From 4ddd51ccff911a2e9e961307692532a325f6c78a Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Jul 2024 16:54:53 +0800 Subject: 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) -- cgit From aa4f76ef09a993efa9b5fab6ddf5d6d324baaea3 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Jul 2024 16:54:54 +0800 Subject: 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; } -- cgit From aebb1813c279ce8f3a2dfa3f86def0c0ec1cbb8d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:41 +0200 Subject: 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]; } -- cgit From 3f6fb03dae9c7dfba7670858d29e03c8faaa89fe Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:42 +0200 Subject: 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]; } -- cgit From 74a79977c4e1d09eced33e6e22f875a5bb3fad29 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:43 +0200 Subject: 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]; } -- cgit From eb11c3bb64ad0a05aeacdb01039863aa2aa3614b Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:44 +0200 Subject: 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; -- cgit From 6801ac36f25690e14955f7f9eace1eaa29edbdd0 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:45 +0200 Subject: 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; -- cgit From dcb6631d05152930e2ea70fd2abfd811b0e970b5 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Jul 2024 16:10:46 +0200 Subject: 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; -- cgit From 6b99068d5ea0aa295f15f30afc98db74d056ec7b Mon Sep 17 00:00:00 2001 From: Jerome Audu Date: Sat, 27 Jul 2024 15:40:15 +0200 Subject: 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, -- cgit From c118478665f467e57d06b2354de65974b246b82b Mon Sep 17 00:00:00 2001 From: Bruno Ancona Date: Sun, 28 Jul 2024 22:50:32 -0600 Subject: 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 = { -- cgit From 45d763fe503e6e0f180f873b750aea307e73fdcf Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Fri, 26 Jul 2024 10:11:11 -0500 Subject: 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), -- cgit From 9da8aa3b3ca05b22be5ba312771e6df4366e56cc Mon Sep 17 00:00:00 2001 From: Francesco Dolcini Date: Wed, 31 Jul 2024 13:48:28 +0200 Subject: 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); -- cgit From 7354eb7f1558466e92e926802d36e69e42938ea9 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Wed, 31 Jul 2024 14:21:44 -0700 Subject: 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" -- cgit From becfa08bfefa2cbb22c84d9e583e81387f2f3bf2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:31 +0100 Subject: 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, -- cgit From c8a132e2e032b00828d51141ab34f9aeb24f44ae Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:32 +0100 Subject: 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); -int snd_soc_component_notify_control(struct snd_soc_component *component, - const char * const ctl) +static void soc_get_kcontrol_name(struct snd_soc_component *component, + char *buf, int size, 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); + snprintf(buf, size, "%s %s", component->name_prefix, ctl); else - snprintf(name, ARRAY_SIZE(name), "%s", ctl); + 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) +{ + struct snd_kcontrol *kctl; - 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); -- cgit From 4791c422981350d0de4ad02a14a08b99c766d06f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:33 +0100 Subject: 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; } -- cgit From 93afd028fb5f06a46a32375fd1f0473451eb1c5a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Aug 2024 11:57:34 +0100 Subject: 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) -- cgit From 45b4acab4cac79503663f0a4be9eb3752db04d4b Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 5 Aug 2024 10:27:20 +0000 Subject: 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, -- cgit From 2c3640b82213cf2beb7c1cc3cfce2ecf5349b0de Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 5 Aug 2024 10:27:21 +0000 Subject: 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); -- cgit From dc268085e499666b9f4f0fcb4c5a94e1c0b193b3 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Aug 2024 12:42:22 +0100 Subject: 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; -- cgit From 9a1af1e218779724ff29ca75f2b9397dc3ed11e7 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Mon, 29 Jul 2024 15:13:51 +0200 Subject: 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)) -- cgit From e42066df07c0fcedebb32ed56f8bc39b4bf86337 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Aug 2024 15:08:39 +0100 Subject: 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) -- cgit