From f101583fa9f8c3f372d4feb61d67da0ccbf4d9a5 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 7 Sep 2023 20:32:24 +0530 Subject: [PATCH 01/79] ASoC: soc-utils: Export snd_soc_dai_is_dummy() symbol Export symbol snd_soc_dai_is_dummy() for usage outside core driver modules. This is required by Tegra ASoC machine driver. Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1694098945-32760-2-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 11607c5f5d5a..9c746e4edef7 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -217,6 +217,7 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) return 1; return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy); int snd_soc_component_is_dummy(struct snd_soc_component *component) { From e765886249c533e1bb5cbc3cd741bad677417312 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 7 Sep 2023 20:32:25 +0530 Subject: [PATCH 02/79] ASoC: tegra: Fix redundant PLLA and PLLA_OUT0 updates Tegra audio graph card has many DAI links which connects internal AHUB modules and external audio codecs. Since these are DPCM links, hw_params() call in the machine driver happens for each connected BE link and PLLA is updated every time. This is not really needed for all links as only I/O link DAIs derive respective clocks from PLLA_OUT0 and thus from PLLA. Hence add checks to limit the clock updates to DAIs over I/O links. This found to be fixing a DMIC clock discrepancy which is suspected to happen because of back to back quick PLLA and PLLA_OUT0 rate updates. This was observed on Jetson TX2 platform where DMIC clock ended up with unexpected value. Fixes: 202e2f774543 ("ASoC: tegra: Add audio graph based card driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1694098945-32760-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_audio_graph_card.c | 30 ++++++++++++++---------- 1 file changed, 17 insertions(+), 13 deletions(-) diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index 1f2c5018bf5a..4737e776d383 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -10,6 +10,7 @@ #include #include #include +#include #define MAX_PLLA_OUT0_DIV 128 @@ -44,6 +45,21 @@ struct tegra_audio_cdata { unsigned int plla_out0_rates[NUM_RATE_TYPE]; }; +static bool need_clk_update(struct snd_soc_dai *dai) +{ + if (snd_soc_dai_is_dummy(dai) || + !dai->driver->ops || + !dai->driver->name) + return false; + + if (strstr(dai->driver->name, "I2S") || + strstr(dai->driver->name, "DMIC") || + strstr(dai->driver->name, "DSPK")) + return true; + + return false; +} + /* Setup PLL clock as per the given sample rate */ static int tegra_audio_graph_update_pll(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -140,19 +156,7 @@ static int tegra_audio_graph_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; - /* - * This gets called for each DAI link (FE or BE) when DPCM is used. - * We may not want to update PLLA rate for each call. So PLLA update - * must be restricted to external I/O links (I2S, DMIC or DSPK) since - * they actually depend on it. I/O modules update their clocks in - * hw_param() of their respective component driver and PLLA rate - * update here helps them to derive appropriate rates. - * - * TODO: When more HW accelerators get added (like sample rate - * converter, volume gain controller etc., which don't really - * depend on PLLA) we need a better way to filter here. - */ - if (cpu_dai->driver->ops && rtd->dai_link->no_pcm) { + if (need_clk_update(cpu_dai)) { err = tegra_audio_graph_update_pll(substream, params); if (err) return err; From ec03804552e9a723569e14d2512f36a8e70dc640 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 8 Sep 2023 11:17:16 +0100 Subject: [PATCH 03/79] ASoC: cs35l56: Call pm_runtime_dont_use_autosuspend() Driver remove() must call pm_runtime_dont_use_autosuspend(). Drivers that call pm_runtime_use_autosuspend() must disable it in driver remove(). Unfortunately until recently this was only mentioned in 1 line in a 900+ line document so most people hadn't noticed this. It has only recently been added to the kerneldoc of pm_runtime_use_autosuspend(). THIS WON'T APPLY CLEANLY TO V6.5 AND EARLIER: We will send a separate backported patch to stable. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20230908101716.2658582-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 600b79c62ec4..f2e7c6d0be46 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1207,6 +1207,7 @@ void cs35l56_remove(struct cs35l56_private *cs35l56) flush_workqueue(cs35l56->dsp_wq); destroy_workqueue(cs35l56->dsp_wq); + pm_runtime_dont_use_autosuspend(cs35l56->base.dev); pm_runtime_suspend(cs35l56->base.dev); pm_runtime_disable(cs35l56->base.dev); From aedf323b66b2b875137422ecb7d2525179759076 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 7 Sep 2023 11:05:04 +0200 Subject: [PATCH 04/79] ASoC: meson: spdifin: start hw on dai probe For spdif input to report the locked rate correctly, even when no capture is running, the HW and reference clock must be started as soon as the dai is probed. Fixes: 5ce5658375e6 ("ASoC: meson: add axg spdif input") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20230907090504.12700-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-spdifin.c | 49 ++++++++++++----------------------- 1 file changed, 17 insertions(+), 32 deletions(-) diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d86880169075..bc2f2849ecfb 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -112,34 +112,6 @@ static int axg_spdifin_prepare(struct snd_pcm_substream *substream, return 0; } -static int axg_spdifin_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = clk_prepare_enable(priv->refclk); - if (ret) { - dev_err(dai->dev, - "failed to enable spdifin reference clock\n"); - return ret; - } - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, - SPDIFIN_CTRL0_EN); - - return 0; -} - -static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); - clk_disable_unprepare(priv->refclk); -} - static void axg_spdifin_write_mode_param(struct regmap *map, int mode, unsigned int val, unsigned int num_per_reg, @@ -251,17 +223,32 @@ static int axg_spdifin_dai_probe(struct snd_soc_dai *dai) ret = axg_spdifin_sample_mode_config(dai, priv); if (ret) { dev_err(dai->dev, "mode configuration failed\n"); - clk_disable_unprepare(priv->pclk); - return ret; + goto pclk_err; } + ret = clk_prepare_enable(priv->refclk); + if (ret) { + dev_err(dai->dev, + "failed to enable spdifin reference clock\n"); + goto pclk_err; + } + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, + SPDIFIN_CTRL0_EN); + return 0; + +pclk_err: + clk_disable_unprepare(priv->pclk); + return ret; } static int axg_spdifin_dai_remove(struct snd_soc_dai *dai) { struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); + clk_disable_unprepare(priv->refclk); clk_disable_unprepare(priv->pclk); return 0; } @@ -270,8 +257,6 @@ static const struct snd_soc_dai_ops axg_spdifin_ops = { .probe = axg_spdifin_dai_probe, .remove = axg_spdifin_dai_remove, .prepare = axg_spdifin_prepare, - .startup = axg_spdifin_startup, - .shutdown = axg_spdifin_shutdown, }; static int axg_spdifin_iec958_info(struct snd_kcontrol *kcontrol, From 28115b1c4f2bb76e786436bf6597c5eb27638a5c Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 7 Sep 2023 11:55:20 +0200 Subject: [PATCH 05/79] ASoC: rsnd: add missing of_node_put for_each_child_of_node performs an of_node_get on each iteration, so a break out of the loop requires an of_node_put. This was done using the Coccinelle semantic patch iterators/for_each_child.cocci Signed-off-by: Julia Lawall Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20230907095521.14053-11-Julia.Lawall@inria.fr Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e29c2fee9521..1bd7114c472a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1303,6 +1303,7 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); of_node_put(node); + of_node_put(ports); break; } } From d7e47e32192bb88f5b2dc8e655fa587ecf9d71e0 Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Sat, 9 Sep 2023 05:02:37 -0700 Subject: [PATCH 06/79] ASoC: wm8960: Fix error handling in probe Commit 422f10adc3eb ("ASoC: wm8960: Add support for the power supplies") added regulator support to the wm8960 driver, but neglected to update error handling in the probe function. This results in warning backtraces if the probe function fails. Fixes: 422f10adc3eb ("ASoC: wm8960: Add support for the power supplies") Signed-off-by: Guenter Roeck Reviewed-by: Fabio Estevam Link: https://lore.kernel.org/r/20230909120237.2646275-1-linux@roeck-us.net Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a50180750e8..7689fe3cc86d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1468,8 +1468,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) } wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); - if (IS_ERR(wm8960->regmap)) - return PTR_ERR(wm8960->regmap); + if (IS_ERR(wm8960->regmap)) { + ret = PTR_ERR(wm8960->regmap); + goto bulk_disable; + } if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); @@ -1479,13 +1481,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) ret = i2c_master_recv(i2c, &val, sizeof(val)); if (ret >= 0) { dev_err(&i2c->dev, "Not wm8960, wm8960 reg can not read by i2c\n"); - return -EINVAL; + ret = -EINVAL; + goto bulk_disable; } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); - return ret; + goto bulk_disable; } if (wm8960->pdata.shared_lrclk) { @@ -1494,7 +1497,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) if (ret != 0) { dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", ret); - return ret; + goto bulk_disable; } } @@ -1528,7 +1531,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8960, &wm8960_dai, 1); + if (ret) + goto bulk_disable; + return 0; + +bulk_disable: + regulator_bulk_disable(ARRAY_SIZE(wm8960->supplies), wm8960->supplies); return ret; } From 396b907919e028d89bac912e49de014485deb8dc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 8 Sep 2023 09:59:20 +0100 Subject: [PATCH 07/79] ASoC: soc-pcm: Shrink stack frame for __soc_pcm_hw_params Commit ac950278b087 ("ASoC: add N cpus to M codecs dai link support") added an additional local params in __soc_pcm_hw_params, for the CPU side of the DAI. The snd_pcm_hw_params struct is pretty large (604 bytes) and keeping two local copies of it can make the stack frame really large. It is worth noting the variables are in separate code blocks so for some optimisation levels in the compiler these will get automatically combined keeping the stack frame reasonable. But better to manually combine them to cover all cases. Add a single local variable for __soc_pcm_hw_params and use in both loops to shrink the stack frame. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20230908085920.2906359-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb0723876851..54704250c0a2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -985,6 +985,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; + struct snd_pcm_hw_params tmp_params; int i, ret = 0; snd_soc_dpcm_mutex_assert_held(rtd); @@ -998,7 +999,6 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { - struct snd_pcm_hw_params codec_params; unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream); /* @@ -1019,23 +1019,22 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, continue; /* copy params for each codec */ - codec_params = *params; + tmp_params = *params; /* fixup params based on TDM slot masks */ if (tdm_mask) - soc_pcm_codec_params_fixup(&codec_params, tdm_mask); + soc_pcm_codec_params_fixup(&tmp_params, tdm_mask); ret = snd_soc_dai_hw_params(codec_dai, substream, - &codec_params); + &tmp_params); if(ret < 0) goto out; - soc_pcm_set_dai_params(codec_dai, &codec_params); - snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); + soc_pcm_set_dai_params(codec_dai, &tmp_params); + snd_soc_dapm_update_dai(substream, &tmp_params, codec_dai); } for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - struct snd_pcm_hw_params cpu_params; unsigned int ch_mask = 0; int j; @@ -1047,7 +1046,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, continue; /* copy params for each cpu */ - cpu_params = *params; + tmp_params = *params; if (!rtd->dai_link->codec_ch_maps) goto hw_params; @@ -1062,16 +1061,16 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, /* fixup cpu channel number */ if (ch_mask) - soc_pcm_codec_params_fixup(&cpu_params, ch_mask); + soc_pcm_codec_params_fixup(&tmp_params, ch_mask); hw_params: - ret = snd_soc_dai_hw_params(cpu_dai, substream, &cpu_params); + ret = snd_soc_dai_hw_params(cpu_dai, substream, &tmp_params); if (ret < 0) goto out; /* store the parameters for each DAI */ - soc_pcm_set_dai_params(cpu_dai, &cpu_params); - snd_soc_dapm_update_dai(substream, &cpu_params, cpu_dai); + soc_pcm_set_dai_params(cpu_dai, &tmp_params); + snd_soc_dapm_update_dai(substream, &tmp_params, cpu_dai); } ret = snd_soc_pcm_component_hw_params(substream, params); From e616a916fe8431ebd5eb3cf4ac224d143c57083c Mon Sep 17 00:00:00 2001 From: Walt Holman Date: Sun, 10 Sep 2023 13:54:34 -0500 Subject: [PATCH 08/79] Add DMI ID for MSI Bravo 15 B7ED Signed-off-by: Walt Holman Link: https://lore.kernel.org/r/20230910185433.13677-1-waltholman09@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 3ec15b46fa35..59aa2e9d3a79 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -262,6 +262,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"), + } + }, { .driver_data = &acp6x_card, .matches = { From 2f9426905a63be7ccf8cd10109caf1848aa0993a Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 11 Sep 2023 14:38:07 +0800 Subject: [PATCH 09/79] ASoC: fsl: imx-pcm-rpmsg: Add SNDRV_PCM_INFO_BATCH flag The rpmsg pcm device is a device which should support double buffering. Found this issue with pipewire. When there is no SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will set headroom to be zero, and because rpmsg pcm device don't support residue report, when the latency setting is small, the "delay" always larger than "target" in alsa-pcm.c, that reading next period data is not scheduled on time. With SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will select a smaller period size for device, then the task of reading next period data will be scheduled on time. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1694414287-13291-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-rpmsg.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index d63782b8bdef..bb736d45c9e0 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -19,6 +19,7 @@ static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | From 1263cc0f414d212129c0f1289b49b7df77f92084 Mon Sep 17 00:00:00 2001 From: August Wikerfors Date: Mon, 11 Sep 2023 23:34:09 +0200 Subject: [PATCH 10/79] ASoC: amd: yc: Fix non-functional mic on Lenovo 82QF and 82UG Like the Lenovo 82TL and 82V2, the Lenovo 82QF (Yoga 7 14ARB7) and 82UG (Legion S7 16ARHA7) both need a quirk entry for the internal microphone to function. Commit c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") restricted the quirk that previously matched "82" to "82V2", breaking microphone functionality on these devices. Fix this by adding specific quirks for these models, as was done for the Lenovo 82TL. Fixes: c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") Closes: https://github.com/tomsom/yoga-linux/issues/51 Link: https://bugzilla.kernel.org/show_bug.cgi?id=208555#c780 Cc: stable@vger.kernel.org Signed-off-by: August Wikerfors Link: https://lore.kernel.org/r/20230911213409.6106-1-git@augustwikerfors.se Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 59aa2e9d3a79..94e9eb8e73f2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ 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, "82QF"), + } + }, { .driver_data = &acp6x_card, .matches = { @@ -220,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "82TL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82UG"), + } + }, { .driver_data = &acp6x_card, .matches = { From fa6a0c0c1dd53b3949ca56bf7213648dfd6a62ee Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 12 Sep 2023 13:32:40 +0200 Subject: [PATCH 11/79] ASoC: rt5640: Revert "Fix sleep in atomic context" Commit 70a6404ff610 ("ASoC: rt5640: Fix sleep in atomic context") not only switched from request_irq() to request_threaded_irq(), to fix the sleep in atomic context issue, but it also added devm management of the IRQ by actually switching to devm_request_threaded_irq() (without any explanation in the commit message for this change). This is wrong since the IRQ was already explicitly managed by the driver. On unbind the ASoC core will call rt5640_set_jack(NULL) which in turn will call rt5640_disable_jack_detect() which frees the IRQ already. So now we have a double free. Besides the unexplained switch to devm being wrong, the actual fix for the sleep in atomic context issue also is not the best solution. The only thing which rt5640_irq() does is cancel + (re-)queue the jack_work delayed_work. This can be done in a single non sleeping call by replacing queue_delayed_work() with mod_delayed_work(), which does not sleep. Using mod_delayed_work() is a much better fix then adding a thread which does nothing other then queuing a work-item. This patch is a straight revert of the troublesome changes, the switch to mod_delayed_work() is done in a separate follow-up patch. Fixes: 70a6404ff610 ("ASoC: rt5640: Fix sleep in atomic context") Cc: Sameer Pujar Cc: Oder Chiou Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20230912113245.320159-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 15e1a62b9e57..05ff8066171b 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2565,10 +2565,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, if (jack_data && jack_data->use_platform_clock) rt5640->use_platform_clock = jack_data->use_platform_clock; - ret = devm_request_threaded_irq(component->dev, rt5640->irq, - NULL, rt5640_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "rt5640", rt5640); + ret = request_irq(rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640_disable_jack_detect(component); @@ -2621,9 +2620,8 @@ static void rt5640_enable_hda_jack_detect( rt5640->jack = jack; - ret = devm_request_threaded_irq(component->dev, rt5640->irq, - NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT, - "rt5640", rt5640); + ret = request_irq(rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640->irq = -ENXIO; From df7d595f6bd9dc96cc275cc4b0f313fcfa423c58 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 12 Sep 2023 13:32:41 +0200 Subject: [PATCH 12/79] ASoC: rt5640: Fix sleep in atomic context Following prints are observed while testing audio on Jetson AGX Orin which has onboard RT5640 audio codec: BUG: sleeping function called from invalid context at kernel/workqueue.c:3027 in_atomic(): 1, irqs_disabled(): 128, non_block: 0, pid: 0, name: swapper/0 preempt_count: 10001, expected: 0 RCU nest depth: 0, expected: 0 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 0 at kernel/irq/handle.c:159 __handle_irq_event_percpu+0x1e0/0x270 ---[ end trace ad1c64905aac14a6 ]- The IRQ handler rt5640_irq() runs in interrupt context and can sleep during cancel_delayed_work_sync(). The only thing which rt5640_irq() does is cancel + (re-)queue the jack_work delayed_work. This can be done in a single non sleeping call by replacing queue_delayed_work() with mod_delayed_work(), avoiding the sleep in atomic context. Fixes: 051dade34695 ("ASoC: rt5640: Fix the wrong state of JD1 and JD2") Reported-by: Sameer Pujar Closes: https://lore.kernel.org/r/1688015537-31682-4-git-send-email-spujar@nvidia.com Cc: Oder Chiou Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20230912113245.320159-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 05ff8066171b..5c34c045d396 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2403,13 +2403,11 @@ static irqreturn_t rt5640_irq(int irq, void *data) struct rt5640_priv *rt5640 = data; int delay = 0; - if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { - cancel_delayed_work_sync(&rt5640->jack_work); + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) delay = 100; - } if (rt5640->jack) - queue_delayed_work(system_long_wq, &rt5640->jack_work, delay); + mod_delayed_work(system_long_wq, &rt5640->jack_work, delay); return IRQ_HANDLED; } From 786120ebb649b166021f0212250e8627e53d068a Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 12 Sep 2023 13:32:42 +0200 Subject: [PATCH 13/79] ASoC: rt5640: Do not disable/enable IRQ twice on suspend/resume When jack-detect was originally added disabling the IRQ during suspend was done by the sound/soc/intel/boards/bytcr_rt5640.c driver calling snd_soc_component_set_jack(NULL) on suspend, which calls rt5640_disable_jack_detect(), which calls free_irq() which also disables it. Commit 5fabcc90e79b ("ASoC: rt5640: Fix Jack work after system suspend") added disable_irq() / enable_irq() calls on suspend/resume for machine drivers which do not call snd_soc_component_set_jack(NULL) on suspend. The new disable_irq() / enable_irq() are made conditional by "if (rt5640->irq)" statements, but this is true for the machine drivers which do call snd_soc_component_set_jack(NULL) on suspend too, causing a disable_irq() call there on the already free-ed IRQ. Change the "if (rt5640->irq)" condition to "if (rt5640->jack)" to fix this, rt5640->jack is only set if the jack-detect IRQ handler is still active when rt5640_suspend() runs. And adjust rt5640_enable_hda_jack_detect()'s request_irq() error handling to set rt5640->jack to NULL to match (note that the old setting of irq to -ENOXIO still resulted in disable_irq(-ENOXIO) calls on suspend). Fixes: 5fabcc90e79b ("ASoC: rt5640: Fix Jack work after system suspend") Cc: Oder Chiou Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20230912113245.320159-4-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 5c34c045d396..1bc281d42ca8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2622,7 +2622,7 @@ static void rt5640_enable_hda_jack_detect( IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); - rt5640->irq = -ENXIO; + rt5640->jack = NULL; return; } @@ -2797,7 +2797,7 @@ static int rt5640_suspend(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); - if (rt5640->irq) { + if (rt5640->jack) { /* disable jack interrupts during system suspend */ disable_irq(rt5640->irq); } @@ -2825,10 +2825,9 @@ static int rt5640_resume(struct snd_soc_component *component) regcache_cache_only(rt5640->regmap, false); regcache_sync(rt5640->regmap); - if (rt5640->irq) + if (rt5640->jack) { enable_irq(rt5640->irq); - if (rt5640->jack) { if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { snd_soc_component_update_bits(component, RT5640_DUMMY2, 0x1100, 0x1100); From b5e85e535551bf82242aa5896e14a136ed3c156d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 12 Sep 2023 13:32:43 +0200 Subject: [PATCH 14/79] ASoC: rt5640: Enable the IRQ on resume after configuring jack-detect The jack-detect IRQ should be enabled *after* the jack-detect related configuration registers have been programmed. Move the enable_irq() call for this to after the register setup. Fixes: 5fabcc90e79b ("ASoC: rt5640: Fix Jack work after system suspend") Cc: Oder Chiou Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20230912113245.320159-5-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1bc281d42ca8..03c866c04c7a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2826,8 +2826,6 @@ static int rt5640_resume(struct snd_soc_component *component) regcache_sync(rt5640->regmap); if (rt5640->jack) { - enable_irq(rt5640->irq); - if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { snd_soc_component_update_bits(component, RT5640_DUMMY2, 0x1100, 0x1100); @@ -2854,6 +2852,7 @@ static int rt5640_resume(struct snd_soc_component *component) } } + enable_irq(rt5640->irq); queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } From 8c8bf3df6b7c0ed1c4dd373b23eb0ce13a63f452 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 12 Sep 2023 13:32:44 +0200 Subject: [PATCH 15/79] ASoC: rt5640: Fix IRQ not being free-ed for HDA jack detect mode Set "rt5640->irq_requested = true" after a successful request_irq() in rt5640_enable_hda_jack_detect(), so that rt5640_disable_jack_detect() properly frees the IRQ. This fixes the IRQ not being freed on rmmod / driver unbind. Fixes: 2b9c8d2b3c89 ("ASoC: rt5640: Add the HDA header support") Cc: Oder Chiou Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20230912113245.320159-6-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 03c866c04c7a..a4a11407ab10 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2625,6 +2625,7 @@ static void rt5640_enable_hda_jack_detect( rt5640->jack = NULL; return; } + rt5640->irq_requested = true; /* sync initial jack state */ queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); From 8fc7cc507d61fc655172836c74fb7fcc8b7a978b Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 12 Sep 2023 13:32:45 +0200 Subject: [PATCH 16/79] ASoC: rt5640: Only cancel jack-detect work on suspend if active If jack-detection is not used; or has already been disabled then there is no need to call rt5640_cancel_work(). Move the rt5640_cancel_work() inside the "if (rt5640->jack) {}" block, grouping it together with the disabling of the IRQ which queues the work in the first place. This also makes suspend() symetrical with resume() which re-queues the work in an "if (rt5640->jack) {}" block. Cc: Oder Chiou Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20230912113245.320159-7-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a4a11407ab10..e8cdc166bdaa 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2801,9 +2801,9 @@ static int rt5640_suspend(struct snd_soc_component *component) if (rt5640->jack) { /* disable jack interrupts during system suspend */ disable_irq(rt5640->irq); + rt5640_cancel_work(rt5640); } - rt5640_cancel_work(rt5640); snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); rt5640_reset(component); regcache_cache_only(rt5640->regmap, true); From 18789be8e0d9fbb78b2290dcf93f500726ed19f0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 12 Sep 2023 14:38:41 +0100 Subject: [PATCH 17/79] ASoC: cs35l56: Disable low-power hibernation mode Do not allow the CS35L56 to be put into its lowest power "hibernation" mode. This only affects I2C because "hibernation" is already disabled on SPI and SoundWire. Recent firmwares need a different wake-up sequence. Until that sequence has been specified, the chip "hibernation" mode must be disabled otherwise it can intermittently fail to wake. THIS WILL NOT APPLY CLEANLY TO 6.5 AND EARLIER: We will send a separate backport patch to stable. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20230912133841.3480466-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 9f4f2f4f23f5..d10e0e2380e8 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return -ENOMEM; cs35l56->base.dev = dev; - cs35l56->base.can_hibernate = true; i2c_set_clientdata(client, cs35l56); cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config); From cf0ba445f5e4dd74c1e9d7a83ca721ba69204a11 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 13 Sep 2023 11:18:22 +0300 Subject: [PATCH 18/79] ASoC: codecs: aw88395: Fix some error codes These error paths should return -EINVAL instead of success. Fixes: 7f4ec77802aa ("ASoC: codecs: Add code for bin parsing compatible with aw88261") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/81476e78-05c2-4656-b754-f314c7ccdb81@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/codecs/aw88395/aw88395_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 8ee1baa03269..87dd0ccade4c 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -452,11 +452,13 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, if ((aw_bin->all_bin_parse_num != 1) || (aw_bin->header_info[0].bin_data_type != DATA_TYPE_REGISTER)) { dev_err(aw_dev->dev, "bin num or type error"); + ret = -EINVAL; goto parse_bin_failed; } if (aw_bin->header_info[0].valid_data_len % 4) { dev_err(aw_dev->dev, "bin data len get error!"); + ret = -EINVAL; goto parse_bin_failed; } From 41dac81b56c82c51a6d00fda5f3af7691ffee2d7 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 13 Sep 2023 16:00:10 +0100 Subject: [PATCH 19/79] ASoC: cs42l42: Ensure a reset pulse meets minimum pulse width. The CS42L42 can accept very short reset pulses of a few microseconds but there's no reason to force a very short pulse. Allow a wide range for the usleep_range() so it can be relaxed about the choice of timing source. Signed-off-by: Richard Fitzgerald Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20230913150012.604775-2-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a0de0329406a..56d2857a4f01 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2320,6 +2320,10 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42, if (cs42l42->reset_gpio) { dev_dbg(cs42l42->dev, "Found reset GPIO\n"); + + /* Ensure minimum reset pulse width */ + usleep_range(10, 500); + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); From a479b44ac0a0ac25cd48e5356200078924d78022 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 13 Sep 2023 16:00:11 +0100 Subject: [PATCH 20/79] ASoC: cs42l42: Don't rely on GPIOD_OUT_LOW to set RESET initially low The ACPI setting for a GPIO default state has higher priority than the flag passed to devm_gpiod_get_optional() so ACPI can override the GPIOD_OUT_LOW. Explicitly set the GPIO low when hard resetting. Although GPIOD_OUT_LOW can't be relied on this doesn't seem like a reason to stop passing it to devm_gpiod_get_optional(). So we still pass it to state our intent, but can deal with it having no effect. Signed-off-by: Richard Fitzgerald Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20230913150012.604775-3-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 56d2857a4f01..dc93861ddfb0 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2321,6 +2321,12 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42, if (cs42l42->reset_gpio) { dev_dbg(cs42l42->dev, "Found reset GPIO\n"); + /* + * ACPI can override the default GPIO state we requested + * so ensure that we start with RESET low. + */ + gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); + /* Ensure minimum reset pulse width */ usleep_range(10, 500); From 2d066c6a78654c179f95c9beda1985d4c6befa4e Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 13 Sep 2023 16:00:12 +0100 Subject: [PATCH 21/79] ASoC: cs42l42: Avoid stale SoundWire ATTACH after hard reset In SoundWire mode leave hard RESET asserted when exiting probe, and wait for an UNATTACHED notification before deasserting RESET. If the boot state of the reset GPIO was deasserted it is possible that the SoundWire core had already enumerated the CS42L42 before cs42l42_sdw_probe() is called. When cs42l42_common_probe() hard resets the CS42L42 it triggers a race condition: 1) After cs42l42_sdw_probe() returns the thread that called it will call cs42l42_sdw_update_status() to report the last status recorded by the SoundWire core. 2) The SoundWire bus master will see a PING with the CS42L42 now reporting as unenumerated and will trigger the core SoundWire code to start enumerating CS42L42. These two threads are racing against each other. If (1) happens before (2) a stale ATTACHED notification will be reported to the cs42l42 driver when in fact the status of cs42l42 is now unattached. To avoid this race condition: - Leave RESET asserted on exit from cs42l42_sdw_probe(). This ensures that an UNATTACHED notification must be sent to the cs42l42 driver. If cs42l42 was already enumerated it will be seen to drop off the bus, causing an UNATTACH notification. If it was never enumerated the status is already UNATTACHED and this will be reported by thread (1). - When the UNATTACH notification is received, release RESET. This will cause CS42L42 to be enumerated and eventually report an ATTACHED notification. - The ATTACHED notification is now valid. Signed-off-by: Richard Fitzgerald Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20230913150012.604775-4-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42-sdw.c | 20 ++++++++++++++++++++ sound/soc/codecs/cs42l42.c | 11 ++++++++++- sound/soc/codecs/cs42l42.h | 1 + 3 files changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index eeab07c850f9..974bae4abfad 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -344,6 +344,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral, switch (status) { case SDW_SLAVE_ATTACHED: dev_dbg(cs42l42->dev, "ATTACHED\n"); + + /* + * The SoundWire core can report stale ATTACH notifications + * if we hard-reset CS42L42 in probe() but it had already been + * enumerated. Reject the ATTACH if we haven't yet seen an + * UNATTACH report for the device being in reset. + */ + if (cs42l42->sdw_waiting_first_unattach) + break; + /* * Initialise codec, this only needs to be done once. * When resuming from suspend, resume callback will handle re-init of codec, @@ -354,6 +364,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral, break; case SDW_SLAVE_UNATTACHED: dev_dbg(cs42l42->dev, "UNATTACHED\n"); + + if (cs42l42->sdw_waiting_first_unattach) { + /* + * SoundWire core has seen that CS42L42 is not on + * the bus so release RESET and wait for ATTACH. + */ + cs42l42->sdw_waiting_first_unattach = false; + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + } + break; default: break; diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index dc93861ddfb0..2961340f15e2 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2330,7 +2330,16 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42, /* Ensure minimum reset pulse width */ usleep_range(10, 500); - gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + /* + * On SoundWire keep the chip in reset until we get an UNATTACH + * notification from the SoundWire core. This acts as a + * synchronization point to reject stale ATTACH notifications + * if the chip was already enumerated before we reset it. + */ + if (cs42l42->sdw_peripheral) + cs42l42->sdw_waiting_first_unattach = true; + else + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 4bd7b85a5747..7785125b73ab 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -53,6 +53,7 @@ struct cs42l42_private { u8 stream_use; bool hp_adc_up_pending; bool suspended; + bool sdw_waiting_first_unattach; bool init_done; }; From 69343ce91435f222052015c5af86b550391bac85 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 13 Sep 2023 17:05:23 +0100 Subject: [PATCH 22/79] firmware: cirrus: cs_dsp: Only log list of algorithms in debug build Change the logging of each algorithm from info level to debug level. On the original devices supported by this code there were typically only one or two algorithms in a firmware and one or two DSPs so this logging only used a small number of log lines. However, for the latest devices there could be 30-40 algorithms in a firmware and 8 DSPs being loaded in parallel, so using 300+ lines of log for information that isn't particularly important to have logged. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20230913160523.3701189-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- drivers/firmware/cirrus/cs_dsp.c | 34 ++++++++++++++++---------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 49b70c70dc69..79d4254d1f9b 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1863,15 +1863,15 @@ static int cs_dsp_adsp2_setup_algs(struct cs_dsp *dsp) return PTR_ERR(adsp2_alg); for (i = 0; i < n_algs; i++) { - cs_dsp_info(dsp, - "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n", - i, be32_to_cpu(adsp2_alg[i].alg.id), - (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, - be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff, - be32_to_cpu(adsp2_alg[i].xm), - be32_to_cpu(adsp2_alg[i].ym), - be32_to_cpu(adsp2_alg[i].zm)); + cs_dsp_dbg(dsp, + "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n", + i, be32_to_cpu(adsp2_alg[i].alg.id), + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff, + be32_to_cpu(adsp2_alg[i].xm), + be32_to_cpu(adsp2_alg[i].ym), + be32_to_cpu(adsp2_alg[i].zm)); alg_region = cs_dsp_create_region(dsp, WMFW_ADSP2_XM, adsp2_alg[i].alg.id, @@ -1996,14 +1996,14 @@ static int cs_dsp_halo_setup_algs(struct cs_dsp *dsp) return PTR_ERR(halo_alg); for (i = 0; i < n_algs; i++) { - cs_dsp_info(dsp, - "%d: ID %x v%d.%d.%d XM@%x YM@%x\n", - i, be32_to_cpu(halo_alg[i].alg.id), - (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16, - (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8, - be32_to_cpu(halo_alg[i].alg.ver) & 0xff, - be32_to_cpu(halo_alg[i].xm_base), - be32_to_cpu(halo_alg[i].ym_base)); + cs_dsp_dbg(dsp, + "%d: ID %x v%d.%d.%d XM@%x YM@%x\n", + i, be32_to_cpu(halo_alg[i].alg.id), + (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(halo_alg[i].alg.ver) & 0xff, + be32_to_cpu(halo_alg[i].xm_base), + be32_to_cpu(halo_alg[i].ym_base)); ret = cs_dsp_halo_create_regions(dsp, halo_alg[i].alg.id, halo_alg[i].alg.ver, From 781118bc2fc1026c8285f83ea7ecab07071a09c4 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 13 Sep 2023 17:02:50 +0100 Subject: [PATCH 23/79] ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl() wm_adsp_read_ctl() and wm_adsp_write_ctl() must hold the cs_dsp pwr_lock mutex when calling cs_dsp_coeff_read_ctrl() and cs_dsp_coeff_write_ctrl(). Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20230913160250.3700346-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6fc34f41b175..d1b9238d391e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -687,7 +687,10 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, struct wm_coeff_ctl *ctl; int ret; + mutex_lock(&dsp->cs_dsp.pwr_lock); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); + mutex_unlock(&dsp->cs_dsp.pwr_lock); + if (ret < 0) return ret; @@ -703,8 +706,14 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), - 0, buf, len); + int ret; + + mutex_lock(&dsp->cs_dsp.pwr_lock); + ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), + 0, buf, len); + mutex_unlock(&dsp->cs_dsp.pwr_lock); + + return ret; } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); From fac58baf8fcfcd7481e8f6d60206ce2a47c1476c Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Wed, 13 Sep 2023 18:26:56 +0800 Subject: [PATCH 24/79] ASoC: imx-rpmsg: Set ignore_pmdown_time for dai_link i.MX rpmsg sound cards work on codec slave mode. MCLK will be disabled by CPU DAI driver in hw_free(). Some codec requires MCLK present at power up/down sequence. So need to set ignore_pmdown_time to power down codec immediately before MCLK is turned off. Take WM8962 as an example, if MCLK is disabled before DAPM power down playback stream, FIFO error will arise in WM8962 which will have bad impact on playback next. Signed-off-by: Chancel Liu Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20230913102656.2966757-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-rpmsg.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 3c7b95db2eac..b578f9a32d7f 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -89,6 +89,14 @@ static int imx_rpmsg_probe(struct platform_device *pdev) SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + /* + * i.MX rpmsg sound cards work on codec slave mode. MCLK will be + * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK + * present at power up/down sequence. So need to set ignore_pmdown_time + * to power down codec immediately before MCLK is turned off. + */ + data->dai.ignore_pmdown_time = 1; + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { From 6ba59c008f08e84b3c87be10f3391c9735e4f833 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 14 Sep 2023 16:25:04 +0300 Subject: [PATCH 25/79] ASoC: SOF: ipc4-topology: fix wrong sizeof argument MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit available_fmt is a pointer. Fixes: 4fdef47a44d6 ("ASoC: SOF: ipc4-topology: Add new tokens for input/output pin format count") Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230914132504.18463-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f2a30cd31378..7cb63e6b24dc 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -231,7 +231,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp, ret = sof_update_ipc_object(scomp, available_fmt, SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(available_fmt), 1); + swidget->num_tuples, sizeof(*available_fmt), 1); if (ret) { dev_err(scomp->dev, "Failed to parse audio format token count\n"); return ret; From bb0216d4db9ecaa51af45d8504757becbe5c050d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Sep 2023 15:47:25 +0300 Subject: [PATCH 26/79] ASoC: SOF: sof-audio: Fix DSP core put imbalance on widget setup failure In case the widget setup fails we should only decrement the core usage count if the sof_widget_free_unlocked() has not been called as part of the error handling. sof_widget_free_unlocked() calls snd_sof_dsp_core_put() and the additional core_put will cause imbalance in core usage count. Use the existing use_count_decremented to handle this issue. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20230914124725.17397-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index e7ef77012c35..e5405f854a91 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -212,7 +212,8 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, sof_widget_free_unlocked(sdev, swidget); use_count_decremented = true; core_put: - snd_sof_dsp_core_put(sdev, swidget->core); + if (!use_count_decremented) + snd_sof_dsp_core_put(sdev, swidget->core); pipe_widget_free: if (swidget->id != snd_soc_dapm_scheduler) sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget); From c04efbfd76d23157e64e6d6147518c187ab4233a Mon Sep 17 00:00:00 2001 From: Chen Ni Date: Fri, 15 Sep 2023 02:13:44 +0000 Subject: [PATCH 27/79] ASoC: hdaudio.c: Add missing check for devm_kstrdup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Because of the potential failure of the devm_kstrdup(), the dl[i].codecs->name could be NULL. Therefore, we need to check it and return -ENOMEM in order to transfer the error. Fixes: 97030a43371e ("ASoC: Intel: avs: Add HDAudio machine board") Signed-off-by: Chen Ni Reviewed-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230915021344.3078-1-nichen@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/hdaudio.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index cb00bc86ac94..8876558f19a1 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -55,6 +55,9 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int return -ENOMEM; dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + if (!dl[i].codecs->name) + return -ENOMEM; + dl[i].codecs->dai_name = pcm->name; dl[i].num_codecs = 1; dl[i].num_cpus = 1; From b19a5733de255cabba5feecabf6e900638b582d1 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 15 Sep 2023 14:02:11 +0800 Subject: [PATCH 28/79] ASoC: imx-audmix: Fix return error with devm_clk_get() The devm_clk_get() can return -EPROBE_DEFER error, modify the error code to be -EINVAL is not correct, which cause the -EPROBE_DEFER error is not correctly handled. This patch is to fix the return error code. Fixes: b86ef5367761 ("ASoC: fsl: Add Audio Mixer machine driver") Signed-off-by: Shengjiu Wang Reviewed-by: Daniel Baluta Link: https://lore.kernel.org/r/1694757731-18308-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 0b58df56f4da..aeb81aa61184 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -315,7 +315,7 @@ static int imx_audmix_probe(struct platform_device *pdev) if (IS_ERR(priv->cpu_mclk)) { ret = PTR_ERR(priv->cpu_mclk); dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); - return -EINVAL; + return ret; } priv->audmix_pdev = audmix_pdev; From c923e7759a29cf67aa4dda77b816263771380f86 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 15 Sep 2023 15:43:00 +0100 Subject: [PATCH 29/79] ASoC: cs42l43: Add shared IRQ flag for shutters The microphone and speaker shutters on cs42l43 can be configured to trigger from the same GPIO, in this case the current code returns an error as we attempt to request two IRQ handlers for the same IRQ. Fix this by always requesting the shutter IRQs with the IRQF_SHARED flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20230915144300.120100-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 1a95c370fc4c..5643c666d7d0 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2077,7 +2077,8 @@ static const struct cs42l43_irq cs42l43_irqs[] = { static int cs42l43_request_irq(struct cs42l43_codec *priv, struct irq_domain *dom, const char * const name, - unsigned int irq, irq_handler_t handler) + unsigned int irq, irq_handler_t handler, + unsigned long flags) { int ret; @@ -2087,8 +2088,8 @@ static int cs42l43_request_irq(struct cs42l43_codec *priv, dev_dbg(priv->dev, "Request IRQ %d for %s\n", ret, name); - ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, IRQF_ONESHOT, - name, priv); + ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, + IRQF_ONESHOT | flags, name, priv); if (ret) return dev_err_probe(priv->dev, ret, "Failed to request IRQ %s\n", name); @@ -2124,11 +2125,11 @@ static int cs42l43_shutter_irq(struct cs42l43_codec *priv, return 0; } - ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler); + ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler, IRQF_SHARED); if (ret) return ret; - return cs42l43_request_irq(priv, dom, open_name, open_irq, handler); + return cs42l43_request_irq(priv, dom, open_name, open_irq, handler, IRQF_SHARED); } static int cs42l43_codec_probe(struct platform_device *pdev) @@ -2178,7 +2179,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev) for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) { ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name, - cs42l43_irqs[i].irq, cs42l43_irqs[i].handler); + cs42l43_irqs[i].irq, + cs42l43_irqs[i].handler, 0); if (ret) goto err_pm; } From e0f96246c4402514acda040be19ee24c1619e01a Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 15 Sep 2023 16:41:53 +0300 Subject: [PATCH 30/79] ASoC: SOF: Intel: MTL: Reduce the DSP init timeout MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 20s seems unnecessarily large for the DSP init timeout. This coupled with multiple FW boot attempts causes an excessive delay in the error path when booting in recovery mode. Reduce it to 0.5s and use the existing HDA_DSP_INIT_TIMEOUT_US. Link: https://github.com/thesofproject/linux/issues/4565 Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230915134153.9688-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 2 +- sound/soc/sof/intel/mtl.h | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index b84ca58da9d5..f9412517eaf2 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -460,7 +460,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) /* step 3: wait for IPC DONE bit from ROM */ ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->ipc_ack, status, ((status & chip->ipc_ack_mask) == chip->ipc_ack_mask), - HDA_DSP_REG_POLL_INTERVAL_US, MTL_DSP_PURGE_TIMEOUT_US); + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_INIT_TIMEOUT_US); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, "timeout waiting for purge IPC done\n"); diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 02181490f12a..95696b3d7c4c 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -62,7 +62,6 @@ #define MTL_DSP_IRQSTS_IPC BIT(0) #define MTL_DSP_IRQSTS_SDW BIT(6) -#define MTL_DSP_PURGE_TIMEOUT_US 20000000 /* 20s */ #define MTL_DSP_REG_POLL_INTERVAL_US 10 /* 10 us */ /* Memory windows */ From 31bb7bd9ffee50d09ec931998b823a86132ab807 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 15 Sep 2023 15:40:15 +0300 Subject: [PATCH 31/79] ASoC: SOF: core: Only call sof_ops_free() on remove if the probe was successful All the fail paths during probe will free up the ops, on remove we should only free it if the probe was successful. Fixes: bc433fd76fae ("ASoC: SOF: Add ops_free") Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Rander Wang Link: https://lore.kernel.org/r/20230915124015.19637-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 30db685cc5f4..2d1616b81485 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -486,10 +486,9 @@ int snd_sof_device_remove(struct device *dev) snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_remove(sdev); + sof_ops_free(sdev); } - sof_ops_free(sdev); - /* release firmware */ snd_sof_fw_unload(sdev); From aadb0330cfb60829318ef02ccfb9dd09cd14d920 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 19 Sep 2023 10:12:05 +0300 Subject: [PATCH 32/79] ALSA: usb-audio: scarlett_gen2: Fix another -Wformat-truncation warning The recent enablement of -Wformat-truncation leads to a false-positive warning for mixer_scarlett_gen2.c. For suppressing the warning, replace snprintf() with scnprintf(). As stated in the above, truncation doesn't matter. Fixes: 78bd8f5126f8 ("ALSA: usb-audio: scarlett_gen2: Fix -Wformat-truncation warning") Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230919071205.10684-1-peter.ujfalusi@linux.intel.com Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett_gen2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 5c6f50f38840..d260be8cb6bc 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -3205,8 +3205,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input phantom controls */ if (info->inputs_per_phantom == 1) { for (i = 0; i < info->phantom_count; i++) { - snprintf(s, sizeof(s), fmt, i + 1, - "Phantom Power", "Switch"); + scnprintf(s, sizeof(s), fmt, i + 1, + "Phantom Power", "Switch"); err = scarlett2_add_new_ctl( mixer, &scarlett2_phantom_ctl, i, 1, s, &private->phantom_ctls[i]); From deff8486a40e75813f2841f533c7572489981bae Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 19 Sep 2023 09:11:53 +0100 Subject: [PATCH 33/79] ALSA: hda: cs35l56: Use the new RUNTIME_PM_OPS() macro Use RUNTIME_PM_OPS() instead of the old SET_RUNTIME_PM_OPS(). This means we don't need __maybe_unused on the functions. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20230919081153.19793-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 87ffe8fbff99..7adc1d373d65 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -105,7 +105,7 @@ static void cs35l56_hda_playback_hook(struct device *dev, int action) } } -static int __maybe_unused cs35l56_hda_runtime_suspend(struct device *dev) +static int cs35l56_hda_runtime_suspend(struct device *dev) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); @@ -115,7 +115,7 @@ static int __maybe_unused cs35l56_hda_runtime_suspend(struct device *dev) return cs35l56_runtime_suspend_common(&cs35l56->base); } -static int __maybe_unused cs35l56_hda_runtime_resume(struct device *dev) +static int cs35l56_hda_runtime_resume(struct device *dev) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); int ret; @@ -1015,7 +1015,7 @@ void cs35l56_hda_remove(struct device *dev) EXPORT_SYMBOL_NS_GPL(cs35l56_hda_remove, SND_HDA_SCODEC_CS35L56); const struct dev_pm_ops cs35l56_hda_pm_ops = { - SET_RUNTIME_PM_OPS(cs35l56_hda_runtime_suspend, cs35l56_hda_runtime_resume, NULL) + RUNTIME_PM_OPS(cs35l56_hda_runtime_suspend, cs35l56_hda_runtime_resume, NULL) SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend, cs35l56_hda_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_late, cs35l56_hda_system_resume_early) From 381ddcd5875e496f2eae06bb65853271b7150fee Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Tue, 19 Sep 2023 17:11:35 +0800 Subject: [PATCH 34/79] ASoC: Intel: soc-acpi: Add entry for sof_es8336 in MTL match table. Adding support for ES83x6 codec in MTL match table. Signed-off-by: Balamurugan C Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20230919091136.1922253-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 0304246d2922..9cec1a4a6cd8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -30,6 +30,11 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-mtl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); From d1f67278d4b2de3bf544ea9bcd9f64d03584df87 Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Tue, 19 Sep 2023 17:11:36 +0800 Subject: [PATCH 35/79] ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in MTL match table Adding HDMI-In capture via I2S feature support in MTL platform. Signed-off-by: Balamurugan C Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20230919091136.1922253-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 10 ++++++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 12 ++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index f8a3e8a91761..9904a9e33ccc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -808,6 +808,16 @@ static const struct platform_device_id board_ids[] = { SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | SOF_ES8336_JD_INVERTED), }, + { + .name = "mtl_es83x6_c1_h02", + .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | + SOF_NO_OF_HDMI_CAPTURE_SSP(2) | + SOF_HDMI_CAPTURE_1_SSP(0) | + SOF_HDMI_CAPTURE_2_SSP(2) | + SOF_SSP_HDMI_CAPTURE_PRESENT | + SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | + SOF_ES8336_JD_INVERTED), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 9cec1a4a6cd8..92498d1d6c8d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -35,6 +35,11 @@ static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, }; +static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = { + .num_codecs = 1, + .codecs = {"INTC10B0"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -57,6 +62,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "mtl_es83x6_c1_h02", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mtl_lt6911_hdmi, + .sof_tplg_filename = "sof-mtl-es83x6-ssp1-hdmi-ssp02.tplg", + }, { .comp_ids = &mtl_essx_83x6, .drv_name = "sof-essx8336", From b399f9706a1cbae42731cc420a46cfb9c3c6b10f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 19 Sep 2023 16:36:06 +0800 Subject: [PATCH 36/79] ASoC: Intel: soc-acpi: fix Dell SKU 0B34 The rule for the SoundWire tables is that the platforms with more devices need to be added first. We broke that rule with the Dell SKU 0B34, and caused the second amplifier for SKU 0AF3 to be ignored. The fix is simple, we need to move the single-amplifier entry after the two-amplifier one. Fixes: b62a1a839b48 ("ASoC: Intel: soc-acpi: add tables for Dell SKU 0B34") Closes: https://github.com/thesofproject/linux/issues/4559 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Chao Song Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20230919083606.1920202-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 8e995edf4c10..5103e75ac830 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -655,18 +655,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l3.tplg", }, - { - .link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */ - .links = adl_sdw_rt1316_link1_rt714_link0, - .drv_name = "sof_sdw", - .sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg", - }, { .link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */ .links = adl_sdw_rt1316_link12_rt714_link0, .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt1316-l12-rt714-l0.tplg", }, + { + .link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */ + .links = adl_sdw_rt1316_link1_rt714_link0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg", + }, { .link_mask = 0x5, /* 2 active links required */ .links = adl_sdw_rt1316_link2_rt714_link0, From fb0b8d299781be8d46b3612aa96cef28da0d93f4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 19 Sep 2023 17:21:25 +0800 Subject: [PATCH 37/79] ASoC: Intel: sof_sdw: add support for SKU 0B14 One more missing SKU in the list. Closes: https://github.com/thesofproject/linux/issues/4543 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Chao Song Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20230919092125.1922468-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5a1c750e6ae6..842649501e30 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -376,6 +376,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, + { .callback = sof_sdw_quirk_cb, .matches = { From 69cf63b6560205a390a736b88d112374655adb28 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Sep 2023 01:22:57 +0000 Subject: [PATCH 38/79] ASoC: simple-card-utils: fixup simple_util_startup() error handling It should use "goto" instead of "return" Fixes: 5ca2ab459817 ("ASoC: simple-card-utils: Add new system-clock-fixed flag") Reported-by: kernel test robot Reported-by: Dan Carpenter Closes: https://lore.kernel.org/all/202309141205.ITZeDJxV-lkp@intel.com/ Closes: https://lore.kernel.org/all/202309151840.au9Aa2W4-lkp@intel.com/ Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87v8c76jnz.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 5b18a4af022f..2588ec735dbd 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -310,7 +310,8 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) if (fixed_sysclk % props->mclk_fs) { dev_err(rtd->dev, "fixed sysclk %u not divisible by mclk_fs %u\n", fixed_sysclk, props->mclk_fs); - return -EINVAL; + ret = -EINVAL; + goto codec_err; } ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, fixed_rate, fixed_rate); From 41bae58df411f9accf01ea660730649b2fab1dab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Sep 2023 05:34:18 +0000 Subject: [PATCH 39/79] ASoC: simple-card: fixup asoc_simple_probe() error handling asoc_simple_probe() is used for both "DT probe" (A) and "platform probe" (B). It uses "goto err" when error case, but it is not needed for "platform probe" case (B). Thus it is using "return" directly there. static int asoc_simple_probe(...) { ^ if (...) { | ... (A) if (ret < 0) | goto err; v } else { ^ ... | if (ret < 0) (B) return -Exxx; v } ... ^ if (ret < 0) (C) goto err; v ... err: (D) simple_util_clean_reference(card); return ret; } Both case are using (C) part, and it calls (D) when err case. But (D) will do nothing for (B) case. Because of these behavior, current code itself is not wrong, but is confusable, and more, static analyzing tool will warning on (B) part (should use goto err). To avoid static analyzing tool warning, this patch uses "goto err" on (B) part. Reported-by: kernel test robot Reported-by: Dan Carpenter Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87o7hy7mlh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 190f11366e84..274417e39e7d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -759,10 +759,12 @@ static int asoc_simple_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link = priv->dai_link; struct simple_dai_props *dai_props = priv->dai_props; + ret = -EINVAL; + cinfo = dev->platform_data; if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + goto err; } if (!cinfo->name || @@ -771,7 +773,7 @@ static int asoc_simple_probe(struct platform_device *pdev) !cinfo->platform || !cinfo->cpu_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + goto err; } cpus = dai_link->cpus; From 41b07476da38ac2878a14e5b8fe0312c41ea36e3 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 19 Sep 2023 16:27:16 +0800 Subject: [PATCH 40/79] ALSA: hda/realtek - ALC287 Realtek I2S speaker platform support New platform SSID:0x231f. 0x17 was only speaker pin, DAC assigned will be 0x03. Headphone assigned to 0x02. Playback via headphone will get EQ filter processing. So, it needs to swap DAC. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/8d63c6e360124e3ea2523753050e6f05@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 883a7e865bc5..751783f3a15c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10577,6 +10577,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x90170110}, {0x19, 0x03a11030}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC287_FIXUP_THINKPAD_I2S_SPK, + {0x17, 0x90170110}, /* 0x231f with RTK I2S AMP */ + {0x19, 0x04a11040}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, From d93eeca627db512a56145285dc94feac5b88a1d4 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 21 Sep 2023 15:20:41 +0800 Subject: [PATCH 41/79] ALSA: hda/realtek - ALC287 merge RTK codec with CS CS35L41 AMP This is merge model ALC287_FIXUP_THINKPAD_I2S_SPK and ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI. Signed-off-by: Kailang Yang Fixes: f7b069cf0881 ("ALSA: hda/realtek: Fix generic fixup definition for cs35l41 amp") Link: https://lore.kernel.org/r/82a45234327c4c50b4988a27e9f64c37@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++-------- 1 file changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 751783f3a15c..48bb57bd8d11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7343,6 +7343,7 @@ enum { ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, ALC287_FIXUP_THINKPAD_I2S_SPK, + ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9441,6 +9442,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc287_fixup_bind_dacs, }, + [ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_bind_dacs, + .chained = true, + .chain_id = ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9988,14 +9995,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22c2, "Thinkpad X1 Extreme Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), - SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), - SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), From e52dca7216cfeae76a99908a2eea6e850d3f918f Mon Sep 17 00:00:00 2001 From: Miquel Raynal Date: Fri, 22 Sep 2023 18:15:47 +0200 Subject: [PATCH 42/79] ASoC: soc-generic-dmaengine-pcm: Fix function name in comment While browsing/grepping in the sound core, I found that snd_dmaengine_set_config_from_dai_data() did not exist, in favor of snd_dmaengine_pcm_set_config_from_dai_data(). Fix the typo. Signed-off-by: Miquel Raynal Link: https://lore.kernel.org/r/20230922161547.594484-1-miquel.raynal@bootlin.com Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d0653d775c87..cad222eb9a29 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -44,8 +44,8 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, * platforms which make use of the snd_dmaengine_dai_dma_data struct for their * DAI DMA data. Internally the function will first call * snd_hwparams_to_dma_slave_config to fill in the slave config based on the - * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the - * remaining fields based on the DAI DMA data. + * hw_params, followed by snd_dmaengine_pcm_set_config_from_dai_data to fill in + * the remaining fields based on the DAI DMA data. */ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) From 5c8a033f5674ae62d5aa2ebbdb9980b89380c34f Mon Sep 17 00:00:00 2001 From: Alex Bee Date: Tue, 29 Aug 2023 19:16:19 +0200 Subject: [PATCH 43/79] dt-bindings: ASoC: rockchip: Add compatible for RK3128 spdif Add compatible for RK3128's S/PDIF. Signed-off-by: Alex Bee Acked-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20230829171647.187787-4-knaerzche@gmail.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index 4f51b2fa82db..c3c989ef2a2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -26,6 +26,7 @@ properties: - const: rockchip,rk3568-spdif - items: - enum: + - rockchip,rk3128-spdif - rockchip,rk3188-spdif - rockchip,rk3288-spdif - rockchip,rk3308-spdif From 197c53c8ecb34f2cd5922f4bdcffa8f701a134eb Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 19 Sep 2023 17:42:13 +0800 Subject: [PATCH 44/79] ASoC: fsl_sai: Don't disable bitclock for i.MX8MP On i.MX8MP, the BCE and TERE bit are binding with mclk enablement, if BCE and TERE are cleared the MCLK also be disabled on output pin, that cause the external codec (wm8960) in wrong state. Codec (wm8960) is using the mclk to generate PLL clock, if mclk is disabled before disabling PLL, the codec (wm8960) won't generate bclk and frameclk when sysclk switch to MCLK source in next test case. The test case: $aplay -r44100 test1.wav (PLL source) $aplay -r48000 test2.wav (MCLK source) aplay: pcm_write:2127: write error: Input/output error Fixes: 269f399dc19f ("ASoC: fsl_sai: Disable bit clock with transmitter") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1695116533-23287-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1e4020fae05a..8a9a30dd31e2 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -710,10 +710,15 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) { unsigned int ofs = sai->soc_data->reg_offset; bool tx = dir == TX; - u32 xcsr, count = 100; + u32 xcsr, count = 100, mask; + + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output) + mask = FSL_SAI_CSR_TERE; + else + mask = FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE; regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), - FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0); + mask, 0); /* TERE will remain set till the end of current frame */ do { From 2b21207afd06714986a3d22442ed4860ba4f9ced Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 20 Sep 2023 17:43:12 +0800 Subject: [PATCH 45/79] ASoC: fsl-asoc-card: use integer type for fll_id and pll_id As the pll_id and pll_id can be zero (WM8960_SYSCLK_AUTO) with the commit 2bbc2df46e67 ("ASoC: wm8960: Make automatic the default clocking mode") Then the machine driver will skip to call set_sysclk() and set_pll() for codec, when the sysclk rate is different with what wm8960 read at probe, the output sound frequency is wrong. So change the fll_id and pll_id initial value, still keep machine driver's behavior same as before. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1695202992-24864-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 76b5bfc288fd..bab7d34cf585 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -52,8 +52,8 @@ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; - u32 fll_id; - u32 pll_id; + int fll_id; + int pll_id; }; /** @@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } /* Specific configuration for PLL */ - if (codec_priv->pll_id && codec_priv->fll_id) { + if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * 384; else @@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) priv->streams &= ~BIT(substream->stream); - if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), codec_priv->mclk_id, @@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; + + priv->codec_priv.fll_id = -1; + priv->codec_priv.pll_id = -1; + /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; From 7e1fe5d9e7eae67e218f878195d1d348d01f9af7 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 27 Sep 2023 12:44:10 +0530 Subject: [PATCH 46/79] ASoC: SOF: amd: fix for firmware reload failure after playback Setting ACP ACLK as clock source when ACP enters D0 state causing firmware load failure as mentioned in below scenario. - Load snd_sof_amd_rembrandt - Play or Record audio - Stop audio - Unload snd_sof_amd_rembrandt - Reload snd_sof_amd_rembrandt If acp_clkmux_sel register field is set, then clock source will be set to ACP ACLK when ACP enters D0 state. During stream stop, if there is no active stream is running then acp firmware will set the ACP ACLK value to zero. When driver is reloaded and clock source is selected as ACP ACLK, as ACP ACLK is programmed to zero, firmware loading will fail. For RMB platform, remove the clock mux selection field so that ACP will use internal clock source when ACP enters D0 state. Fixes: 41cb85bc4b52 ("ASoC: SOF: amd: Add support for Rembrandt plaform.") Reported-by: coolstar Closes: https://github.com/thesofproject/sof/issues/8137 Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20230927071412.2416250-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/pci-rmb.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 9935e457b467..a7ae76efc2dd 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -35,7 +35,6 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = { .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, - .acp_clkmux_sel = ACP6X_CLKMUX_SEL, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, .probe_reg_offset = ACP6X_FUTURE_REG_ACLK_0, }; From e80f238d2bc0c0f27dc52ac824ca80b938a43ace Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 29 Sep 2023 12:32:42 +0200 Subject: [PATCH 47/79] ASoC: core: Print component name when printing log MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When printing log related to component it is useful to know, to which component it applies to. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230929103243.705433-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cc442c52cdea..33eb5e2bb8bc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1445,8 +1445,8 @@ static int soc_probe_component(struct snd_soc_card *card, if (component->card) { if (component->card != card) { dev_err(component->dev, - "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", - card->name, component->card->name); + "Trying to bind component \"%s\" to card \"%s\" but is already bound to card \"%s\"\n", + component->name, card->name, component->card->name); return -ENODEV; } return 0; From dd9f9cc1e6b9391140afa5cf27bb47c9e2a08d02 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 29 Sep 2023 12:32:43 +0200 Subject: [PATCH 48/79] ASoC: core: Do not call link_exit() on uninitialized rtd objects MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On init we have sequence: for_each_card_prelinks(card, i, dai_link) { ret = snd_soc_add_pcm_runtime(card, dai_link); ret = init_some_other_things(...); if (ret) goto probe_end: for_each_card_rtds(card, rtd) { ret = soc_init_pcm_runtime(card, rtd); probe_end: while on exit: for_each_card_rtds(card, rtd) snd_soc_link_exit(rtd); If init_some_other_things() step fails due to error we end up with not fully setup rtds and try to call snd_soc_link_exit on them, which depending on contents on .link_exit handler, can end up dereferencing NULL pointer. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230929103243.705433-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 20 +++++++++++++++----- 2 files changed, 17 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index fa2337a3cf4c..37f9d3fe302a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1126,6 +1126,8 @@ struct snd_soc_pcm_runtime { unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ + bool initialized; + int num_components; struct snd_soc_component *components[]; /* CPU/Codec/Platform */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 33eb5e2bb8bc..9de98c01d815 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1347,7 +1347,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, snd_soc_runtime_get_dai_fmt(rtd); ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); if (ret) - return ret; + goto err; /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); @@ -1372,17 +1372,26 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* create compress_device if possible */ ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); if (ret != -ENOTSUPP) - return ret; + goto err; /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); - return ret; + goto err; } - return snd_soc_pcm_dai_new(rtd); + ret = snd_soc_pcm_dai_new(rtd); + if (ret < 0) + goto err; + + rtd->initialized = true; + + return 0; +err: + snd_soc_link_exit(rtd); + return ret; } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1980,7 +1989,8 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) /* release machine specific resources */ for_each_card_rtds(card, rtd) - snd_soc_link_exit(rtd); + if (rtd->initialized) + snd_soc_link_exit(rtd); /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); From b84b53149476b22cc3b8677b771fb4cf06d1d455 Mon Sep 17 00:00:00 2001 From: Matthias Reichl Date: Fri, 29 Sep 2023 21:50:28 +0200 Subject: [PATCH 49/79] ASoC: hdmi-codec: Fix broken channel map reporting Commit 4e0871333661 ("ASoC: hdmi-codec: fix channel info for compressed formats") accidentally changed hcp->chmap_idx from ca_id, the CEA channel allocation ID, to idx, the index to the table of channel mappings ordered by preference. This resulted in wrong channel maps being reported to userspace, eg for 5.1 "FL,FR,LFE,FC" was reported instead of the expected "FL,FR,LFE,FC,RL,RR": ~ # speaker-test -c 6 -t sine ... 0 - Front Left 3 - Front Center 1 - Front Right 2 - LFE 4 - Unknown 5 - Unknown ~ # amixer cget iface=PCM,name='Playback Channel Map' | grep ': values' : values=3,4,8,7,0,0,0,0 Switch this back to ca_id in case of PCM audio so the correct channel map is reported again and set it to HDMI_CODEC_CHMAP_IDX_UNKNOWN in case of non-PCM audio so the PCM channel map control returns "Unknown" channels (value 0). Fixes: 4e0871333661 ("ASoC: hdmi-codec: fix channel info for compressed formats") Cc: stable@vger.kernel.org Signed-off-by: Matthias Reichl Link: https://lore.kernel.org/r/20230929195027.97136-1-hias@horus.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 13689e718d36..09eef6042aad 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -531,7 +531,10 @@ static int hdmi_codec_fill_codec_params(struct snd_soc_dai *dai, hp->sample_rate = sample_rate; hp->channels = channels; - hcp->chmap_idx = idx; + if (pcm_audio) + hcp->chmap_idx = ca_id; + else + hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; return 0; } From 892fbdb203945d887ad2a109a3700b091a8e3b97 Mon Sep 17 00:00:00 2001 From: Zhang Shurong Date: Sat, 30 Sep 2023 17:55:50 +0800 Subject: [PATCH 50/79] ASoC: rt5682: Fix regulator enable/disable sequence This will attempt to disable the regulators if the initial enable fails which is a bug. Fix this bug by modifying the code to the correct sequence. Signed-off-by: Zhang Shurong Link: https://lore.kernel.org/r/tencent_4F37C9B5315B7960041E8E0ADDA869128F08@qq.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-i2c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index b05b4f73d8aa..fbad1ed06626 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -157,11 +157,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators, - rt5682); - if (ret) - return ret; - ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), rt5682->supplies); if (ret) { @@ -169,6 +164,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5682_i2c_disable_regulators, + rt5682); + if (ret) + return ret; + ret = rt5682_get_ldo1(rt5682, &i2c->dev); if (ret) return ret; From e930bea4124b8a4a47ba4092d99da30099b9242d Mon Sep 17 00:00:00 2001 From: Antoine Gennart Date: Fri, 29 Sep 2023 15:01:17 +0200 Subject: [PATCH 51/79] ASoC: tlv320adc3xxx: BUG: Correct micbias setting The micbias setting for tlv320adc can also have the value '3' which means that the micbias ouput pin is connected to the input pin AVDD. Signed-off-by: Antoine Gennart Link: https://lore.kernel.org/r/20230929130117.77661-1-gennartan@disroot.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adc3xxx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index b976c1946286..420bbf588efe 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -293,7 +293,7 @@ #define ADC3XXX_BYPASS_RPGA 0x80 /* MICBIAS control bits */ -#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS_MASK 0x3 #define ADC3XXX_MICBIAS1_SHIFT 5 #define ADC3XXX_MICBIAS2_SHIFT 3 @@ -1099,7 +1099,7 @@ static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, unsigned int val; if (!of_property_read_u32(np, propname, &val)) { - if (val >= ADC3XXX_MICBIAS_AVDD) { + if (val > ADC3XXX_MICBIAS_AVDD) { dev_err(dev, "Invalid property value for '%s'\n", propname); return -EINVAL; } From 1948fa64727685ac3f6584755212e2e738b6b051 Mon Sep 17 00:00:00 2001 From: Sven Frotscher Date: Thu, 28 Sep 2023 00:36:07 +0200 Subject: [PATCH 52/79] ASoC: amd: yc: Fix non-functional mic on Lenovo 82YM Like the Lenovo 82TL, 82V2, 82QF and 82UG, the 82YM (Yoga 7 14ARP8) requires an entry in the quirk list to enable the internal microphone. The latter two received similar fixes in commit 1263cc0f414d ("ASoC: amd: yc: Fix non-functional mic on Lenovo 82QF and 82UG"). Fixes: c008323fe361 ("ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ") Cc: stable@vger.kernel.org Signed-off-by: Sven Frotscher Link: https://lore.kernel.org/r/20230927223758.18870-1-sven.frotscher@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 94e9eb8e73f2..15a864dcd7bd 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -241,6 +241,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "82V2"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82YM"), + } + }, { .driver_data = &acp6x_card, .matches = { From 1426b9ba7c453755d182ebf7e7f2367ba249dcf4 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 Oct 2023 09:29:35 -0300 Subject: [PATCH 53/79] ASoC: dt-bindings: fsl,micfil: Document #sound-dai-cells imx8mp.dtsi passes #sound-dai-cells = <0> in the micfil node. Document #sound-dai-cells to fix the following schema warning: audio-controller@30ca0000: '#sound-dai-cells' does not match any of the regexes: 'pinctrl-[0-9]+' from schema $id: http://devicetree.org/schemas/sound/fsl,micfil.yaml# Signed-off-by: Fabio Estevam Reviewed-by: Adam Ford Link: https://lore.kernel.org/r/20231004122935.2250889-1-festevam@gmail.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl,micfil.yaml | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml index 4b99a18c79a0..b7e605835639 100644 --- a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml @@ -56,6 +56,9 @@ properties: - const: clkext3 minItems: 2 + "#sound-dai-cells": + const: 0 + required: - compatible - reg From 5d542b850d40cb08a38ad4bb2a944dbf1b7b0683 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 3 Oct 2023 15:21:38 +0100 Subject: [PATCH 54/79] ALSA: hda: cs35l41: Cleanup and fix double free in firmware request There is an unlikely but possible double free when loading firmware, and a missing free calls if a firmware is successfully requested but the coefficient file request fails, leading to the fallback firmware request occurring without clearing the previously loaded firmware. Fixes: cd40dad2ca91 ("ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded") Reported-by: kernel test robot Reported-by: Dan Carpenter Closes: https://lore.kernel.org/r/202309291331.0JUUQnPT-lkp@intel.com/ Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231003142138.180108-1-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 115 +++++++++++++++++++++++++----------- 1 file changed, 79 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index f9b77353c266..c6031f744099 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -185,10 +185,14 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, cs35l41->speaker_id, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ - return cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, - cs35l41->acpi_subsystem_id, cs35l41->amp_name, - cs35l41->speaker_id, "bin"); + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, cs35l41->amp_name, + cs35l41->speaker_id, "bin"); + if (ret) + goto coeff_err; + + return 0; } /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ @@ -197,10 +201,14 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, cs35l41->amp_name, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ - return cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, - cs35l41->acpi_subsystem_id, cs35l41->amp_name, - cs35l41->speaker_id, "bin"); + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, cs35l41->amp_name, + cs35l41->speaker_id, "bin"); + if (ret) + goto coeff_err; + + return 0; } /* try cirrus/part-dspN-fwtype-sub<-spkidN>.wmfw */ @@ -215,10 +223,14 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, cs35l41->amp_name, cs35l41->speaker_id, "bin"); if (ret) /* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */ - return cs35l41_request_firmware_file(cs35l41, coeff_firmware, - coeff_filename, CS35L41_FIRMWARE_ROOT, - cs35l41->acpi_subsystem_id, NULL, - cs35l41->speaker_id, "bin"); + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, + coeff_filename, CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, NULL, + cs35l41->speaker_id, "bin"); + if (ret) + goto coeff_err; + + return 0; } /* try cirrus/part-dspN-fwtype-sub.wmfw */ @@ -233,12 +245,50 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, cs35l41->speaker_id, "bin"); if (ret) /* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */ - return cs35l41_request_firmware_file(cs35l41, coeff_firmware, - coeff_filename, CS35L41_FIRMWARE_ROOT, - cs35l41->acpi_subsystem_id, NULL, - cs35l41->speaker_id, "bin"); + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, + coeff_filename, CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, NULL, + cs35l41->speaker_id, "bin"); + if (ret) + goto coeff_err; } + return ret; +coeff_err: + release_firmware(*wmfw_firmware); + kfree(*wmfw_filename); + return ret; +} + +static int cs35l41_fallback_firmware_file(struct cs35l41_hda *cs35l41, + const struct firmware **wmfw_firmware, + char **wmfw_filename, + const struct firmware **coeff_firmware, + char **coeff_filename) +{ + int ret; + + /* Handle fallback */ + dev_warn(cs35l41->dev, "Falling back to default firmware.\n"); + + /* fallback try cirrus/part-dspN-fwtype.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw"); + if (ret) + goto err; + + /* fallback try cirrus/part-dspN-fwtype.bin */ + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin"); + if (ret) { + release_firmware(*wmfw_firmware); + kfree(*wmfw_filename); + goto err; + } + return 0; + +err: + dev_warn(cs35l41->dev, "Unable to find firmware and tuning\n"); return ret; } @@ -254,7 +304,6 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, ret = cs35l41_request_firmware_files_spkid(cs35l41, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename); goto out; - } /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ @@ -267,6 +316,9 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, -1, "bin"); + if (ret) + goto coeff_err; + goto out; } @@ -286,32 +338,23 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, NULL, -1, "bin"); + if (ret) + goto coeff_err; } out: - if (!ret) - return 0; + if (ret) + /* if all attempts at finding firmware fail, try fallback */ + goto fallback; - /* Handle fallback */ - dev_warn(cs35l41->dev, "Falling back to default firmware.\n"); + return 0; +coeff_err: release_firmware(*wmfw_firmware); kfree(*wmfw_filename); - - /* fallback try cirrus/part-dspN-fwtype.wmfw */ - ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw"); - if (!ret) - /* fallback try cirrus/part-dspN-fwtype.bin */ - ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin"); - - if (ret) { - release_firmware(*wmfw_firmware); - kfree(*wmfw_filename); - dev_warn(cs35l41->dev, "Unable to find firmware and tuning\n"); - } - return ret; +fallback: + return cs35l41_fallback_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + coeff_firmware, coeff_filename); } #if IS_ENABLED(CONFIG_EFI) From 6a83d6f3bb3c329a73e3483651fb77b78bac1878 Mon Sep 17 00:00:00 2001 From: WhaleChang Date: Fri, 6 Oct 2023 12:48:49 +0800 Subject: [PATCH 55/79] ALSA: usb-audio: Fix microphone sound on Opencomm2 Headset When a Opencomm2 Headset is connected to a Bluetooth USB dongle, the audio playback functions properly, but the microphone does not work. In the dmesg logs, there are messages indicating that the init_pitch function fails when the capture process begins. The microphone only functions when the ep pitch control is not set. Toggling the pitch control off bypasses the init_piatch function and allows the microphone to work. Signed-off-by: WhaleChang Link: https://lore.kernel.org/r/20231006044852.4181022-1-whalechang@google.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 598659d761cc..d4bbef70d2f7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1994,7 +1994,11 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* mic works only when ep packet size is set to wMaxPacketSize */ fp->attributes |= UAC_EP_CS_ATTR_FILL_MAX; break; - + case USB_ID(0x3511, 0x2b1e): /* Opencomm2 UC USB Bluetooth dongle */ + /* mic works only when ep pitch control is not set */ + if (stream == SNDRV_PCM_STREAM_CAPTURE) + fp->attributes &= ~UAC_EP_CS_ATTR_PITCH_CONTROL; + break; } } From ccbd88be057a38531f835e8a04948ebf80cb0c5d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 6 Oct 2023 14:47:37 +0800 Subject: [PATCH 56/79] ALSA: hda/realtek: Change model for Intel RVP board Intel RVP board (0x12cc) has Headset Mic issue for reboot. If system plugged headset when system reboot the headset Mic was gone. Fixes: 1a93f10c5b12 ("ALSA: hda/realtek: Add "Intel Reference board" and "NUC 13" SSID in the ALC256") Signed-off-by: Kailang Yang Link: https://lore.kernel.org/r/28112f54c0c6496f97ac845645bc0256@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 48bb57bd8d11..3eeecf67c17b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9861,7 +9861,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), + SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), @@ -10098,7 +10098,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), - SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC225_FIXUP_HEADSET_JACK), + 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), #if 0 From 4a63e68a295187ae3c1cb3fa0c583c96a959714f Mon Sep 17 00:00:00 2001 From: Christos Skevis Date: Fri, 6 Oct 2023 17:53:30 +0200 Subject: [PATCH 57/79] ALSA: usb-audio: Fix microphone sound on Nexigo webcam. I own an external usb Webcam, model NexiGo N930AF, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ +0.047857] usb 5-1: new high-speed USB device number 2 using xhci_hcd [ +0.003406] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17 [ +0.000007] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam [ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD [ +0.000004] usb 5-1: SerialNumber: 20201217011 [ +0.003900] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283) [ +0.025726] usb 5-1: 3:1: cannot get usb sound sample rate freq at ep 0x86 [ +0.071482] usb 5-1: 3:2: cannot get usb sound sample rate freq at ep 0x86 [ +0.004679] usb 5-1: 3:3: cannot get usb sound sample rate freq at ep 0x86 [ +0.051607] usb 5-1: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ +0.000005] usb 5-1: [7] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, after 3 failures All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : (snip) Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 8 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x005c 1x 92 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Based on the usb data about manufacturer, SPCA2281B3 is the most likely controller IC Manufacturer does not provide link for datasheet nor detailed specs. No way to confirm if the firmware supports any other way of getting the sample rate. Testing patch provides consistent good sound recording quality and volume range. (snip) [ +0.045764] usb 5-1: new high-speed USB device number 2 using xhci_hcd [ +0.106290] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17 [ +0.000006] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam [ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD [ +0.000004] usb 5-1: SerialNumber: 20201217011 [ +0.043700] usb 5-1: set resolution quirk: cval->res = 16 [ +0.002585] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283) Signed-off-by: Christos Skevis Link: https://lore.kernel.org/r/20231006155330.399393-1-xristos.thes@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 985b1aea9cdc..409fc1164694 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1204,6 +1204,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2283): /* NexiGo N930AF FHD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d4bbef70d2f7..4e64842245e1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2177,6 +2177,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ QUIRK_FLAG_FIXED_RATE), + DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ From 76aca10ccb7c23a7b7a0d56e0bfde2c8cdddfe24 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 3 Oct 2023 17:57:09 +0200 Subject: [PATCH 58/79] ASoC: soc-dapm: Add helper for comparing widget name Some drivers use one event callback for multiple widgets but still need to perform a bit different actions based on actual widget. This is done by comparing widget name, however drivers tend to miss possible name prefix. Add a helper to solve common mistakes. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231003155710.821315-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + sound/soc/soc-component.c | 1 + sound/soc/soc-dapm.c | 12 ++++++++++++ 3 files changed, 14 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d2faec9a323e..433543eb82b9 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -469,6 +469,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai); +int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ba7c0ae82e00..566033f7dd2e 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -242,6 +242,7 @@ int snd_soc_component_notify_control(struct snd_soc_component *component, 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 diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f07e83678373..312e55579831 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2728,6 +2728,18 @@ int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai); +int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm); + const char *wname = widget->name; + + if (component->name_prefix) + wname += strlen(component->name_prefix) + 1; /* plus space */ + + return strcmp(wname, s); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_widget_name_cmp); + /* * dapm_update_widget_flags() - Re-compute widget sink and source flags * @w: The widget for which to update the flags From c29e5263d32a6d0ec094d425ae7fef3fa8d4da1c Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 3 Oct 2023 17:57:10 +0200 Subject: [PATCH 59/79] ASoC: codecs: wsa-macro: handle component name prefix When comparing widget names in wsa_macro_spk_boost_event(), consider also the component's name prefix. Otherwise the WSA codec won't have proper mixer setup resulting in no sound playback through speakers. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231003155710.821315-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-wsa-macro.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index ec6859ec0d38..fff4a8b862a7 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1675,12 +1675,12 @@ static int wsa_macro_spk_boost_event(struct snd_soc_dapm_widget *w, u16 boost_path_ctl, boost_path_cfg1; u16 reg, reg_mix; - if (!strcmp(w->name, "WSA_RX INT0 CHAIN")) { + if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT0 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST0_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX0_RX_PATH_CFG1; reg = CDC_WSA_RX0_RX_PATH_CTL; reg_mix = CDC_WSA_RX0_RX_PATH_MIX_CTL; - } else if (!strcmp(w->name, "WSA_RX INT1 CHAIN")) { + } else if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT1 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST1_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX1_RX_PATH_CFG1; reg = CDC_WSA_RX1_RX_PATH_CTL; From bfbc79de60c53e5fed505390440b87ef59ee268c Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:52 +0200 Subject: [PATCH 60/79] ASoC: codecs: wcd938x: drop bogus bind error handling Drop the bogus error handling for a soundwire device backcast during bind() that cannot fail. Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-2-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index a3c680661377..cf1eaf678fc2 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3448,10 +3448,6 @@ static int wcd938x_bind(struct device *dev) wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); - if (!wcd938x->tx_sdw_dev) { - dev_err(dev, "could not get txslave with matching of dev\n"); - return -EINVAL; - } /* As TX is main CSR reg interface, which should not be suspended first. * expicilty add the dependency link */ From fa2f8a991ba4aa733ac1c3b1be0c86148aa4c52c Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:53 +0200 Subject: [PATCH 61/79] ASoC: codecs: wcd938x: fix unbind tear down order Make sure to deregister the component before tearing down the resources it depends on during unbind(). Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-3-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index cf1eaf678fc2..c617fc3ce489 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3504,10 +3504,10 @@ static void wcd938x_unbind(struct device *dev) { struct wcd938x_priv *wcd938x = dev_get_drvdata(dev); + snd_soc_unregister_component(dev); device_link_remove(dev, wcd938x->txdev); device_link_remove(dev, wcd938x->rxdev); device_link_remove(wcd938x->rxdev, wcd938x->txdev); - snd_soc_unregister_component(dev); component_unbind_all(dev, wcd938x); } From da29b94ed3547cee9d510d02eca4009f2de476cf Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:54 +0200 Subject: [PATCH 62/79] ASoC: codecs: wcd938x: fix resource leaks on bind errors Add the missing code to release resources on bind errors, including the references taken by wcd938x_sdw_device_get() which also need to be dropped on unbind(). Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-4-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 44 +++++++++++++++++++++++++++++--------- 1 file changed, 34 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index c617fc3ce489..7e0b07eeed77 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3435,7 +3435,8 @@ static int wcd938x_bind(struct device *dev) wcd938x->rxdev = wcd938x_sdw_device_get(wcd938x->rxnode); if (!wcd938x->rxdev) { dev_err(dev, "could not find slave with matching of node\n"); - return -EINVAL; + ret = -EINVAL; + goto err_unbind; } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; @@ -3443,7 +3444,8 @@ static int wcd938x_bind(struct device *dev) wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { dev_err(dev, "could not find txslave with matching of node\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put_rxdev; } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; @@ -3454,31 +3456,35 @@ static int wcd938x_bind(struct device *dev) if (!device_link_add(wcd938x->rxdev, wcd938x->txdev, DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME)) { dev_err(dev, "could not devlink tx and rx\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put_txdev; } if (!device_link_add(dev, wcd938x->txdev, DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME)) { dev_err(dev, "could not devlink wcd and tx\n"); - return -EINVAL; + ret = -EINVAL; + goto err_remove_rxtx_link; } if (!device_link_add(dev, wcd938x->rxdev, DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME)) { dev_err(dev, "could not devlink wcd and rx\n"); - return -EINVAL; + ret = -EINVAL; + goto err_remove_tx_link; } wcd938x->regmap = dev_get_regmap(&wcd938x->tx_sdw_dev->dev, NULL); if (!wcd938x->regmap) { dev_err(dev, "could not get TX device regmap\n"); - return -EINVAL; + ret = -EINVAL; + goto err_remove_rx_link; } ret = wcd938x_irq_init(wcd938x, dev); if (ret) { dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); - return ret; + goto err_remove_rx_link; } wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; @@ -3487,17 +3493,33 @@ static int wcd938x_bind(struct device *dev) ret = wcd938x_set_micbias_data(wcd938x); if (ret < 0) { dev_err(dev, "%s: bad micbias pdata\n", __func__); - return ret; + goto err_remove_rx_link; } ret = snd_soc_register_component(dev, &soc_codec_dev_wcd938x, wcd938x_dais, ARRAY_SIZE(wcd938x_dais)); - if (ret) + if (ret) { dev_err(dev, "%s: Codec registration failed\n", __func__); + goto err_remove_rx_link; + } + + return 0; + +err_remove_rx_link: + device_link_remove(dev, wcd938x->rxdev); +err_remove_tx_link: + device_link_remove(dev, wcd938x->txdev); +err_remove_rxtx_link: + device_link_remove(wcd938x->rxdev, wcd938x->txdev); +err_put_txdev: + put_device(wcd938x->txdev); +err_put_rxdev: + put_device(wcd938x->rxdev); +err_unbind: + component_unbind_all(dev, wcd938x); return ret; - } static void wcd938x_unbind(struct device *dev) @@ -3508,6 +3530,8 @@ static void wcd938x_unbind(struct device *dev) device_link_remove(dev, wcd938x->txdev); device_link_remove(dev, wcd938x->rxdev); device_link_remove(wcd938x->rxdev, wcd938x->txdev); + put_device(wcd938x->txdev); + put_device(wcd938x->rxdev); component_unbind_all(dev, wcd938x); } From 69a026a2357ee69983690d07976de44ef26ee38a Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:55 +0200 Subject: [PATCH 63/79] ASoC: codecs: wcd938x: fix regulator leaks on probe errors Make sure to disable and free the regulators on probe errors and on driver unbind. Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-5-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 7e0b07eeed77..679c627f7eaa 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3325,8 +3325,10 @@ static int wcd938x_populate_dt_data(struct wcd938x_priv *wcd938x, struct device return dev_err_probe(dev, ret, "Failed to get supplies\n"); ret = regulator_bulk_enable(WCD938X_MAX_SUPPLY, wcd938x->supplies); - if (ret) + if (ret) { + regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); return dev_err_probe(dev, ret, "Failed to enable supplies\n"); + } wcd938x_dt_parse_micbias_info(dev, wcd938x); @@ -3592,13 +3594,13 @@ static int wcd938x_probe(struct platform_device *pdev) ret = wcd938x_add_slave_components(wcd938x, dev, &match); if (ret) - return ret; + goto err_disable_regulators; wcd938x_reset(wcd938x); ret = component_master_add_with_match(dev, &wcd938x_comp_ops, match); if (ret) - return ret; + goto err_disable_regulators; pm_runtime_set_autosuspend_delay(dev, 1000); pm_runtime_use_autosuspend(dev); @@ -3608,11 +3610,21 @@ static int wcd938x_probe(struct platform_device *pdev) pm_runtime_idle(dev); return 0; + +err_disable_regulators: + regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies); + regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); + + return ret; } static void wcd938x_remove(struct platform_device *pdev) { + struct wcd938x_priv *wcd938x = dev_get_drvdata(&pdev->dev); + component_master_del(&pdev->dev, &wcd938x_comp_ops); + regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies); + regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); } #if defined(CONFIG_OF) From 3ebebb2c1eca92a15107b2d7aeff34196fd9e217 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:56 +0200 Subject: [PATCH 64/79] ASoC: codecs: wcd938x: fix runtime PM imbalance on remove Make sure to balance the runtime PM operations, including the disable count, on driver unbind. Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-6-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 679c627f7eaa..d27b919c63b4 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3620,9 +3620,15 @@ static int wcd938x_probe(struct platform_device *pdev) static void wcd938x_remove(struct platform_device *pdev) { - struct wcd938x_priv *wcd938x = dev_get_drvdata(&pdev->dev); + struct device *dev = &pdev->dev; + struct wcd938x_priv *wcd938x = dev_get_drvdata(dev); + + component_master_del(dev, &wcd938x_comp_ops); + + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); - component_master_del(&pdev->dev, &wcd938x_comp_ops); regulator_bulk_disable(WCD938X_MAX_SUPPLY, wcd938x->supplies); regulator_bulk_free(WCD938X_MAX_SUPPLY, wcd938x->supplies); } From f0dfdcbe706462495d47982eecd13a61aabd644d Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:57 +0200 Subject: [PATCH 65/79] ASoC: codecs: wcd938x-sdw: fix use after free on driver unbind Make sure to deregister the component when the driver is being unbound and before the underlying device-managed resources are freed. Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-7-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x-sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index 6951120057e5..1baea04480e2 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1281,6 +1281,15 @@ static int wcd9380_probe(struct sdw_slave *pdev, return component_add(dev, &wcd938x_sdw_component_ops); } +static int wcd9380_remove(struct sdw_slave *pdev) +{ + struct device *dev = &pdev->dev; + + component_del(dev, &wcd938x_sdw_component_ops); + + return 0; +} + static const struct sdw_device_id wcd9380_slave_id[] = { SDW_SLAVE_ENTRY(0x0217, 0x10d, 0), {}, @@ -1320,6 +1329,7 @@ static const struct dev_pm_ops wcd938x_sdw_pm_ops = { static struct sdw_driver wcd9380_codec_driver = { .probe = wcd9380_probe, + .remove = wcd9380_remove, .ops = &wcd9380_slave_ops, .id_table = wcd9380_slave_id, .driver = { From c5c0383082eace13da2ffceeea154db2780165e7 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Tue, 3 Oct 2023 17:55:58 +0200 Subject: [PATCH 66/79] ASoC: codecs: wcd938x-sdw: fix runtime PM imbalance on probe errors Make sure to balance the runtime PM operations, including the disable count, on probe errors and on driver unbind. Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: stable@vger.kernel.org # 5.14 Cc: Srinivas Kandagatla Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231003155558.27079-8-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x-sdw.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index 1baea04480e2..a1f04010da95 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1278,7 +1278,18 @@ static int wcd9380_probe(struct sdw_slave *pdev, pm_runtime_set_active(dev); pm_runtime_enable(dev); - return component_add(dev, &wcd938x_sdw_component_ops); + ret = component_add(dev, &wcd938x_sdw_component_ops); + if (ret) + goto err_disable_rpm; + + return 0; + +err_disable_rpm: + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); + + return ret; } static int wcd9380_remove(struct sdw_slave *pdev) @@ -1287,6 +1298,10 @@ static int wcd9380_remove(struct sdw_slave *pdev) component_del(dev, &wcd938x_sdw_component_ops); + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); + return 0; } From af5fd122d7bd739a2b66405f6e8ab92557279325 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 6 Oct 2023 17:44:05 +0100 Subject: [PATCH 67/79] ASoC: cs35l56: Fix illegal use of init_completion() Fix cs35l56_patch() to call reinit_completion() to reinitialize the completion object. It was incorrectly using init_completion(). Signed-off-by: Richard Fitzgerald Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Link: https://lore.kernel.org/r/20231006164405.253796-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index f2e7c6d0be46..9c2d9cbc63d3 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -706,7 +706,7 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56) mutex_lock(&cs35l56->base.irq_lock); - init_completion(&cs35l56->init_completion); + reinit_completion(&cs35l56->init_completion); cs35l56->soft_resetting = true; cs35l56_system_reset(&cs35l56->base, !!cs35l56->sdw_peripheral); From aa6464edbd51af4a2f8db43df866a7642b244b5f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 5 Oct 2023 17:00:24 +0300 Subject: [PATCH 68/79] ASoC: pxa: fix a memory leak in probe() Free the "priv" pointer before returning the error code. Fixes: 90eb6b59d311 ("ASoC: pxa-ssp: add support for an external clock in devicetree") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/84ac2313-1420-471a-b2cb-3269a2e12a7c@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b70034c07eee..b8a3cb8b7597 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -773,7 +773,7 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) if (IS_ERR(priv->extclk)) { ret = PTR_ERR(priv->extclk); if (ret == -EPROBE_DEFER) - return ret; + goto err_priv; priv->extclk = NULL; } From 1bba0badff0ede8dc51641cff4b153422baa3369 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 9 Oct 2023 16:34:12 +0100 Subject: [PATCH 69/79] ASoC: cs35l56: ASP1 DOUT must default to Hi-Z when not transmitting The ASP1 DOUT line must be defaulted to be high-impedance when it is not actually transmitting data for an active channel. In non-SoundWire modes ASP1 will usually be shared by multiple amps so each amp must only drive the line during the slot for an enabled TX channel. In SoundWire mode a custom firmware can use ASP1 as a secondary chip-to-chip audio link or as GPIO. It should be defaulted to high-impedance since by default the purpose of this pin is not known. Backport note: On kernel versions before 6.6 the cs35l56->base.regmap argument to regmap_set_bits() must be changed to cs35l56->regmap. Signed-off-by: Richard Fitzgerald Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Link: https://lore.kernel.org/r/20231009153412.30380-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 9c2d9cbc63d3..f9059780b7a7 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1186,6 +1186,12 @@ int cs35l56_init(struct cs35l56_private *cs35l56) /* Registers could be dirty after soft reset or SoundWire enumeration */ regcache_sync(cs35l56->base.regmap); + /* Set ASP1 DOUT to high-impedance when it is not transmitting audio data. */ + ret = regmap_set_bits(cs35l56->base.regmap, CS35L56_ASP1_CONTROL3, + CS35L56_ASP1_DOUT_HIZ_CTRL_MASK); + if (ret) + return dev_err_probe(cs35l56->base.dev, ret, "Failed to write ASP1_CONTROL3\n"); + cs35l56->base.init_done = true; complete(&cs35l56->init_completion); From 53ba32acb5ab137ba333c20e0c987bdd6273a366 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 10 Oct 2023 11:24:24 +0100 Subject: [PATCH 70/79] ASoC: dt-bindings: cirrus,cs42l43: Update values for bias sense Due to an error in the datasheet the bias sense values currently don't match the hardware. Whilst this is a change to the binding no devices have yet shipped so updating the binding will not cause any issues. Acked-by: Krzysztof Kozlowski Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20231010102425.3662364-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml index 7a6de938b11d..4118aa54bbd5 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml @@ -82,7 +82,7 @@ properties: description: Current at which the headset micbias sense clamp will engage, 0 to disable. - enum: [ 0, 14, 23, 41, 50, 60, 68, 86, 95 ] + enum: [ 0, 14, 24, 43, 52, 61, 71, 90, 99 ] default: 0 cirrus,bias-ramp-ms: From 99d426c6dd2d6f9734617ec12def856ee35b9218 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 10 Oct 2023 11:24:25 +0100 Subject: [PATCH 71/79] ASoC: cs42l43: Update values for bias sense Due to an error in the datasheet the bias sense values currently don't match the hardware. Whilst this is a change to the binding no devices have yet shipped so updating the binding will not cause any issues. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20231010102425.3662364-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 92e37bc1df9d..9f5f1a92561d 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -34,7 +34,7 @@ static const unsigned int cs42l43_accdet_db_ms[] = { static const unsigned int cs42l43_accdet_ramp_ms[] = { 10, 40, 90, 170 }; static const unsigned int cs42l43_accdet_bias_sense[] = { - 14, 23, 41, 50, 60, 68, 86, 95, 0, + 14, 24, 43, 52, 61, 71, 90, 99, 0, }; static int cs42l43_find_index(struct cs42l43_codec *priv, const char * const prop, From d6cbc6a3a856a7d8047316d81e2e039e44432acb Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 11 Oct 2023 14:48:53 +0100 Subject: [PATCH 72/79] ASoC: cs42l42: Fix missing include of gpio/consumer.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The call to gpiod_set_value_cansleep() in cs42l42_sdw_update_status() needs the header file gpio/consumer.h to be included. This was introduced by commit 2d066c6a7865 ("ASoC: cs42l42: Avoid stale SoundWire ATTACH after hard reset") and caused error: sound/soc/codecs/cs42l42-sdw.c:374:4: error: implicit declaration of function ‘gpiod_set_value_cansleep’; did you mean gpio_set_value_cansleep’? Signed-off-by: Richard Fitzgerald Fixes: 2d066c6a7865 ("ASoC: cs42l42: Avoid stale SoundWire ATTACH after hard reset") Link: https://lore.kernel.org/r/20231011134853.20059-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42-sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 974bae4abfad..94a66a325303 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -6,6 +6,7 @@ #include #include +#include #include #include #include From f88dfbf333b3661faff996bb03af2024d907b76a Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Fri, 13 Oct 2023 17:45:25 +0800 Subject: [PATCH 73/79] ASoC: rt5650: fix the wrong result of key button The RT5650 should enable a power setting for button detection to avoid the wrong result. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20231013094525.715518-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1a137ca3f496..7938b52d741d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3257,6 +3257,8 @@ int rt5645_set_jack_detect(struct snd_soc_component *component, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); } rt5645_irq(0, rt5645); From 4e9a429ae80657bdc502d3f5078e2073656ec5fd Mon Sep 17 00:00:00 2001 From: Roy Chateau Date: Fri, 13 Oct 2023 13:02:39 +0200 Subject: [PATCH 74/79] ASoC: codecs: tas2780: Fix log of failed reset via I2C. Correctly log failures of reset via I2C. Signed-off-by: Roy Chateau Link: https://lore.kernel.org/r/20231013110239.473123-1-roy.chateau@mep-info.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2780.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 86bd6c18a944..41076be23854 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -39,7 +39,7 @@ static void tas2780_reset(struct tas2780_priv *tas2780) usleep_range(2000, 2050); } - snd_soc_component_write(tas2780->component, TAS2780_SW_RST, + ret = snd_soc_component_write(tas2780->component, TAS2780_SW_RST, TAS2780_RST); if (ret) dev_err(tas2780->dev, "%s:errCode:0x%x Reset error!\n", From 9c97790a07dc4f9bdc6e1701003dc9b86f749c71 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Oct 2023 18:37:33 +0100 Subject: [PATCH 75/79] ASoC: dwc: Fix non-DT instantiation Commit d6d6c513f5d2 ("ASoC: dwc: Use ops to get platform data") converted the DesignWare I2S driver to use a DT specific function to obtain platform data but this breaks at least non-DT systems such as AMD. Revert it. Fixes: d6d6c513f5d2 ("ASoC: dwc: Use ops to get platform data") Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20231013-asoc-fix-dwc-v1-1-63211bb746b9@kernel.org Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 22c004179214..9ea4be56d3b7 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -917,7 +917,7 @@ static int jh7110_i2stx0_clk_cfg(struct i2s_clk_config_data *config) static int dw_i2s_probe(struct platform_device *pdev) { - const struct i2s_platform_data *pdata = of_device_get_match_data(&pdev->dev); + const struct i2s_platform_data *pdata = pdev->dev.platform_data; struct dw_i2s_dev *dev; struct resource *res; int ret, irq; From 56e85993896b914032d11e32ecbf8415e7b2f621 Mon Sep 17 00:00:00 2001 From: Luka Guzenko Date: Tue, 17 Oct 2023 00:13:28 +0200 Subject: [PATCH 76/79] ALSA: hda/relatek: Enable Mute LED on HP Laptop 15s-fq5xxx This HP Laptop uses ALC236 codec with COEF 0x07 controlling the mute LED. Enable existing quirk for this device. Signed-off-by: Luka Guzenko Cc: Link: https://lore.kernel.org/r/20231016221328.1521674-1-l.guzenko@web.de 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 3eeecf67c17b..4b68d3df9473 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9722,6 +9722,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), From 5dedc9f53eef7ec07b23686381100d03fb259f50 Mon Sep 17 00:00:00 2001 From: Artem Borisov Date: Sat, 14 Oct 2023 10:50:42 +0300 Subject: [PATCH 77/79] ALSA: hda/realtek: Add quirk for ASUS ROG GU603ZV Enables the SPI-connected Cirrus amp and the required pins for headset mic detection. As of BIOS version 313 it is still necessary to modify the ACPI table to add the related _DSD properties: https://gist.github.com/Flex1911/1bce378645fc95a5743671bd5deabfc8 Signed-off-by: Artem Borisov Cc: Link: https://lore.kernel.org/r/20231014075044.17474-1-dedsa2002@gmail.com 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 4b68d3df9473..b75e28fb4c53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9792,6 +9792,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), From c8c0a03ec1be6b3f3ec1ce91685351235212db19 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 17 Oct 2023 15:30:24 +0800 Subject: [PATCH 78/79] ALSA: hda/realtek - Fixed ASUS platform headset Mic issue ASUS platform Headset Mic was disable by default. Assigned verb table for Mic pin will enable it. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/1155d914c20c40569f56d36c79254879@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b75e28fb4c53..9677c09cf7a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7078,6 +7078,24 @@ static void alc287_fixup_bind_dacs(struct hda_codec *codec, 0x0); /* Make sure 0x14 was disable */ } } +/* Fix none verb table of Headset Mic pin */ +static void alc_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x19, 0x03a1103c }, + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + alc_update_coef_idx(codec, 0x45, 0xf<<12 | 1<<10, 5<<12); + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + break; + } +} enum { @@ -7344,6 +7362,7 @@ enum { ALC245_FIXUP_HP_X360_MUTE_LEDS, ALC287_FIXUP_THINKPAD_I2S_SPK, ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD, + ALC2XX_FIXUP_HEADSET_MIC, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9448,6 +9467,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, }, + [ALC2XX_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10752,6 +10775,8 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC2XX_FIXUP_HEADSET_MIC, + {0x19, 0x40000000}), {} }; From e8ecffd9962fe051d53a0761921b26d653b3df6b Mon Sep 17 00:00:00 2001 From: David Rau Date: Tue, 17 Oct 2023 10:12:58 +0800 Subject: [PATCH 79/79] ASoC: da7219: Correct the process of setting up Gnd switch in AAD Enable Gnd switch to improve stability when Jack insert event occurs, and then disable Gnd switch after Jack type detection is finished. Signed-off-by: David Rau Link: https://lore.kernel.org/r/20231017021258.5929-1-David.Rau.opensource@dm.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7219-aad.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 581b334a6631..3bbe85091649 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,9 +59,6 @@ static void da7219_aad_btn_det_work(struct work_struct *work) bool micbias_up = false; int retries = 0; - /* Disable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); - /* Drive headphones/lineout */ snd_soc_component_update_bits(component, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, @@ -155,9 +152,6 @@ static void da7219_aad_hptest_work(struct work_struct *work) tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ_INT_OSC); } - /* Disable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); - /* Ensure gain ramping at fastest rate */ gain_ramp_ctrl = snd_soc_component_read(component, DA7219_GAIN_RAMP_CTRL); snd_soc_component_write(component, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_X8); @@ -421,6 +415,11 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) * handle a removal, and we can check at the end of * hptest if we have a valid result or not. */ + + cancel_delayed_work_sync(&da7219_aad->jack_det_work); + /* Disable ground switch */ + snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); + if (statusa & DA7219_JACK_TYPE_STS_MASK) { report |= SND_JACK_HEADSET; mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT;