September 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ASoC: imx-rpmsg: Set ignore_pmdown_time for dai_link
by Chancel Liu 14 Sep '23

14 Sep '23

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 <chancel.liu(a)nxp.com> --- 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) { -- 2.25.1

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[RFC PATCH v2 1/2] ALSA: core: add snd_device_init
by cujomalainey＠chromium.org 14 Sep '23

14 Sep '23

From: Curtis Malainey <cujomalainey(a)chromium.org> Begin allowing refactored modules to allocate their own device but use a common initialization procedure for their devices. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- include/sound/core.h | 1 + sound/core/init.c | 19 ++++++++++++++++--- 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index dfef0c9d4b9f7..a4744e142c7e3 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -240,6 +240,7 @@ extern struct dentry *sound_debugfs_root; void snd_request_card(int card); int snd_device_alloc(struct device **dev_p, struct snd_card *card); +void snd_device_init(struct device *dev, struct snd_card *card); int snd_register_device(int type, struct snd_card *card, int dev, const struct file_operations *f_ops, diff --git a/sound/core/init.c b/sound/core/init.c index 22c0d217b8608..87b5368d20350 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -132,15 +132,28 @@ int snd_device_alloc(struct device **dev_p, struct snd_card *card) dev = kzalloc(sizeof(*dev), GFP_KERNEL); if (!dev) return -ENOMEM; + snd_device_init(dev, card); + *dev_p = dev; + return 0; +} +EXPORT_SYMBOL_GPL(snd_device_alloc); + +/** + * snd_device_init - Initialize struct device for sound devices + * @dev_p: pointer to store the allocated device + * @card: card to assign, optional + * + * For releasing the allocated device, call put_device(). + */ +void snd_device_init(struct device *dev, struct snd_card *card) +{ device_initialize(dev); if (card) dev->parent = &card->card_dev; dev->class = &sound_class; dev->release = default_release_alloc; - *dev_p = dev; - return 0; } -EXPORT_SYMBOL_GPL(snd_device_alloc); +EXPORT_SYMBOL_GPL(snd_device_init); static int snd_card_init(struct snd_card *card, struct device *parent, int idx, const char *xid, struct module *module, -- 2.42.0.283.g2d96d420d3-goog

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[PATCH] ALSA: hda: cs35l56: Fix missing RESET GPIO if _SUB is missing
by Richard Fitzgerald 14 Sep '23

14 Sep '23

In cs35l56_hda_read_acpi() do not return if ACPI _SUB is missing. A missing _SUB means that the driver cannot load a system-specific firmware, because the firmware is identified by the _SUB. But it can fallback to a generic firmware. Unfortunately this was being handled by immediately returning 0, which would skip the remaining ACPI configuration in cs35l56_hda_read_acpi() and so it would not get the RESET GPIO. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") --- sound/pci/hda/cs35l56_hda.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 9e4976bdb5e0..a9844336bdc9 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -864,15 +864,11 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int id) sub = acpi_get_subsystem_id(ACPI_HANDLE(cs35l56->base.dev)); - if (IS_ERR(sub)) { - /* If no ACPI SUB, return 0 and fallback to legacy firmware path, otherwise fail */ - if (PTR_ERR(sub) == -ENODATA) - return 0; - else - return PTR_ERR(sub); - } - - cs35l56->system_name = sub; + if (IS_ERR(sub)) + dev_err_probe(cs35l56->base.dev, PTR_ERR(sub), + "Read ACPI _SUB failed: fallback to generic firmware\n"); + else + cs35l56->system_name = sub; cs35l56->base.reset_gpio = devm_gpiod_get_index_optional(cs35l56->base.dev, "reset", -- 2.30.2

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[PATCH] ASoC: SOF: ipc4-topology: fix wrong sizeof argument
by Peter Ujfalusi 14 Sep '23

14 Sep '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> 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 <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- 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; -- 2.42.0

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Why is there no SND_SOC_DAIFMT_TDM bit?
by Daniel Baluta 14 Sep '23

14 Sep '23

Hi, I wonder why there is no SND_SOC_DAIFMT_TDM format although there actual codecs supporting TDM. Most of the drivers I see use SND_SOC_DAIFMT_DSP_A or SND_SOC_DAIFMT_DSP_B to signal that the format used is TDM.

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[PATCH] [v2] ASoC: dwc: fix typo in comment
by Myunguk Kim 14 Sep '23

14 Sep '23

ISR_RXFO means "Status of Data Overrun interrupt for the RX channel" according to the datasheet. So, the comment should be RX, not TX Signed-off-by: Myunguk Kim <mwkim(a)gaonchips.com> --- 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..c71c17ef961d 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -138,7 +138,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Error Handling: TX */ + /* Error Handling: RX */ if (isr[i] & ISR_RXFO) { dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; -- 2.34.1

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[PATCH] ASoC: amd: acp: Fix -Wmissing-prototypes warning
by Marian Postevca 14 Sep '23

14 Sep '23

Fix prototype missing warning for acp3x_es83xx_init_ops() by including the header acp3x-es83xx.h Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202309111220.g63yHDfH-lkp@intel.com/ Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index 47ce2f6c74bb..7ce15216c3f0 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -20,6 +20,7 @@ #include <linux/acpi.h> #include <linux/dmi.h> #include "../acp-mach.h" +#include "acp3x-es83xx.h" #define get_mach_priv(card) ((struct acp3x_es83xx_private *)((acp_get_drvdata(card))->mach_priv)) -- 2.41.0

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[PATCH] firmware: cirrus: cs_dsp: Only log list of algorithms in debug build
by Richard Fitzgerald 14 Sep '23

14 Sep '23

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 <rf(a)opensource.cirrus.com> --- 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 17b7198b87dc..4d702442565e 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1827,15 +1827,15 @@ 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, @@ -1960,14 +1960,14 @@ 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, -- 2.30.2

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[PATCH] ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()
by Richard Fitzgerald 14 Sep '23

14 Sep '23

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 <rf(a)opensource.cirrus.com> --- These functions aren't currently used so this fix doesn't need to be backported. --- 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); -- 2.30.2

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[PATCH v1 0/3] ASoC: cs42l42: Fix handling of hard reset
by Stefan Binding 14 Sep '23

14 Sep '23

These patches fix 3 problems with hard reset: 1. Ensure a minimum reset pulse width 2. Deal with ACPI overriding the requested default GPIO state 3. Avoid a race condition when hard-resetting a SoundWire peripheral that is already enumerated Richard Fitzgerald (3): ASoC: cs42l42: Ensure a reset pulse meets minimum pulse width. ASoC: cs42l42: Don't rely on GPIOD_OUT_LOW to set RESET initially low ASoC: cs42l42: Avoid stale SoundWire ATTACH after hard reset sound/soc/codecs/cs42l42-sdw.c | 20 ++++++++++++++++++++ sound/soc/codecs/cs42l42.c | 21 ++++++++++++++++++++- sound/soc/codecs/cs42l42.h | 1 + 3 files changed, 41 insertions(+), 1 deletion(-) -- 2.34.1

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