From 4ddd51ccff911a2e9e961307692532a325f6c78a Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Jul 2024 16:54:53 +0800 Subject: [PATCH 01/39] ASoC: fsl_micfil: Expand the range of FIFO watermark mask On the i.MX9x platforms, the mask of FIFO watermark is 0x1F, on i.MX8x platforms, the mask of FIFO watermark is 0X7. So use the mask 0x1F for all platforms to make them compatible. Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/1721897694-6088-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 0d37edb70261..96a6b88d0d67 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -831,7 +831,7 @@ static const struct reg_default fsl_micfil_reg_defaults[] = { {REG_MICFIL_CTRL1, 0x00000000}, {REG_MICFIL_CTRL2, 0x00000000}, {REG_MICFIL_STAT, 0x00000000}, - {REG_MICFIL_FIFO_CTRL, 0x00000007}, + {REG_MICFIL_FIFO_CTRL, 0x0000001F}, {REG_MICFIL_FIFO_STAT, 0x00000000}, {REG_MICFIL_DATACH0, 0x00000000}, {REG_MICFIL_DATACH1, 0x00000000}, diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index c6b902ba0a53..b7798a7cbf2a 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -72,7 +72,7 @@ #define MICFIL_STAT_CHXF(ch) BIT(ch) /* MICFIL FIFO Control Register -- REG_MICFIL_FIFO_CTRL 0x10 */ -#define MICFIL_FIFO_CTRL_FIFOWMK GENMASK(2, 0) +#define MICFIL_FIFO_CTRL_FIFOWMK GENMASK(4, 0) /* MICFIL FIFO Status Register -- REG_MICFIL_FIFO_STAT 0x14 */ #define MICFIL_FIFO_STAT_FIFOX_OVER(ch) BIT(ch) From aa4f76ef09a993efa9b5fab6ddf5d6d324baaea3 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Jul 2024 16:54:54 +0800 Subject: [PATCH 02/39] ASoC: fsl_micfil: Differentiate register access permission for platforms On i.MX9x platforms, the REG_MICFIL_FSYNC_CTRL, REG_MICFIL_VERID, REG_MICFIL_PARAM are added, but they are not existed on i.MX8x platforms. Use the existed micfil->soc->use_verid to distinguish the access permission for these platforms. Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/1721897694-6088-3-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 96a6b88d0d67..22b240a70ad4 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -855,6 +855,8 @@ static const struct reg_default fsl_micfil_reg_defaults[] = { static bool fsl_micfil_readable_reg(struct device *dev, unsigned int reg) { + struct fsl_micfil *micfil = dev_get_drvdata(dev); + switch (reg) { case REG_MICFIL_CTRL1: case REG_MICFIL_CTRL2: @@ -872,9 +874,6 @@ static bool fsl_micfil_readable_reg(struct device *dev, unsigned int reg) case REG_MICFIL_DC_CTRL: case REG_MICFIL_OUT_CTRL: case REG_MICFIL_OUT_STAT: - case REG_MICFIL_FSYNC_CTRL: - case REG_MICFIL_VERID: - case REG_MICFIL_PARAM: case REG_MICFIL_VAD0_CTRL1: case REG_MICFIL_VAD0_CTRL2: case REG_MICFIL_VAD0_STAT: @@ -883,6 +882,12 @@ static bool fsl_micfil_readable_reg(struct device *dev, unsigned int reg) case REG_MICFIL_VAD0_NDATA: case REG_MICFIL_VAD0_ZCD: return true; + case REG_MICFIL_FSYNC_CTRL: + case REG_MICFIL_VERID: + case REG_MICFIL_PARAM: + if (micfil->soc->use_verid) + return true; + fallthrough; default: return false; } @@ -890,6 +895,8 @@ static bool fsl_micfil_readable_reg(struct device *dev, unsigned int reg) static bool fsl_micfil_writeable_reg(struct device *dev, unsigned int reg) { + struct fsl_micfil *micfil = dev_get_drvdata(dev); + switch (reg) { case REG_MICFIL_CTRL1: case REG_MICFIL_CTRL2: @@ -899,7 +906,6 @@ static bool fsl_micfil_writeable_reg(struct device *dev, unsigned int reg) case REG_MICFIL_DC_CTRL: case REG_MICFIL_OUT_CTRL: case REG_MICFIL_OUT_STAT: /* Write 1 to Clear */ - case REG_MICFIL_FSYNC_CTRL: case REG_MICFIL_VAD0_CTRL1: case REG_MICFIL_VAD0_CTRL2: case REG_MICFIL_VAD0_STAT: /* Write 1 to Clear */ @@ -907,6 +913,10 @@ static bool fsl_micfil_writeable_reg(struct device *dev, unsigned int reg) case REG_MICFIL_VAD0_NCONFIG: case REG_MICFIL_VAD0_ZCD: return true; + case REG_MICFIL_FSYNC_CTRL: + if (micfil->soc->use_verid) + return true; + fallthrough; default: return false; } From aebb1813c279ce8f3a2dfa3f86def0c0ec1cbb8d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:41 +0200 Subject: [PATCH 03/39] ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask Device has up to WCD937X_MAX_TX_SWR_PORTS (or WCD937X_MAX_SWR_PORTS for sink) number of ports and the array assigned to prop.src_dpn_prop and prop.sink_dpn_prop has 0..WCD937X_MAX_TX_SWR_PORTS-1 elements. On the other hand, GENMASK(high, low) creates an inclusive mask between , so we need the mask from 0 up to WCD937X_MAX_TX_SWR_PORTS-1. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Fixes: c99a515ff153 ("ASoC: codecs: wcd937x-sdw: add SoundWire driver") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240726-asoc-wcd-wsa-swr-ports-genmask-v1-1-d4d7a8b56f05@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd937x-sdw.c b/sound/soc/codecs/wcd937x-sdw.c index 3abc8041406a..0c33f7f3dc25 100644 --- a/sound/soc/codecs/wcd937x-sdw.c +++ b/sound/soc/codecs/wcd937x-sdw.c @@ -1049,7 +1049,7 @@ static int wcd9370_probe(struct sdw_slave *pdev, pdev->prop.lane_control_support = true; pdev->prop.simple_clk_stop_capable = true; if (wcd->is_tx) { - pdev->prop.source_ports = GENMASK(WCD937X_MAX_TX_SWR_PORTS, 0); + pdev->prop.source_ports = GENMASK(WCD937X_MAX_TX_SWR_PORTS - 1, 0); pdev->prop.src_dpn_prop = wcd937x_dpn_prop; wcd->ch_info = &wcd937x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; @@ -1062,7 +1062,7 @@ static int wcd9370_probe(struct sdw_slave *pdev, /* Start in cache-only until device is enumerated */ regcache_cache_only(wcd->regmap, true); } else { - pdev->prop.sink_ports = GENMASK(WCD937X_MAX_SWR_PORTS, 0); + pdev->prop.sink_ports = GENMASK(WCD937X_MAX_SWR_PORTS - 1, 0); pdev->prop.sink_dpn_prop = wcd937x_dpn_prop; wcd->ch_info = &wcd937x_sdw_rx_ch_info[0]; } From 3f6fb03dae9c7dfba7670858d29e03c8faaa89fe Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:42 +0200 Subject: [PATCH 04/39] ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask Device has up to WCD938X_MAX_SWR_PORTS number of ports and the array assigned to prop.src_dpn_prop and prop.sink_dpn_prop has 0..WCD938X_MAX_SWR_PORTS-1 elements. On the other hand, GENMASK(high, low) creates an inclusive mask between , so we need the mask from 0 up to WCD938X_MAX_SWR_PORTS-1. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240726-asoc-wcd-wsa-swr-ports-genmask-v1-2-d4d7a8b56f05@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index c995bcc59ead..7da8a10bd0a9 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1252,12 +1252,12 @@ static int wcd9380_probe(struct sdw_slave *pdev, pdev->prop.lane_control_support = true; pdev->prop.simple_clk_stop_capable = true; if (wcd->is_tx) { - pdev->prop.source_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); + pdev->prop.source_ports = GENMASK(WCD938X_MAX_SWR_PORTS - 1, 0); pdev->prop.src_dpn_prop = wcd938x_dpn_prop; wcd->ch_info = &wcd938x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; } else { - pdev->prop.sink_ports = GENMASK(WCD938X_MAX_SWR_PORTS, 0); + pdev->prop.sink_ports = GENMASK(WCD938X_MAX_SWR_PORTS - 1, 0); pdev->prop.sink_dpn_prop = wcd938x_dpn_prop; wcd->ch_info = &wcd938x_sdw_rx_ch_info[0]; } From 74a79977c4e1d09eced33e6e22f875a5bb3fad29 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:43 +0200 Subject: [PATCH 05/39] ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask Device has up to WCD939X_MAX_TX_SWR_PORTS (or WCD939X_MAX_RX_SWR_PORTS for sink) number of ports and the array assigned to prop.src_dpn_prop and prop.sink_dpn_prop has 0..WCD939X_MAX_TX_SWR_PORTS-1 elements. On the other hand, GENMASK(high, low) creates an inclusive mask between , so we need the mask from 0 up to WCD939X_MAX_TX_SWR_PORTS-1. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Fixes: be2af391cea0 ("ASoC: codecs: Add WCD939x Soundwire devices driver") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240726-asoc-wcd-wsa-swr-ports-genmask-v1-3-d4d7a8b56f05@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd939x-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd939x-sdw.c b/sound/soc/codecs/wcd939x-sdw.c index 94b1e99a3ca0..fca95777a75a 100644 --- a/sound/soc/codecs/wcd939x-sdw.c +++ b/sound/soc/codecs/wcd939x-sdw.c @@ -1453,12 +1453,12 @@ static int wcd9390_probe(struct sdw_slave *pdev, const struct sdw_device_id *id) pdev->prop.lane_control_support = true; pdev->prop.simple_clk_stop_capable = true; if (wcd->is_tx) { - pdev->prop.source_ports = GENMASK(WCD939X_MAX_TX_SWR_PORTS, 0); + pdev->prop.source_ports = GENMASK(WCD939X_MAX_TX_SWR_PORTS - 1, 0); pdev->prop.src_dpn_prop = wcd939x_tx_dpn_prop; wcd->ch_info = &wcd939x_sdw_tx_ch_info[0]; pdev->prop.wake_capable = true; } else { - pdev->prop.sink_ports = GENMASK(WCD939X_MAX_RX_SWR_PORTS, 0); + pdev->prop.sink_ports = GENMASK(WCD939X_MAX_RX_SWR_PORTS - 1, 0); pdev->prop.sink_dpn_prop = wcd939x_rx_dpn_prop; wcd->ch_info = &wcd939x_sdw_rx_ch_info[0]; } From eb11c3bb64ad0a05aeacdb01039863aa2aa3614b Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:44 +0200 Subject: [PATCH 06/39] ASoC: codecs: wsa881x: Correct Soundwire ports mask Device has up to WSA881X_MAX_SWR_PORTS number of ports and the array assigned to prop.sink_dpn_prop has 0..WSA881X_MAX_SWR_PORTS-1 elements. On the other hand, GENMASK(high, low) creates an inclusive mask between , so we need the mask from 0 up to WSA881X_MAX_SWR_PORTS-1. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240726-asoc-wcd-wsa-swr-ports-genmask-v1-4-d4d7a8b56f05@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0478599d0f35..fb9e92f08d98 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1152,7 +1152,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, wsa881x->sconfig.frame_rate = 48000; wsa881x->sconfig.direction = SDW_DATA_DIR_RX; wsa881x->sconfig.type = SDW_STREAM_PDM; - pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); + pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS - 1, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; pdev->prop.clk_stop_mode1 = true; From 6801ac36f25690e14955f7f9eace1eaa29edbdd0 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:45 +0200 Subject: [PATCH 07/39] ASoC: codecs: wsa883x: Correct Soundwire ports mask Device has up to WSA883X_MAX_SWR_PORTS number of ports and the array assigned to prop.sink_dpn_prop has 0..WSA883X_MAX_SWR_PORTS-1 elements. On the other hand, GENMASK(high, low) creates an inclusive mask between , so we need the mask from 0 up to WSA883X_MAX_SWR_PORTS-1. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Fixes: 43b8c7dc85a1 ("ASoC: codecs: add wsa883x amplifier support") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240726-asoc-wcd-wsa-swr-ports-genmask-v1-5-d4d7a8b56f05@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index d0ab4e2290b6..3e4fdaa3f44f 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1406,7 +1406,7 @@ static int wsa883x_probe(struct sdw_slave *pdev, WSA883X_MAX_SWR_PORTS)) dev_dbg(dev, "Static Port mapping not specified\n"); - pdev->prop.sink_ports = GENMASK(WSA883X_MAX_SWR_PORTS, 0); + pdev->prop.sink_ports = GENMASK(WSA883X_MAX_SWR_PORTS - 1, 0); pdev->prop.simple_clk_stop_capable = true; pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; From dcb6631d05152930e2ea70fd2abfd811b0e970b5 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:46 +0200 Subject: [PATCH 08/39] ASoC: codecs: wsa884x: Correct Soundwire ports mask Device has up to WSA884X_MAX_SWR_PORTS number of ports and the array assigned to prop.sink_dpn_prop has 0..WSA884X_MAX_SWR_PORTS-1 elements. On the other hand, GENMASK(high, low) creates an inclusive mask between , so we need the mask from 0 up to WSA884X_MAX_SWR_PORTS-1. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Fixes: aa21a7d4f68a ("ASoC: codecs: wsa884x: Add WSA884x family of speakers") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240726-asoc-wcd-wsa-swr-ports-genmask-v1-6-d4d7a8b56f05@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa884x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index d17ae17b2938..89eb5e03a617 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -1895,7 +1895,7 @@ static int wsa884x_probe(struct sdw_slave *pdev, WSA884X_MAX_SWR_PORTS)) dev_dbg(dev, "Static Port mapping not specified\n"); - pdev->prop.sink_ports = GENMASK(WSA884X_MAX_SWR_PORTS, 0); + pdev->prop.sink_ports = GENMASK(WSA884X_MAX_SWR_PORTS - 1, 0); pdev->prop.simple_clk_stop_capable = true; pdev->prop.sink_dpn_prop = wsa884x_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; From 6b99068d5ea0aa295f15f30afc98db74d056ec7b Mon Sep 17 00:00:00 2001 From: Jerome Audu Date: Sat, 27 Jul 2024 15:40:15 +0200 Subject: [PATCH 09/39] ASoC: sti: add missing probe entry for player and reader This patch addresses a regression in the ASoC STI drivers that was introduced in Linux version 6.6.y. The issue originated from a series of patches (see https://lore.kernel.org/all/87wmy5b0wt.wl-kuninori.morimoto.gx@renesas.com/) that unintentionally omitted necessary probe functions for the player and reader components. Probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. This patch correctly reinserts the missing probe entries, restoring the intended functionality. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu Link: https://patch.msgid.link/20240727-sti-audio-fix-v2-1-208bde546c3f@free.fr Signed-off-by: Mark Brown --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, From c118478665f467e57d06b2354de65974b246b82b Mon Sep 17 00:00:00 2001 From: Bruno Ancona Date: Sun, 28 Jul 2024 22:50:32 -0600 Subject: [PATCH 10/39] ASoC: amd: yc: Support mic on HP 14-em0002la Add support for the internal microphone for HP 14-em0002la laptop using a quirk entry. Signed-off-by: Bruno Ancona Link: https://patch.msgid.link/20240729045032.223230-1-brunoanconasala@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 1769e07e83dc..f4bbfffe9fcb 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -423,6 +423,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8A3E"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8B27"), + } + }, { .driver_data = &acp6x_card, .matches = { From 45d763fe503e6e0f180f873b750aea307e73fdcf Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Fri, 26 Jul 2024 10:11:11 -0500 Subject: [PATCH 11/39] ASoC: cs530x: Change IN HPF Select kcontrol name Change to the IN HPF Select kcontrol to the correct name IN DEC Filter Select. Signed-off-by: Paul Handrigan Link: https://patch.msgid.link/20240726151111.3247774-1-paulha@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs530x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs530x.c b/sound/soc/codecs/cs530x.c index 25a86a32e936..da52afe56c3c 100644 --- a/sound/soc/codecs/cs530x.c +++ b/sound/soc/codecs/cs530x.c @@ -129,16 +129,16 @@ volsw_err: static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1270, 50, 0); -static const char * const cs530x_in_hpf_text[] = { +static const char * const cs530x_in_filter_text[] = { "Min Phase Slow Roll-off", "Min Phase Fast Roll-off", "Linear Phase Slow Roll-off", "Linear Phase Fast Roll-off", }; -static SOC_ENUM_SINGLE_DECL(cs530x_in_hpf_enum, CS530X_IN_FILTER, +static SOC_ENUM_SINGLE_DECL(cs530x_in_filter_enum, CS530X_IN_FILTER, CS530X_IN_FILTER_SHIFT, - cs530x_in_hpf_text); + cs530x_in_filter_text); static const char * const cs530x_in_4ch_sum_text[] = { "None", @@ -189,7 +189,7 @@ SOC_SINGLE_EXT_TLV("IN1 Volume", CS530X_IN_VOL_CTRL1_0, 0, 255, 1, SOC_SINGLE_EXT_TLV("IN2 Volume", CS530X_IN_VOL_CTRL1_1, 0, 255, 1, snd_soc_get_volsw, cs530x_put_volsw_vu, in_vol_tlv), -SOC_ENUM("IN HPF Select", cs530x_in_hpf_enum), +SOC_ENUM("IN DEC Filter Select", cs530x_in_filter_enum), SOC_ENUM("Input Ramp Up", cs530x_ramp_inc_enum), SOC_ENUM("Input Ramp Down", cs530x_ramp_dec_enum), From 9da8aa3b3ca05b22be5ba312771e6df4366e56cc Mon Sep 17 00:00:00 2001 From: Francesco Dolcini Date: Wed, 31 Jul 2024 13:48:28 +0200 Subject: [PATCH 12/39] ASoC: nau8822: Lower debug print priority NAU8822 codec PLL parameters are not an information that the general user should care about, this print is supposed to be used for debugging, adjust the debug print priority accordingly. Signed-off-by: Francesco Dolcini Link: https://patch.msgid.link/20240731114828.61238-1-francesco@dolcini.it Signed-off-by: Mark Brown --- sound/soc/codecs/nau8822.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index e1cbaf8a944d..fd4a96a12060 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -736,7 +736,7 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, return ret; } - dev_info(component->dev, + dev_dbg(component->dev, "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); From 7354eb7f1558466e92e926802d36e69e42938ea9 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Wed, 31 Jul 2024 14:21:44 -0700 Subject: [PATCH 13/39] ASoC: SOF: Remove libraries from topology lookups Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey Cc: Wojciech Macek Reviewed-by: AngeloGioacchino Del Regno Link: https://patch.msgid.link/20240731212153.921327-1-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 24ae1d4959be..1c6e035fd313 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static const struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" From becfa08bfefa2cbb22c84d9e583e81387f2f3bf2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:31 +0100 Subject: [PATCH 14/39] ASoC: cs42l43: Remove redundant semi-colon at end of function Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20240802105734.2309788-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 92674314227c..80825777048a 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -608,7 +608,7 @@ static int cs42l43_sdw_hw_params(struct snd_pcm_substream *substream, return ret; return cs42l43_set_sample_rate(substream, params, dai); -}; +} static const struct snd_soc_dai_ops cs42l43_sdw_ops = { .startup = cs42l43_startup, From c8a132e2e032b00828d51141ab34f9aeb24f44ae Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:32 +0100 Subject: [PATCH 15/39] ASoC: soc-component: Add new snd_soc_component_get_kcontrol() helpers Add new helper functions snd_soc_component_get_kcontrol() and snd_soc_component_get_kcontrol_locked() that returns a kcontrol by name, but will factor in the components name_prefix, to handle situations where multiple components are present with the same controls. Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20240802105734.2309788-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 5 +++++ sound/soc/soc-component.c | 42 ++++++++++++++++++++++++++++------- 2 files changed, 39 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index ceca69b46a82..bf2e381cd124 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -462,6 +462,11 @@ int snd_soc_component_force_enable_pin_unlocked( const char *pin); /* component controls */ +struct snd_kcontrol *snd_soc_component_get_kcontrol(struct snd_soc_component *component, + const char * const ctl); +struct snd_kcontrol * +snd_soc_component_get_kcontrol_locked(struct snd_soc_component *component, + const char * const ctl); int snd_soc_component_notify_control(struct snd_soc_component *component, const char * const ctl); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 4d7c2e3c929a..42f481321919 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -236,19 +236,45 @@ int snd_soc_component_force_enable_pin_unlocked( } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked); +static void soc_get_kcontrol_name(struct snd_soc_component *component, + char *buf, int size, const char * const ctl) +{ + /* When updating, change also snd_soc_dapm_widget_name_cmp() */ + if (component->name_prefix) + snprintf(buf, size, "%s %s", component->name_prefix, ctl); + else + snprintf(buf, size, "%s", ctl); +} + +struct snd_kcontrol *snd_soc_component_get_kcontrol(struct snd_soc_component *component, + const char * const ctl) +{ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + soc_get_kcontrol_name(component, name, ARRAY_SIZE(name), ctl); + + return snd_soc_card_get_kcontrol(component->card, name); +} +EXPORT_SYMBOL_GPL(snd_soc_component_get_kcontrol); + +struct snd_kcontrol * +snd_soc_component_get_kcontrol_locked(struct snd_soc_component *component, + const char * const ctl) +{ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + soc_get_kcontrol_name(component, name, ARRAY_SIZE(name), ctl); + + return snd_soc_card_get_kcontrol_locked(component->card, name); +} +EXPORT_SYMBOL_GPL(snd_soc_component_get_kcontrol_locked); + int snd_soc_component_notify_control(struct snd_soc_component *component, const char * const ctl) { - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; struct snd_kcontrol *kctl; - /* When updating, change also snd_soc_dapm_widget_name_cmp() */ - if (component->name_prefix) - snprintf(name, ARRAY_SIZE(name), "%s %s", component->name_prefix, ctl); - else - snprintf(name, ARRAY_SIZE(name), "%s", ctl); - - kctl = snd_soc_card_get_kcontrol(component->card, name); + kctl = snd_soc_component_get_kcontrol(component, ctl); if (!kctl) return soc_component_ret(component, -EINVAL); From 4791c422981350d0de4ad02a14a08b99c766d06f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:33 +0100 Subject: [PATCH 16/39] ASoC: cs35l45: Use new snd_soc_component_get_kcontrol_locked() helper No longer any need to hard code the addition of the name prefix, use the new helper function. Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20240802105734.2309788-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 2392c6effed8..1e9d73bee3b4 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -176,17 +176,10 @@ static int cs35l45_activate_ctl(struct snd_soc_component *component, struct snd_kcontrol *kcontrol; struct snd_kcontrol_volatile *vd; unsigned int index_offset; - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - if (component->name_prefix) - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s", - component->name_prefix, ctl_name); - else - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s", ctl_name); - - kcontrol = snd_soc_card_get_kcontrol_locked(component->card, name); + kcontrol = snd_soc_component_get_kcontrol_locked(component, ctl_name); if (!kcontrol) { - dev_err(component->dev, "Can't find kcontrol %s\n", name); + dev_err(component->dev, "Can't find kcontrol %s\n", ctl_name); return -EINVAL; } From 93afd028fb5f06a46a32375fd1f0473451eb1c5a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:34 +0100 Subject: [PATCH 17/39] ASoC: cs42l43: Cache shutter IRQ control pointers The microphone/speaker privacy shutter ALSA control handlers need to call pm_runtime_resume, since the hardware needs to be powered up to check the hardware state of the shutter. The IRQ handler for the shutters also needs to notify the ALSA control to inform user-space the shutters updated. However this leads to a mutex inversion, between the sdw_dev_lock and the controls_rwsem. To avoid this mutex inversion cache the kctl pointers before the IRQ handler, which avoids the need to lookup the control and take the controls_rwsem. Suggested-by: Jaroslav Kysela Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20240802105734.2309788-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 73 +++++++++++++++++++++++++++++--------- sound/soc/codecs/cs42l43.h | 2 ++ 2 files changed, 58 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 80825777048a..5183b4586424 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -7,6 +7,7 @@ #include #include +#include #include #include #include @@ -252,24 +253,20 @@ CS42L43_IRQ_COMPLETE(load_detect) static irqreturn_t cs42l43_mic_shutter(int irq, void *data) { struct cs42l43_codec *priv = data; - static const char * const controls[] = { - "Decimator 1 Switch", - "Decimator 2 Switch", - "Decimator 3 Switch", - "Decimator 4 Switch", - }; - int i, ret; + struct snd_soc_component *component = priv->component; + int i; dev_dbg(priv->dev, "Microphone shutter changed\n"); - if (!priv->component) + if (!component) return IRQ_NONE; - for (i = 0; i < ARRAY_SIZE(controls); i++) { - ret = snd_soc_component_notify_control(priv->component, - controls[i]); - if (ret) + for (i = 1; i < ARRAY_SIZE(priv->kctl); i++) { + if (!priv->kctl[i]) return IRQ_NONE; + + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, &priv->kctl[i]->id); } return IRQ_HANDLED; @@ -278,18 +275,19 @@ static irqreturn_t cs42l43_mic_shutter(int irq, void *data) static irqreturn_t cs42l43_spk_shutter(int irq, void *data) { struct cs42l43_codec *priv = data; - int ret; + struct snd_soc_component *component = priv->component; dev_dbg(priv->dev, "Speaker shutter changed\n"); - if (!priv->component) + if (!component) return IRQ_NONE; - ret = snd_soc_component_notify_control(priv->component, - "Speaker Digital Switch"); - if (ret) + if (!priv->kctl[0]) return IRQ_NONE; + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, &priv->kctl[0]->id); + return IRQ_HANDLED; } @@ -590,7 +588,46 @@ static int cs42l43_asp_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mas return 0; } +static int cs42l43_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); + static const char * const controls[] = { + "Speaker Digital Switch", + "Decimator 1 Switch", + "Decimator 2 Switch", + "Decimator 3 Switch", + "Decimator 4 Switch", + }; + int i; + + static_assert(ARRAY_SIZE(controls) == ARRAY_SIZE(priv->kctl)); + + for (i = 0; i < ARRAY_SIZE(controls); i++) { + if (priv->kctl[i]) + continue; + + priv->kctl[i] = snd_soc_component_get_kcontrol(component, controls[i]); + } + + return 0; +} + +static int cs42l43_dai_remove(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); + int i; + + for (i = 0; i < ARRAY_SIZE(priv->kctl); i++) + priv->kctl[i] = NULL; + + return 0; +} + static const struct snd_soc_dai_ops cs42l43_asp_ops = { + .probe = cs42l43_dai_probe, + .remove = cs42l43_dai_remove, .startup = cs42l43_startup, .hw_params = cs42l43_asp_hw_params, .set_fmt = cs42l43_asp_set_fmt, @@ -611,6 +648,8 @@ static int cs42l43_sdw_hw_params(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops cs42l43_sdw_ops = { + .probe = cs42l43_dai_probe, + .remove = cs42l43_dai_remove, .startup = cs42l43_startup, .set_stream = cs42l43_sdw_set_stream, .hw_params = cs42l43_sdw_hw_params, diff --git a/sound/soc/codecs/cs42l43.h b/sound/soc/codecs/cs42l43.h index 9924c13e1eb5..9c144e129535 100644 --- a/sound/soc/codecs/cs42l43.h +++ b/sound/soc/codecs/cs42l43.h @@ -100,6 +100,8 @@ struct cs42l43_codec { struct delayed_work hp_ilimit_clear_work; bool hp_ilimited; int hp_ilimit_count; + + struct snd_kcontrol *kctl[5]; }; #if IS_REACHABLE(CONFIG_SND_SOC_CS42L43_SDW) From 34e1b1bb73244219b3b3e24911e56c6e7b2b679e Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Thu, 1 Aug 2024 14:31:39 +0000 Subject: [PATCH 18/39] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. There is no reason that the UCM or user should need to interact with any of the ALSA controls for the firmware coefficients so they can be removed entirely, this also simplifies the driver. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240801143139.34549-1-simont@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda.c | 38 +------------------------------------ sound/pci/hda/cs35l56_hda.h | 1 - 2 files changed, 1 insertion(+), 38 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..31cc92bac89a 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -559,18 +559,6 @@ static void cs35l56_hda_release_firmware_files(const struct firmware *wmfw_firmw kfree(coeff_filename); } -static void cs35l56_hda_create_dsp_controls_work(struct work_struct *work) -{ - struct cs35l56_hda *cs35l56 = container_of(work, struct cs35l56_hda, control_work); - struct hda_cs_dsp_ctl_info info; - - info.device_name = cs35l56->amp_name; - info.fw_type = HDA_CS_DSP_FW_MISC; - info.card = cs35l56->codec->card; - - hda_cs_dsp_add_controls(&cs35l56->cs_dsp, &info); -} - static void cs35l56_hda_apply_calibration(struct cs35l56_hda *cs35l56) { int ret; @@ -595,26 +583,15 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) char *wmfw_filename = NULL; unsigned int preloaded_fw_ver; bool firmware_missing; - bool add_dsp_controls_required = false; int ret; - /* - * control_work must be flushed before proceeding, but we can't do that - * here as it would create a deadlock on controls_rwsem so it must be - * performed before queuing dsp_work. - */ - WARN_ON_ONCE(work_busy(&cs35l56->control_work)); - /* * Prepare for a new DSP power-up. If the DSP has had firmware * downloaded previously then it needs to be powered down so that it - * can be updated and if hadn't been patched before then the controls - * will need to be added once firmware download succeeds. + * can be updated. */ if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - else - add_dsp_controls_required = true; cs35l56->base.fw_patched = false; @@ -698,15 +675,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) CS35L56_FIRMWARE_MISSING); cs35l56->base.fw_patched = true; - /* - * Adding controls is deferred to prevent a lock inversion - ALSA takes - * the controls_rwsem when adding a control, the get() / put() - * functions of a control are called holding controls_rwsem and those - * that depend on running firmware wait for dsp_work() to complete. - */ - if (add_dsp_controls_required) - queue_work(system_long_wq, &cs35l56->control_work); - ret = cs_dsp_run(&cs35l56->cs_dsp); if (ret) dev_dbg(cs35l56->base.dev, "%s: cs_dsp_run ret %d\n", __func__, ret); @@ -753,7 +721,6 @@ static int cs35l56_hda_bind(struct device *dev, struct device *master, void *mas strscpy(comp->name, dev_name(dev), sizeof(comp->name)); comp->playback_hook = cs35l56_hda_playback_hook; - flush_work(&cs35l56->control_work); queue_work(system_long_wq, &cs35l56->dsp_work); cs35l56_hda_create_controls(cs35l56); @@ -775,7 +742,6 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void * struct hda_component *comp; cancel_work_sync(&cs35l56->dsp_work); - cancel_work_sync(&cs35l56->control_work); cs35l56_hda_remove_controls(cs35l56); @@ -806,7 +772,6 @@ static int cs35l56_hda_system_suspend(struct device *dev) struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); cs35l56_hda_wait_dsp_ready(cs35l56); - flush_work(&cs35l56->control_work); if (cs35l56->playing) cs35l56_hda_pause(cs35l56); @@ -1026,7 +991,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) dev_set_drvdata(cs35l56->base.dev, cs35l56); INIT_WORK(&cs35l56->dsp_work, cs35l56_hda_dsp_work); - INIT_WORK(&cs35l56->control_work, cs35l56_hda_create_dsp_controls_work); ret = cs35l56_hda_read_acpi(cs35l56, hid, id); if (ret) diff --git a/sound/pci/hda/cs35l56_hda.h b/sound/pci/hda/cs35l56_hda.h index c40d159507c2..38d94fb213a5 100644 --- a/sound/pci/hda/cs35l56_hda.h +++ b/sound/pci/hda/cs35l56_hda.h @@ -23,7 +23,6 @@ struct cs35l56_hda { struct cs35l56_base base; struct hda_codec *codec; struct work_struct dsp_work; - struct work_struct control_work; int index; const char *system_name; From 312c04cee408a8448ec8b639fe7f0434017d7161 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 1 Aug 2024 16:50:44 +0100 Subject: [PATCH 19/39] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients When the CS35L41 loads its firmware, it has a number of controls to affect its behaviour. Currently, these controls are exposed as ALSA Controls. These controls were never intended to be exposed to users but the firmware doesn't mark them hidden, so make the driver ignore them. Any changes in the coefficients handled by these controls needs to be matched to the individual system by SSID, which is already handled using the tuning file, when firmware is loaded, so UCM should not be setting these controls anyway. Signed-off-by: Stefan Binding Link: https://patch.msgid.link/20240801155047.456540-1-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 4b411ed8c3fe..3a92e98da72d 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -133,17 +133,6 @@ static const struct reg_sequence cs35l41_hda_mute[] = { { CS35L41_AMP_DIG_VOL_CTRL, 0x0000A678 }, // AMP_HPF_PCM_EN = 1, AMP_VOL_PCM Mute }; -static void cs35l41_add_controls(struct cs35l41_hda *cs35l41) -{ - struct hda_cs_dsp_ctl_info info; - - info.device_name = cs35l41->amp_name; - info.fw_type = cs35l41->firmware_type; - info.card = cs35l41->codec->card; - - hda_cs_dsp_add_controls(&cs35l41->cs_dsp, &info); -} - static const struct cs_dsp_client_ops client_ops = { .control_remove = hda_cs_dsp_control_remove, }; @@ -603,8 +592,6 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) if (ret) goto err; - cs35l41_add_controls(cs35l41); - cs35l41_hda_apply_calibration(cs35l41); err: From 45b4acab4cac79503663f0a4be9eb3752db04d4b Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 5 Aug 2024 10:27:20 +0000 Subject: [PATCH 20/39] ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() The callback allows codec drivers to affect how firmware coefficients are added as controls. For example a codec driver may selectively add controls by choosing to call wm_adsp_control_add() based on some filter logic. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240805102721.30102-2-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 17 ++++++++++++++--- sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9f8549b34e30..e69283195f36 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -583,7 +583,7 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(kcontrol); } -static int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) +int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) { struct wm_adsp *dsp = container_of(cs_ctl->dsp, struct wm_adsp, cs_dsp); struct cs_dsp *cs_dsp = &dsp->cs_dsp; @@ -658,6 +658,17 @@ err_ctl: return ret; } +EXPORT_SYMBOL_GPL(wm_adsp_control_add); + +static int wm_adsp_control_add_cb(struct cs_dsp_coeff_ctl *cs_ctl) +{ + struct wm_adsp *dsp = container_of(cs_ctl->dsp, struct wm_adsp, cs_dsp); + + if (dsp->control_add) + return (dsp->control_add)(dsp, cs_ctl); + else + return wm_adsp_control_add(cs_ctl); +} static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) { @@ -2072,12 +2083,12 @@ irqreturn_t wm_halo_wdt_expire(int irq, void *data) EXPORT_SYMBOL_GPL(wm_halo_wdt_expire); static const struct cs_dsp_client_ops wm_adsp1_client_ops = { - .control_add = wm_adsp_control_add, + .control_add = wm_adsp_control_add_cb, .control_remove = wm_adsp_control_remove, }; static const struct cs_dsp_client_ops wm_adsp2_client_ops = { - .control_add = wm_adsp_control_add, + .control_add = wm_adsp_control_add_cb, .control_remove = wm_adsp_control_remove, .pre_run = wm_adsp_pre_run, .post_run = wm_adsp_event_post_run, diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index e53dfcf1f78f..edc5b02ae765 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -37,6 +37,7 @@ struct wm_adsp { bool wmfw_optional; struct work_struct boot_work; + int (*control_add)(struct wm_adsp *dsp, struct cs_dsp_coeff_ctl *cs_ctl); int (*pre_run)(struct wm_adsp *dsp); bool preloaded; @@ -132,6 +133,8 @@ int wm_adsp_compr_pointer(struct snd_soc_component *component, int wm_adsp_compr_copy(struct snd_soc_component *component, struct snd_compr_stream *stream, char __user *buf, size_t count); + +int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl); int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, From 2c3640b82213cf2beb7c1cc3cfce2ecf5349b0de Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 5 Aug 2024 10:27:21 +0000 Subject: [PATCH 21/39] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. There is no reason that the UCM or user should need to interact with any of the ALSA controls for the firmware coefficients so they can be removed entirely. Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240805102721.30102-3-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 84c34f5b1a51..757ade6373ed 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1095,6 +1095,11 @@ int cs35l56_system_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_resume); +static int cs35l56_control_add_nop(struct wm_adsp *dsp, struct cs_dsp_coeff_ctl *cs_ctl) +{ + return 0; +} + static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) { struct wm_adsp *dsp; @@ -1117,6 +1122,12 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) dsp->fw = 12; dsp->wmfw_optional = true; + /* + * None of the firmware controls need to be exported so add a no-op + * callback that suppresses creating an ALSA control. + */ + dsp->control_add = &cs35l56_control_add_nop; + dev_dbg(cs35l56->base.dev, "DSP system name: '%s'\n", dsp->system_name); ret = wm_halo_init(dsp); From dc268085e499666b9f4f0fcb4c5a94e1c0b193b3 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Aug 2024 12:42:22 +0100 Subject: [PATCH 22/39] ASoC: cs-amp-lib: Fix NULL pointer crash if efi.get_variable is NULL Call efi_rt_services_supported() to check that efi.get_variable exists before calling it. Signed-off-by: Richard Fitzgerald Fixes: 1cad8725f2b9 ("ASoC: cs-amp-lib: Add helpers for factory calibration data") Link: https://patch.msgid.link/20240805114222.15722-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 287ac01a3873..605964af8afa 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -108,7 +108,7 @@ static efi_status_t cs_amp_get_efi_variable(efi_char16_t *name, KUNIT_STATIC_STUB_REDIRECT(cs_amp_get_efi_variable, name, guid, size, buf); - if (IS_ENABLED(CONFIG_EFI)) + if (efi_rt_services_supported(EFI_RT_SUPPORTED_GET_VARIABLE)) return efi.get_variable(name, guid, &attr, size, buf); return EFI_NOT_FOUND; From 15b7a03205b31bc5623378c190d22b7ff60026f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Aug 2024 15:01:28 +0200 Subject: [PATCH 23/39] ALSA: line6: Fix racy access to midibuf There can be concurrent accesses to line6 midibuf from both the URB completion callback and the rawmidi API access. This could be a cause of KMSAN warning triggered by syzkaller below (so put as reported-by here). This patch protects the midibuf call of the former code path with a spinlock for avoiding the possible races. Reported-by: syzbot+78eccfb8b3c9a85fc6c5@syzkaller.appspotmail.com Closes: https://lore.kernel.org/00000000000000949c061df288c5@google.com Cc: Link: https://patch.msgid.link/20240805130129.10872-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/line6/driver.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index f4437015d43a..9df49a880b75 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -286,12 +286,14 @@ static void line6_data_received(struct urb *urb) { struct usb_line6 *line6 = (struct usb_line6 *)urb->context; struct midi_buffer *mb = &line6->line6midi->midibuf_in; + unsigned long flags; int done; if (urb->status == -ESHUTDOWN) return; if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { + spin_lock_irqsave(&line6->line6midi->lock, flags); done = line6_midibuf_write(mb, urb->transfer_buffer, urb->actual_length); @@ -300,12 +302,15 @@ static void line6_data_received(struct urb *urb) dev_dbg(line6->ifcdev, "%d %d buffer overflow - message skipped\n", done, urb->actual_length); } + spin_unlock_irqrestore(&line6->line6midi->lock, flags); for (;;) { + spin_lock_irqsave(&line6->line6midi->lock, flags); done = line6_midibuf_read(mb, line6->buffer_message, LINE6_MIDI_MESSAGE_MAXLEN, LINE6_MIDIBUF_READ_RX); + spin_unlock_irqrestore(&line6->line6midi->lock, flags); if (done <= 0) break; From 9a1af1e218779724ff29ca75f2b9397dc3ed11e7 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Mon, 29 Jul 2024 15:13:51 +0200 Subject: [PATCH 24/39] ASoC: codecs: lpass-macro: fix missing codec version Recent changes that started checking the codec version broke audio on the Lenovo ThinkPad X13s: wsa_macro 3240000.codec: Unsupported Codec version (0) wsa_macro 3240000.codec: probe with driver wsa_macro failed with error -22 rx_macro 3200000.rxmacro: Unsupported Codec version (0) rx_macro 3200000.rxmacro: probe with driver rx_macro failed with error -22 Add the missing codec version to the lookup table so that the codec drivers probe successfully. Note that I'm just assuming that this is a 2.0 codec based on the fact that this device uses the older register layout. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Johan Hovold Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240729131351.27886-1-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-va-macro.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b852cc7ffad9..a62ccd09bacd 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1472,6 +1472,8 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) version = LPASS_CODEC_VERSION_2_0; + if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && core_id_2 == 0x01) + version = LPASS_CODEC_VERSION_2_0; if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) version = LPASS_CODEC_VERSION_2_1; if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) From e42066df07c0fcedebb32ed56f8bc39b4bf86337 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Aug 2024 15:08:39 +0100 Subject: [PATCH 25/39] ASoC: cs35l56: Handle OTP read latency over SoundWire Use the late-read buffer in the CS35L56 SoundWire interface to read OTP memory. The OTP memory has a longer access latency than chip registers and cannot guarantee to return the data value in the SoundWire control response if the bus clock is >4.8 MHz. The Cirrus SoundWire peripheral IP exposes the bridge-to-bus read buffer and status bits. For a read from OTP the bridge status bits are polled to wait for the OTP data to be loaded into the read buffer and the data is then read from there. Signed-off-by: Richard Fitzgerald Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration") Link: https://patch.msgid.link/20240805140839.26042-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 5 +++ sound/soc/codecs/cs35l56-sdw.c | 77 ++++++++++++++++++++++++++++++++++ 2 files changed, 82 insertions(+) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index a6aa112e5741..a51acefa785f 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -277,6 +277,11 @@ static inline int cs35l56_force_sync_asp1_registers_from_cache(struct cs35l56_ba return 0; } +static inline bool cs35l56_is_otp_register(unsigned int reg) +{ + return (reg >> 16) == 3; +} + extern struct regmap_config cs35l56_regmap_i2c; extern struct regmap_config cs35l56_regmap_spi; extern struct regmap_config cs35l56_regmap_sdw; diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index fc03bb7ecae1..7c9a17fe2195 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -23,6 +23,79 @@ /* Register addresses are offset when sent over SoundWire */ #define CS35L56_SDW_ADDR_OFFSET 0x8000 +/* Cirrus bus bridge registers */ +#define CS35L56_SDW_MEM_ACCESS_STATUS 0xd0 +#define CS35L56_SDW_MEM_READ_DATA 0xd8 + +#define CS35L56_SDW_LAST_LATE BIT(3) +#define CS35L56_SDW_CMD_IN_PROGRESS BIT(2) +#define CS35L56_SDW_RDATA_RDY BIT(0) + +#define CS35L56_LATE_READ_POLL_US 10 +#define CS35L56_LATE_READ_TIMEOUT_US 1000 + +static int cs35l56_sdw_poll_mem_status(struct sdw_slave *peripheral, + unsigned int mask, + unsigned int match) +{ + int ret, val; + + ret = read_poll_timeout(sdw_read_no_pm, val, + (val < 0) || ((val & mask) == match), + CS35L56_LATE_READ_POLL_US, CS35L56_LATE_READ_TIMEOUT_US, + false, peripheral, CS35L56_SDW_MEM_ACCESS_STATUS); + if (ret < 0) + return ret; + + if (val < 0) + return val; + + return 0; +} + +static int cs35l56_sdw_slow_read(struct sdw_slave *peripheral, unsigned int reg, + u8 *buf, size_t val_size) +{ + int ret, i; + + reg += CS35L56_SDW_ADDR_OFFSET; + + for (i = 0; i < val_size; i += sizeof(u32)) { + /* Poll for bus bridge idle */ + ret = cs35l56_sdw_poll_mem_status(peripheral, + CS35L56_SDW_CMD_IN_PROGRESS, + 0); + if (ret < 0) { + dev_err(&peripheral->dev, "!CMD_IN_PROGRESS fail: %d\n", ret); + return ret; + } + + /* Reading LSByte triggers read of register to holding buffer */ + sdw_read_no_pm(peripheral, reg + i); + + /* Wait for data available */ + ret = cs35l56_sdw_poll_mem_status(peripheral, + CS35L56_SDW_RDATA_RDY, + CS35L56_SDW_RDATA_RDY); + if (ret < 0) { + dev_err(&peripheral->dev, "RDATA_RDY fail: %d\n", ret); + return ret; + } + + /* Read data from buffer */ + ret = sdw_nread_no_pm(peripheral, CS35L56_SDW_MEM_READ_DATA, + sizeof(u32), &buf[i]); + if (ret) { + dev_err(&peripheral->dev, "Late read @%#x failed: %d\n", reg + i, ret); + return ret; + } + + swab32s((u32 *)&buf[i]); + } + + return 0; +} + static int cs35l56_sdw_read_one(struct sdw_slave *peripheral, unsigned int reg, void *buf) { int ret; @@ -48,6 +121,10 @@ static int cs35l56_sdw_read(void *context, const void *reg_buf, int ret; reg = le32_to_cpu(*(const __le32 *)reg_buf); + + if (cs35l56_is_otp_register(reg)) + return cs35l56_sdw_slow_read(peripheral, reg, buf8, val_size); + reg += CS35L56_SDW_ADDR_OFFSET; if (val_size == 4) From 7e1e206b99f4b3345aeb49d94584a420b7887f1d Mon Sep 17 00:00:00 2001 From: Steven 'Steve' Kendall Date: Tue, 6 Aug 2024 00:08:24 +0000 Subject: [PATCH 26/39] ALSA: hda: Add HP MP9 G4 Retail System AMS to force connect list In recent HP UEFI firmware (likely v2.15 and above, tested on 2.27), these pins are incorrectly set for HDMI/DP audio. Tested on HP MP9 G4 Retail System AMS. Tested audio with two monitors connected via DisplayPort. Link: https://forum.manjaro.org/t/intel-cannon-lake-pch-cavs-conexant-cx20632-no-sound-at-hdmi-or-displayport/133494 Link: https://bbs.archlinux.org/viewtopic.php?id=270523 Signed-off-by: Steven 'Steve' Kendall Cc: Link: https://patch.msgid.link/20240806-hdmi-audio-hp-wrongpins-v2-1-d9eb4ad41043@chromium.org Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 707d203ba652..4e7361d1d518 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,6 +1989,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) } static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x83ef, "HP MP9 G4 Retail System AMS", 1), SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), SND_PCI_QUIRK(0x103c, 0x8711, "HP", 1), From 176fd1511dd9086ab4fa9323cb232177c6235288 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Aug 2024 08:49:16 +0200 Subject: [PATCH 27/39] ALSA: hda/hdmi: Yet more pin fix for HP EliteDesk 800 G4 HP EliteDesk 800 G4 (PCI SSID 103c:83e2) is another Kabylake machine where BIOS misses the HDMI pin initializations. Add the quirk entry. Cc: Link: https://patch.msgid.link/20240806064918.11132-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4e7361d1d518..78042ac2b71f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,6 +1989,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) } static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x83e2, "HP EliteDesk 800 G4", 1), SND_PCI_QUIRK(0x103c, 0x83ef, "HP MP9 G4 Retail System AMS", 1), SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), From eb91c456f3714c336f0812dccab422ec0e72bde4 Mon Sep 17 00:00:00 2001 From: "Dustin L. Howett" Date: Tue, 6 Aug 2024 21:33:51 -0500 Subject: [PATCH 28/39] ALSA: hda/realtek: Add Framework Laptop 13 (Intel Core Ultra) to quirks The Framework Laptop 13 (Intel Core Ultra) has an ALC285 that ships in a similar configuration to the ALC295 in previous models. It requires the same quirk for headset detection. Signed-off-by: Dustin L. Howett Cc: Link: https://patch.msgid.link/20240806-alsa-hda-realtek-add-framework-laptop-13-intel-core-ultra-to-quirks-v1-1-42d6ce2dbf14@howett.net Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1645d21d422f..480e82df7a4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10678,6 +10678,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. From 03898691d42e0170e7d00f07cbe21ce0e9f3a8fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 10:18:01 +0200 Subject: [PATCH 29/39] ALSA: usb-audio: Re-add ScratchAmp quirk entries At the code refactoring of USB-audio quirk handling, I assumed that the quirk entries of Stanton ScratchAmp devices were only about the device name, and moved them completely into the rename table. But it seems that the device requires the quirk entry so that it's probed by the driver itself. This re-adds back the quirk entries of ScratchAmp, but in a minimalistic manner. Fixes: 5436f59bc5bc ("ALSA: usb-audio: Move device rename and profile quirks to an internal table") Link: https://patch.msgid.link/20240808081803.22300-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 73abc38a5400..f13a8d63a019 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2594,6 +2594,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Stanton ScratchAmp */ +{ USB_DEVICE(0x103d, 0x0100) }, +{ USB_DEVICE(0x103d, 0x0101) }, + /* Novation EMS devices */ { USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001), From 23a58b782f864951485d7a0018549729e007cb43 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Krzysztof=20St=C4=99pniak?= Date: Wed, 7 Aug 2024 02:12:19 +0200 Subject: [PATCH 30/39] ASoC: amd: yc: Support mic on Lenovo Thinkpad E14 Gen 6 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Lenovo Thinkpad E14 Gen 6 (model type 21M3) needs a quirk entry for internal mic to work. Signed-off-by: Krzysztof Stępniak Link: https://patch.msgid.link/20240807001219.1147-1-kfs.szk@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index f4bbfffe9fcb..d30752c0dab2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -220,6 +220,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21M3"), + } + }, { .driver_data = &acp6x_card, .matches = { From 4684a2df9c5b3fc914377127faf2515aa9049093 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Wed, 7 Aug 2024 10:53:55 +0800 Subject: [PATCH 31/39] ASoC: codecs: ES8326: button detect issue We find that we need to set snd_jack_types to 0. If not, there will be a probability of button detection errors Signed-off-by: Zhang Yi Link: https://patch.msgid.link/20240807025356.24904-2-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..be3c79232a31 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -843,6 +843,8 @@ static void es8326_jack_detect_handler(struct work_struct *work) es8326_disable_micbias(es8326->component); if (es8326->jack->status & SND_JACK_HEADPHONE) { dev_dbg(comp->dev, "Report hp remove event\n"); + snd_soc_jack_report(es8326->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); From 6675e76a5c441b52b1b983ebb714122087020ebe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 19:02:27 +0200 Subject: [PATCH 32/39] ASoC: amd: yc: Add quirk entry for OMEN by HP Gaming Laptop 16-n0xxx Fix the missing mic on OMEN by HP Gaming Laptop 16-n0xxx by adding the quirk entry with the board ID 8A44. Cc: stable@vger.kernel.org Link: https://bugzilla.suse.com/show_bug.cgi?id=1227182 Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807170249.16490-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index d30752c0dab2..0523c16305db 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -416,6 +416,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8A43"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8A44"), + } + }, { .driver_data = &acp6x_card, .matches = { From 2f3e2c9eaafc272266344d777f8de44f8632e247 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 6 Aug 2024 13:49:28 +0200 Subject: [PATCH 33/39] ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example The reset GPIO of WCD9370/WCD9375 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240806114931.40090-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml index de397d879acc..f94203798f24 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml @@ -42,7 +42,7 @@ examples: pinctrl-names = "default", "sleep"; pinctrl-0 = <&wcd_reset_n>; pinctrl-1 = <&wcd_reset_n_sleep>; - reset-gpios = <&tlmm 83 GPIO_ACTIVE_HIGH>; + reset-gpios = <&tlmm 83 GPIO_ACTIVE_LOW>; vdd-buck-supply = <&vreg_l17b_1p8>; vdd-rxtx-supply = <&vreg_l18b_1p8>; vdd-px-supply = <&vreg_l18b_1p8>; From 55922275702e112652d314a9b6a6ca31d4b7252e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 6 Aug 2024 13:49:29 +0200 Subject: [PATCH 34/39] ASoC: dt-bindings: qcom,wcd934x: Correct reset GPIO polarity in example The reset GPIO of WCD9340/WCD9341 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240806114931.40090-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index beb0ff0245b0..a65b1d1d5fdd 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -199,10 +199,11 @@ additionalProperties: false examples: - | + #include codec@1,0{ compatible = "slim217,250"; reg = <1 0>; - reset-gpios = <&tlmm 64 0>; + reset-gpios = <&tlmm 64 GPIO_ACTIVE_LOW>; slim-ifc-dev = <&wcd9340_ifd>; #sound-dai-cells = <1>; interrupt-parent = <&tlmm>; From 871f1a16fa3506487de24b05d68be45e9185e77a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 6 Aug 2024 13:49:30 +0200 Subject: [PATCH 35/39] ASoC: dt-bindings: qcom,wcd938x: Correct reset GPIO polarity in example The reset GPIO of WCD9380/WCD9385 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240806114931.40090-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml index cf6c3787adfe..10531350c336 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -34,9 +34,10 @@ unevaluatedProperties: false examples: - | + #include codec { compatible = "qcom,wcd9380-codec"; - reset-gpios = <&tlmm 32 0>; + reset-gpios = <&tlmm 32 GPIO_ACTIVE_LOW>; #sound-dai-cells = <1>; qcom,tx-device = <&wcd938x_tx>; qcom,rx-device = <&wcd938x_rx>; From 81f88fddef9cddae6b4e5d9359022c7a2a3e3b6a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 6 Aug 2024 13:49:31 +0200 Subject: [PATCH 36/39] ASoC: dt-bindings: qcom,wcd939x: Correct reset GPIO polarity in example The reset GPIO of WCD9390/WCD9395 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity, instead of IRQ flag (which is a logical mistake on its own). Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240806114931.40090-4-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml index 6e76f6a8634f..c69291f4d575 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml @@ -52,10 +52,10 @@ unevaluatedProperties: false examples: - | - #include + #include codec { compatible = "qcom,wcd9390-codec"; - reset-gpios = <&tlmm 32 IRQ_TYPE_NONE>; + reset-gpios = <&tlmm 32 GPIO_ACTIVE_LOW>; #sound-dai-cells = <1>; qcom,tx-device = <&wcd939x_tx>; qcom,rx-device = <&wcd939x_rx>; From 2f11f61f9d4d5692bcebb9d089429ee0c046e08a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 7 Aug 2024 15:01:40 +0100 Subject: [PATCH 37/39] MAINTAINERS: Update Cirrus Logic parts to linux-sound mailing list Now that most kernel work on sound has moved over to the linux-sound mailing list so should the Cirrus Logic audio parts. Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20240807140140.421359-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 42decde38320..d304054d661e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5306,7 +5306,7 @@ F: drivers/media/cec/i2c/ch7322.c CIRRUS LOGIC AUDIO CODEC DRIVERS M: David Rhodes M: Richard Fitzgerald -L: alsa-devel@alsa-project.org (moderated for non-subscribers) +L: linux-sound@vger.kernel.org L: patches@opensource.cirrus.com S: Maintained F: Documentation/devicetree/bindings/sound/cirrus,cs* @@ -5375,7 +5375,7 @@ F: sound/soc/codecs/lochnagar-sc.c CIRRUS LOGIC MADERA CODEC DRIVERS M: Charles Keepax M: Richard Fitzgerald -L: alsa-devel@alsa-project.org (moderated for non-subscribers) +L: linux-sound@vger.kernel.org L: patches@opensource.cirrus.com S: Supported W: https://github.com/CirrusLogic/linux-drivers/wiki From 5003d0ce5c7da3a02c0aff771f516f99731e7390 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 7 Aug 2024 18:27:03 +0200 Subject: [PATCH 38/39] ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT With PREEMPT_RT enabled a spinlock_t becomes a sleeping lock. This is usually not a problem with spinlocks used in IRQ context since IRQ handlers get threaded. However, if IRQF_ONESHOT is set, the primary handler won't be force-threaded and runs always in hardirq context. This is a problem because spinlock_t requires a preemptible context on PREEMPT_RT. In this particular instance, regmap mmio uses spinlock_t to protect the register access and IRQF_ONESHOT is set on the IRQ. In this case, it is actually better to do everything in threaded handler and it solves the problem with PREEMPT_RT. Reported-by: Arseniy Krasnov Closes: https://lore.kernel.org/linux-amlogic/20240729131652.3012327-1-avkrasnov@salutedevices.com Suggested-by: Sebastian Andrzej Siewior Fixes: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods") Signed-off-by: Jerome Brunet Reviewed-by: Sebastian Andrzej Siewior Link: https://patch.msgid.link/20240807162705.4024136-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 7e6090af720b..75909196b769 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -207,25 +207,18 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) status = FIELD_GET(STATUS1_INT_STS, status); axg_fifo_ack_irq(fifo, status); - /* Use the thread to call period elapsed on nonatomic links */ - if (status & FIFO_INT_COUNT_REPEAT) - return IRQ_WAKE_THREAD; + if (status & ~FIFO_INT_COUNT_REPEAT) + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + if (status & FIFO_INT_COUNT_REPEAT) { + snd_pcm_period_elapsed(ss); + return IRQ_HANDLED; + } return IRQ_NONE; } -static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) -{ - struct snd_pcm_substream *ss = dev_id; - - snd_pcm_period_elapsed(ss); - - return IRQ_HANDLED; -} - int axg_fifo_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *ss) { @@ -251,8 +244,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, - axg_fifo_pcm_irq_block_thread, + /* Use the threaded irq handler only with non-atomic links */ + ret = request_threaded_irq(fifo->irq, NULL, + axg_fifo_pcm_irq_block, IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; From 72776774b55bb59b7b1b09117e915a5030110304 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Wed, 7 Aug 2024 14:26:48 +0000 Subject: [PATCH 39/39] ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value Device tuning files made with early revision tooling may contain configuration that can unmask IRQ signals that are owned by the host. Adding a safe default to the regmap patch ensures that the hardware matches the driver expectations. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240807142648.46932-1-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..bd74fef33d49 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -36,6 +36,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, + { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 },