run “speaker-test” utility. It fails with the error below. I'm just wonder if this has something to do with the changes in ALSA driver from kernel 4.x to kernel 5.14? If not, what could cause the buffer underrun issue? Your advice/help is highly appreciated, Thanks, Kevin Nguyen sudo speaker-test -D plughw:VX882e0 -c 2 speaker-test 1.2.9 Playback device is plughw:VX882e0 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 8192 Period size range from 64 to 4096 Using max buffer size 8192 Periods = 4 was set period_size = 2048 was set buffer_size = 8192 0 - Front Left Write error: -5,Input/outputerror xrun_recovery failed:-5,Input/output error Transfer failed: Input/outputerror The Digigram card is detected with “aplay -l” command. sudo aplay -l **** List of PLAYBACK Hardware Devices **** card 0: VX882e0 [Digigram [PCM #0]], device 0: pcxhr 0[pcxhr 0] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 1: VX882e1 [Digigram [PCM #1]], device 0: pcxhr 1[pcxhr 1] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 2: VX882e2 [Digigram [PCM #2]], device 0: pcxhr 2 [pcxhr2] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 3: VX882e3 [Digigram [PCM #3]], device 0: pcxhr 3[pcxhr 3] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3

2 1

[PATCH v3] ASoC: codecs: wsa884x: Implement temperature reading and hwmon
by Krzysztof Kozlowski 29 Aug '24

29 Aug '24

Read temperature of the speaker and expose it via hwmon interface, which will be later used during calibration of speaker protection algorithms. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v3: 1. Replace Kconfig HWMON dependency in favor of IS_REACHABLE(CONFIG_HWMON) which solves Kernel Test Robot warning for !HWMON and HWMON=y && WSA884x=m Changes in v2: 1. Add missing dependency on HWMON for y!=m builds (Kernel Test Robot report: undefined reference to `devm_hwmon_device_register_with_info'). --- sound/soc/codecs/wsa884x.c | 201 +++++++++++++++++++++++++++++++++++++ 1 file changed, 201 insertions(+) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 8db1380d1f10..86df5152c547 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -5,11 +5,14 @@ */ #include <linux/bitfield.h> +#include <linux/cleanup.h> #include <linux/device.h> #include <linux/gpio/consumer.h> +#include <linux/hwmon.h> #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/mutex.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -301,8 +304,28 @@ #define WSA884X_PA_FSM_MSK1 (WSA884X_DIG_CTRL0_BASE + 0x3b) #define WSA884X_PA_FSM_BYP_CTL (WSA884X_DIG_CTRL0_BASE + 0x3c) #define WSA884X_PA_FSM_BYP0 (WSA884X_DIG_CTRL0_BASE + 0x3d) +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK 0x01 +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_SHIFT 0 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK 0x02 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_SHIFT 1 +#define WSA884X_PA_FSM_BYP0_BG_EN_MASK 0x04 +#define WSA884X_PA_FSM_BYP0_BG_EN_SHIFT 2 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_MASK 0x08 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_SHIFT 3 +#define WSA884X_PA_FSM_BYP0_PA_EN_MASK 0x10 +#define WSA884X_PA_FSM_BYP0_PA_EN_SHIFT 4 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK 0x20 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_SHIFT 5 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK 0x40 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_SHIFT 6 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_MASK 0x80 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_SHIFT 7 #define WSA884X_PA_FSM_BYP1 (WSA884X_DIG_CTRL0_BASE + 0x3e) #define WSA884X_TADC_VALUE_CTL (WSA884X_DIG_CTRL0_BASE + 0x50) +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK 0x01 +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_SHIFT 0 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_MASK 0x02 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_SHIFT 1 #define WSA884X_TEMP_DETECT_CTL (WSA884X_DIG_CTRL0_BASE + 0x51) #define WSA884X_TEMP_DIN_MSB (WSA884X_DIG_CTRL0_BASE + 0x52) #define WSA884X_TEMP_DIN_LSB (WSA884X_DIG_CTRL0_BASE + 0x53) @@ -691,6 +714,17 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +/* Two-point trimming for temperature calibration */ +#define WSA884X_T1_TEMP -10L +#define WSA884X_T2_TEMP 150L + +/* + * Device will report senseless data in many cases, so discard any measurements + * outside of valid range. + */ +#define WSA884X_LOW_TEMP_THRESHOLD 5 +#define WSA884X_HIGH_TEMP_THRESHOLD 45 + struct wsa884x_priv { struct regmap *regmap; struct device *dev; @@ -706,6 +740,13 @@ struct wsa884x_priv { int active_ports; int dev_mode; bool hw_init; + /* + * Protects temperature reading code (related to speaker protection) and + * fields: temperature and pa_on. + */ + struct mutex sp_lock; + unsigned int temperature; + bool pa_on; }; enum { @@ -1660,6 +1701,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = true; + mutex_unlock(&wsa884x->sp_lock); + wsa884x_spkr_post_pmu(component, wsa884x); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, @@ -1671,6 +1716,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); + + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = false; + mutex_unlock(&wsa884x->sp_lock); break; } @@ -1810,6 +1859,144 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; +static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp) +{ + unsigned int d1_msb = 0, d1_lsb = 0, d2_msb = 0, d2_lsb = 0; + unsigned int dmeas_msb = 0, dmeas_lsb = 0; + int d1, d2, dmeas; + unsigned int mask; + long val; + int ret; + + guard(mutex)(&wsa884x->sp_lock); + + if (wsa884x->pa_on) { + /* + * Reading temperature is possible only when Power Amplifier is + * off. Report last cached data. + */ + *temp = wsa884x->temperature; + return 0; + } + + ret = pm_runtime_resume_and_get(wsa884x->dev); + if (ret < 0) + return ret; + + mask = WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK | + WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK | + WSA884X_PA_FSM_BYP0_BG_EN_MASK | + WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK | + WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK | + WSA884X_PA_FSM_BYP0_TSADC_EN_MASK; + /* + * Here and further do not care about read or update failures. + * For example, before turning on Power Amplifier for the first + * time, reading WSA884X_TEMP_DIN_MSB will always return 0. + * Instead, check if returned value is within reasonable + * thresholds. + */ + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, mask); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x0)); + + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_MSB, &dmeas_msb); + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_LSB, &dmeas_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x1)); + + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_1, &d1_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_2, &d1_lsb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_3, &d2_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_4, &d2_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, 0x0); + + dmeas = (((dmeas_msb & 0xff) << 0x8) | (dmeas_lsb & 0xff)) >> 0x6; + d1 = (((d1_msb & 0xff) << 0x8) | (d1_lsb & 0xff)) >> 0x6; + d2 = (((d2_msb & 0xff) << 0x8) | (d2_lsb & 0xff)) >> 0x6; + + if (d1 == d2) { + /* Incorrect data in OTP? */ + ret = -EINVAL; + goto out; + } + + val = WSA884X_T1_TEMP + (((dmeas - d1) * (WSA884X_T2_TEMP - WSA884X_T1_TEMP))/(d2 - d1)); + + dev_dbg(wsa884x->dev, "Measured temp %ld (dmeas=%d, d1=%d, d2=%d)\n", + val, dmeas, d1, d2); + + if ((val > WSA884X_LOW_TEMP_THRESHOLD) && + (val < WSA884X_HIGH_TEMP_THRESHOLD)) { + wsa884x->temperature = val; + *temp = val; + ret = 0; + } else { + ret = -EAGAIN; + } + +out: + pm_runtime_mark_last_busy(wsa884x->dev); + pm_runtime_put_autosuspend(wsa884x->dev); + + return ret; +} + +static umode_t wsa884x_hwmon_is_visible(const void *data, + enum hwmon_sensor_types type, u32 attr, + int channel) +{ + if (type != hwmon_temp) + return 0; + + switch (attr) { + case hwmon_temp_input: + return 0444; + default: + break; + } + + return 0; +} + +static int wsa884x_hwmon_read(struct device *dev, + enum hwmon_sensor_types type, + u32 attr, int channel, long *temp) +{ + int ret; + + switch (attr) { + case hwmon_temp_input: + ret = wsa884x_get_temp(dev_get_drvdata(dev), temp); + break; + default: + ret = -EOPNOTSUPP; + break; + } + + return ret; +} + +static const struct hwmon_channel_info *const wsa884x_hwmon_info[] = { + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + NULL +}; + +static const struct hwmon_ops wsa884x_hwmon_ops = { + .is_visible = wsa884x_hwmon_is_visible, + .read = wsa884x_hwmon_read, +}; + +static const struct hwmon_chip_info wsa884x_hwmon_chip_info = { + .ops = &wsa884x_hwmon_ops, + .info = wsa884x_hwmon_info, +}; + static void wsa884x_reset_powerdown(void *data) { struct wsa884x_priv *wsa884x = data; @@ -1866,6 +2053,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (!wsa884x) return -ENOMEM; + mutex_init(&wsa884x->sp_lock); + for (i = 0; i < WSA884X_SUPPLIES_NUM; i++) wsa884x->supplies[i].supply = wsa884x_supply_name[i]; @@ -1923,6 +2112,18 @@ static int wsa884x_probe(struct sdw_slave *pdev, regcache_cache_only(wsa884x->regmap, true); wsa884x->hw_init = true; + if (IS_REACHABLE(CONFIG_HWMON)) { + struct device *hwmon; + + hwmon = devm_hwmon_device_register_with_info(dev, "wsa884x", + wsa884x, + &wsa884x_hwmon_chip_info, + NULL); + if (IS_ERR(hwmon)) + return dev_err_probe(dev, PTR_ERR(hwmon), + "Failed to register hwmon sensor\n"); + } + pm_runtime_set_autosuspend_delay(dev, 3000); pm_runtime_use_autosuspend(dev); pm_runtime_mark_last_busy(dev); -- 2.43.0

2 1

Re: [PATCH] ALSA: hda/realtek: Autodetect Cirrus Logic companion amplifier bindings
by Takashi Iwai 29 Aug '24

29 Aug '24

On Thu, 29 Aug 2024 18:11:14 +0200, Simon Trimmer wrote: > > We do not need model specific HDA quirks to construct the component > binding details of Cirrus Logic companion amplifiers as this information > is already present in the ACPI. > > Quirks are then only required for special workarounds not described in > the ACPI such as internal configuration of the Realtek codecs. > > The codec pointer is now initialized in hda_component_manager_init() so > that we can detect when companion amplifiers are found in the ACPI but > the SSID invokes a quirk that also attempts to create the component > binding. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> > --- > v1 - This patch applies to for-next with this additional patch: > b1faff28a2bd ("ALSA: hda: hda_component: Fix mutex crash if nothing ever binds") > which is currently only in the for-linus. OK, I now merged for-linus to for-next, and applied yours. > We're looking to do the same for the CS35L41 devices, but that will > come later because there are many more quirks to work through. That's understandable. Thanks! Takashi

1 0

[PATCH] ASoC: mediatek: mt8188-mt6359: Modify key
by Zhang Yi 29 Aug '24

29 Aug '24

In order to get the correct keys when using the ES8326.We will associate SND_JACK_BTN_1 to KEY_VOLUMEUP and SND_JACK_BTN_2 to KEY_VOLUMEDOWN when the ES8326 flag is recognized. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index eba6f4c445ff..08ae962afeb9 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -734,6 +734,7 @@ static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HEADSET]; struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; int ret; ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_nau8825_widgets, @@ -762,10 +763,18 @@ static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + if (card_data->flags & ES8326_HS_PRESENT) { + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + } else { + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + } + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { -- 2.17.1

2 1

[PATCH v1] ASoC: codecs: Simplify with dev_err_probe()
by Yang Ruibin 29 Aug '24

29 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 4 ++-- sound/soc/codecs/adau1701.c | 12 ++++++------ sound/soc/codecs/ssm2602.c | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..25fd8bc4f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -239,8 +239,8 @@ static int ad1980_soc_probe(struct snd_soc_component *component) ac97 = snd_soc_new_ac97_component(component, 0, 0); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to register AC97 component: %d\n", ret); } regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..9fe18b22b 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -678,8 +678,8 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to enable regulators: %d\n", ret); } /* @@ -799,15 +799,15 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; + return dev_err_probe(dev, ret, + "Failed to get regulators: %d\n", ret); } ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; + return dev_err_probe(dev, ret, + "Failed to enable regulators: %d\n", ret); } adau1701->client = client; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..727bcccad 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -606,8 +606,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to issue reset: %d\n", ret); } regmap_register_patch(ssm2602->regmap, ssm2602_patch, -- 2.34.1

1 1

[PATCH -next] aoa: core: Using helper function for_each_child_of_node()
by Lin Ruifeng 29 Aug '24

29 Aug '24

Helper function for_each_child_of_node() can iterate through the DT node, so we don't need to use while loop. Signed-off-by: Lin Ruifeng <linruifeng4(a)huawei.com> --- sound/aoa/core/gpio-feature.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 39bb409b27f6..01060f7c3862 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -72,7 +72,7 @@ static struct device_node *get_gpio(char *name, gpio = of_find_node_by_name(NULL, "gpio"); if (!gpio) return NULL; - while ((np = of_get_next_child(gpio, np))) { + for_each_child_of_node(gpio, np) { audio_gpio = of_get_property(np, "audio-gpio", NULL); if (!audio_gpio) continue; -- 2.17.1

1 0